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voice-cat/server/src/conn_session.cpp

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#include "conn_session.h"
#ifdef VOICECAT_HAS_NET
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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#include <algorithm>
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#include <chrono>
#include <cstdio>
#include <cstring>
#include <sodium.h>
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#include "db.h"
#include "session_registry.h"
#include "core/worker_pool.h"
#include "protocol/envelope.h"
namespace voicecat::server {
static voicecat::v1::Envelope make_env(uint64_t req_id = 0) {
voicecat::v1::Envelope e;
e.set_request_id(req_id);
return e;
}
static voicecat::v1::Permissions all_permissions() {
voicecat::v1::Permissions p;
p.set_can_create_temp_channel(true);
p.set_can_kick(true);
p.set_can_ban(true);
p.set_can_move_users(true);
p.set_can_admin_accounts(true);
p.set_is_admin(true);
return p;
}
static voicecat::v1::Permissions no_permissions() {
voicecat::v1::Permissions p;
return p;
}
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ConnSession::ConnSession(std::shared_ptr<Database> db,
std::shared_ptr<SessionRegistry> registry,
std::shared_ptr<voicecat::WorkerPool> workers,
const std::array<uint8_t, 32>& server_fp,
bool allow_guests,
uint16_t udp_media_port)
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: db_(std::move(db)),
registry_(std::move(registry)),
workers_(std::move(workers)),
server_fp_(server_fp),
allow_guests_(allow_guests),
udp_media_port_(udp_media