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/*
* client.h the implementation type behind the opaque `vc_client*` handle.
*/
#ifndef VOICECAT_CORE_CLIENT_H
#define VOICECAT_CORE_CLIENT_H
#include "voicecat.h"
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#ifdef VOICECAT_HAS_NET
#include <atomic>
#include <condition_variable>
#include <deque>
#include <mutex>
#include <optional>
#include <string>
#include <thread>
#include <unordered_map>
#include <utility>
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#include <vector>
#include "audio/audio_engine.h"
#include "codec/opus_codec.h"
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#include "crypto/crypto.h"
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
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#include "crypto/tofu_store.h"
#include "net/voice_frame.h"
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#include "protocol/envelope.h"
#include "protocol/protocol.h"
#include "session/session.h"
#include "proto/voicecat.pb.h"
#endif // VOICECAT_HAS_NET
struct vc_client {
vc_client(const vc_config& cfg, vc_callbacks cb);
~vc_client();
vc_client(const vc_client&) = delete;
vc_client& operator=(const vc_client&) = delete;
vc_result connect(const char* host, uint16_t port);
vc_result disconnect();
vc_result authenticate_guest(const char* nickname);
vc_result authenticate_user(const char* username, const char* password);
vc_result join_channel(uint32_t channel_id, const char* password);
vc_result leave_channel();
vc_result stream_start(const vc_stream_desc& desc, uint32_t* out_stream_id);
vc_result stream_stop(uint32_t stream_id);
vc_result set_input_device(uint32_t stream_id, const char* device_id);
vc_result set_input_mode(vc_input_mode mode);
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
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vc_result set_vad_threshold(float threshold);
vc_result set_push_to_talk(bool active);
vc_result set_self_mute(bool mic_muted, bool deafened);
vc_result set_remote_stream(uint32_t user_id, uint32_t stream_id, float gain, bool muted,
bool noise_reduction);
vc_result send_text(vc_text_scope scope, uint32_t target_id, const char* utf8);
vc_result list_devices(vc_device_kind kind, vc_device_list* out);
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
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// M4: channel/user/stream snapshot getters (read session_model_; see voicecat.h).
vc_result list_channels(vc_channel_list* out);
vc_result list_users(vc_user_list* out);
vc_result list_user_streams(uint32_t user_id, vc_stream_summary_list* out);
// M4: TOFU server-identity gate (see voicecat.h's VC_EVENT_SERVER_IDENTITY doc comment).
vc_result confirm_server_identity(bool accept);
vc_result get_server_identity_display(char* out_buf, size_t buf_cap, size_t* out_len);
feat(M3): multi-stream & per-channel tuning 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>
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// M3: effective Opus config for a (user_id, stream_id) — our own pending/active local
// streams, or any peer's broadcast StreamInfo.audio.
vc_result get_stream_audio_config(uint32_t user_id, uint32_t stream_id,
vc_audio_config* out);
// TEST-ONLY (see voicecat.h) — inject synthetic PCM into a local stream's encode pipeline.
vc_result test_inject_capture(uint32_t stream_id, const int16_t* pcm, size_t samples);
// M5: moderation & admin.
vc_result kick_user(uint32_t user_id, const char* reason);
vc_result ban_user(uint32_t user_id, const char* reason, uint64_t expires_unix_ms);
vc_result set_permission(uint32_t user_id, const vc_permissions* perms);
vc_result set_server_mute(uint32_t user_id, bool muted, bool deafened);
vc_result move_user(uint32_t user_id, uint32_t channel_id);
vc_result create_channel(const vc_channel_info* info);
vc_result edit_channel(const vc_channel_info* info);
vc_result delete_channel(uint32_t channel_id);
vc_result create_account(const char* username, const char* password);
vc_result reset_password(const char* username, const char* new_password);
vc_result delete_account(const char* username);
vc_result list_accounts();
vc_result get_account_list(vc_account_list* out);
vc_result get_permissions(vc_permissions* out);
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vc_connection_state state() const {
#ifdef VOICECAT_HAS_NET
return state_net_.load(std::memory_order_acquire);
#else
return state_;
#endif
}
private:
void emit(const vc_event& ev) const;
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vc_config cfg_{};
vc_callbacks cb_{};
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#ifdef VOICECAT_HAS_NET
// ── M1: TCP/TLS control channel ─────────────────────────────────────────────
std::atomic<vc_connection_state> state_net_{VC_STATE_DISCONNECTED};
// Blocking I/O thread (one per vc_client lifetime)
std::thread io_thread_;
std::atomic<bool> io_stop_{false};
// Send queue: pushed by any thread, drained by io_thread_
std::mutex send_mutex_;
std::condition_variable send_cv_;
std::deque<std::vector<uint8_t>> send_queue_;
// TLS context — created + used exclusively on io_thread_
std::unique_ptr<voicecat::crypto::TlsContext> tls_;
// Raw socket fd (stored after TCP connect; closed by disconnect())
std::atomic<int> io_fd_{-1};
// Pending auth stored before ServerHello arrives
struct PendingAuth {
bool is_guest{true};
std::string nick_or_user;
std::string password;
};
std::mutex pending_auth_mutex_;
std::optional<PendingAuth> pending_auth_;
// Self identity filled in after AuthResult
uint32_t self_user_id_{0};
uint64_t server_session_id_{0};
std::atomic<uint64_t> next_req_id_{1};
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
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// Client-side session model. Mutated only on io_thread_ (handle_server_state/
// handle_user_event/handle_channel_event), but read from any thread via the M4
// list_channels/list_users/list_user_streams getters — session_model_mu_ guards both.
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voicecat::session::SessionModel session_model_;
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
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mutable std::mutex session_model_mu_;
// M5: last ListAccountsResult snapshot, populated on io_thread_ when
// VC_EVENT_ACCOUNT_LIST fires and read by vc_get_account_list on caller threads.
std::vector<voicecat::v1::AccountEntry> last_account_list_;
mutable std::mutex account_list_mu_;
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
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// ── M4: TOFU server-identity gate ───────────────────────────────────────────
std::unique_ptr<voicecat::crypto::TofuStore> tofu_store_; // owns the pin file
std::mutex tofu_mu_;
std::condition_variable tofu_cv_;
bool tofu_decision_pending_{false};
bool tofu_accept_{false};
std::string pending_identity_fp_hex_; // ServerHello's Ed25519 fp, display-only
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// ── M2: UDP / media plane ────────────────────────────────────────────────────
std::array<uint8_t, 16> udp_token_{};
uint16_t server_udp_port_{0};
std::string udp_host_;
std::atomic<int> udp_fd_{-1};
std::thread udp_thread_;
std::atomic<bool> udp_stop_{false};
std::atomic<bool> udp_ready_{false};
std::unique_ptr<voicecat::crypto::SodiumMediaCrypto> media_send_crypto_;
std::unique_ptr<voicecat::crypto::SodiumMediaCrypto> media_recv_crypto_;
voicecat::audio::AudioEngine audio_engine_;
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
// M3: one LocalStream per concurrently-active stream kind (MIC/SCREEN_AUDIO/AUX_DEVICE
// are each singletons for a given client — see docs/roadmap.md §M3). Replaces the M2
// single-stream fields (local_encoder_/local_stream_active_/etc).
struct LocalStream {
voicecat::codec::OpusEncoder encoder;
std::atomic<bool> active{false};
bool pending{false}; // announced, awaiting StreamAnnounceResult
uint32_t stream_id{0};
uint32_t ssrc{0};
uint32_t timestamp{0};
uint16_t frame_samples{960};
uint64_t pending_request_id{0};
// Full effective OpusParams from the last StreamAnnounceResult — retained so
// vc_get_stream_audio_config() has something to read back for our own streams.
voicecat::codec::OpusParams effective_params;
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// Talk-indicator edge detection (docs/voice.md §7) — updated in on_capture_frame, after
// the VAD/PTT gate so a gated-closed frame doesn't show as "talking".
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
std::atomic<int64_t> last_capture_ms{0};
bool talking = false;
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// Device-enumeration follow-up: the device this stream's capture should use ("" =
// default). Only meaningful for VC_STREAM_MIC today (the real capture device); set via
// vc_set_input_device. Opaque id from AudioEngine::enumerate_devices — see
// audio_engine.h's DeviceInfo doc comment.
std::string capture_device_id;
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
};
mutable std::mutex local_streams_mu_;
std::unordered_map<int, LocalStream> local_streams_; // keyed by vc_stream_kind
std::unordered_map<uint64_t, int> pending_announce_kind_; // request_id -> kind
uint32_t next_local_stream_id_{1};
// ssrc → (user_id, stream_id) for remote streams already wired into audio_engine_.
mutable std::mutex remote_streams_mu_;
std::unordered_map<uint32_t, std::pair<uint32_t, uint32_t>> remote_streams_;
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
// M3: talk-indicator polling thread (separate from udp_thread_ / the miniaudio callback
// thread — see architecture.md §3 real-time rule).
std::thread talk_timer_thread_;
std::atomic<bool> talk_timer_stop_{false};
static constexpr int64_t kTalkPollMs = 100;
static constexpr int64_t kTalkHangoverMs = 300;
// Local UDP destination (server media endpoint), resolved once during binding.
uint32_t udp_dest_addr_{0}; // network byte order
uint16_t udp_dest_port_{0}; // network byte order
std::atomic<bool> self_mic_muted_{false};
std::atomic<bool> self_deafened_{false};
std::atomic<bool> server_muted_{false};
std::atomic<bool> server_deafened_{false};
// M5: permissions from last AuthResult.
vc_permissions own_permissions_{};
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
// Follow-up to M3: send-side input gate (docs/voice.md §11). MIC-only — SCREEN_AUDIO/
// AUX_DEVICE are never gated (see PROGRESS.md for the rationale). Pure local state, no
// protocol traffic. mic_vad_ is constructed once the MIC stream's StreamAnnounceResult
// lands (handle_stream_announce_result, on io_thread_ — not the RT capture callback).
std::atomic<vc_input_mode> current_input_mode_{VC_INPUT_VOICE_ACTIVATION};
std::atomic<bool> ptt_active_{false};
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-17 00:35:16 +02:00
std::atomic<float> vad_threshold_{0.025f}; // remembered across mode switches
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback Closes the three items PROGRESS.md's M3 section explicitly carried forward as out of scope: - Device enumeration (vc_list_devices) + input device selection (vc_set_input_device), backed by AudioEngine::enumerate_devices() via miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded ma_device_id strings. - VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk). webrtc-audio-processing (the originally-planned APM) has no working Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard abseil-cpp dependency), so VAD is a new lightweight, dependency-free energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it. - True stereo playback: AudioEngine's mixer and output device now carry stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded stereo streams to mono before mixing. - Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via miniaudio's loopback device type), replacing test-only injection as the production capture path. Also: vccli gains --list-devices, --input-device, --input-mode, and --share-screen-audio flags, plus a stdin command loop (ptt on/off, mode vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all four items (ABI-level + a white-box AudioEngine stereo-mix check). Docs updated to match: voice.md, roadmap.md (decision-log entry superseding the original webrtc-audio-processing choice), tech-stack.md, README.md, architecture.md, CLAUDE.md, PROGRESS.md. Still explicitly out of scope, documented not silently dropped: real webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS SCREEN_AUDIO capture, process-specific loopback, and a pre-existing RT-thread rule violation in the capture path that predates this work. Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive standalone runs; manually verified live (vccli --list-devices against real hardware, vccli --voice --input-mode vad streaming without incident). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 16:11:52 +02:00
std::unique_ptr<voicecat::audio::ApmProcessor> mic_vad_;
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
// teardown_voice() is called both from run_io()'s own cleanup (on the io_thread_, when
// the read loop exits) and from disconnect() (on the caller's thread) -- without
// serializing those two call sites, both can see udp_thread_/talk_timer_thread_ as
// joinable() at the same time and race to join() the same std::thread object (UB; an
// intermittent "No such process" std::system_error on Windows is the typical symptom).
// This mutex makes teardown_voice() idempotent under concurrent calls.
std::mutex teardown_mu_;
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// ── io_thread_ entry point ──────────────────────────────────────────────────
void run_io(std::string host, uint16_t port);
// ── Protocol dispatch (called on io_thread_) ────────────────────────────────
void handle_envelope(const voicecat::v1::Envelope& env);
void handle_server_hello(const voicecat::v1::ServerHello& msg, uint64_t req_id);
void handle_auth_result(const voicecat::v1::AuthResult& msg);
void handle_server_state(const voicecat::v1::ServerStateSnapshot& snap);
void handle_user_event(const voicecat::v1::UserEvent& ue);
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode Core ABI extensions (voicecat.h): - vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads - VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password - VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_ until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519) - vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only - VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate - vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores it atomically so the audio RT path reads without a lock C++ implementation: - SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id, password_protected, and max_users (were permanently zeroed) - TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS - TofuStore split into peek (read-only) + pin (write) so first-connect only persists after user approval; tofu_store_path in vc_config for per-user pin file location - TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows) - windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests - New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green) Windows client (clients/windows/ — .NET 10 WinForms): - VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks, Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer - VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity- Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode, per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar) - PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation) - PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams - Accessibility: explicit AccessibleName/Description on every control, & mnemonics, Activity log ListBox as durable screen-reader record, AutomationNotification for curated live announcements Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-17 00:35:16 +02:00
void handle_channel_event(const voicecat::v1::ChannelEvent& ce);
void handle_join_channel_result(const voicecat::v1::JoinChannelResult& msg);
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void handle_text_message(const voicecat::v1::TextMessage& msg);
void handle_disconnect(const voicecat::v1::Disconnect& msg);
void handle_udp_binding_ack(const voicecat::v1::UdpBinding& msg);
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
void handle_stream_announce_result(uint64_t req_id,
const voicecat::v1::StreamAnnounceResult& msg);
// ── M2: UDP / media helpers ──────────────────────────────────────────────────
// Kicks off TCP UdpBinding request; called once after a successful AuthResult.
void start_udp_binding();
// Opens the UDP socket, sends the plaintext bootstrap packet, starts udp_thread_.
void finish_udp_binding();
// udp_thread_ entry point: recv loop, AEAD-open, decode, push to audio_engine_.
void run_udp_recv();
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
// capture_cb passed to audio_engine_.start(): encode + seal + send one frame for the
// given local stream `kind` (M3: multiple concurrent local streams are possible).
void on_capture_frame(int kind, const int16_t* pcm, int samples, int channels);
// Inspect a User proto's streams and wire up any new remote ssrc into audio_engine_,
// emitting VC_EVENT_STREAM_STARTED/STOPPED as streams appear/disappear.
void sync_remote_streams(const voicecat::v1::User& user);
// Starts audio_engine_ (capture+playback) if not already running.
void ensure_audio_running();
// Joins udp_thread_, stops audio_engine_, clears media crypto/remote-stream state.
// Safe to call multiple times. Called both from run_io()'s cleanup and disconnect().
void teardown_voice();
feat(M3): multi-stream & per-channel tuning 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>
2026-06-16 14:12:37 +02:00
// talk_timer_thread_ entry point (M3): polls audio_engine_ for remote talk-state edges
// and local capture activity, emitting VC_EVENT_TALK_STATE. Never the audio RT thread.
void run_talk_timer();
// Find a LocalStream by its client-assigned stream_id (held under local_streams_mu_ by
// the caller, or taken internally). Returns nullptr if not found/not active.
LocalStream* find_local_stream_by_id(uint32_t stream_id);
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// ── Helpers (io_thread_ and caller threads) ─────────────────────────────────
// Queue an encoded envelope to be sent on io_thread_.
void queue_envelope(const voicecat::v1::Envelope& env);
// Drain send_queue_ by doing blocking TLS writes (called on io_thread_).
void drain_sends();
// Transition state + emit VC_EVENT_CONNECTION_STATE.
void set_state(vc_connection_state s);
// Convenience event emitters.
void emit_error(vc_result r, const char* text);
void emit_disconnected(vc_result r, const char* reason);
#else // !VOICECAT_HAS_NET
vc_connection_state state_{VC_STATE_DISCONNECTED};
#endif
};
#endif // VOICECAT_CORE_CLIENT_H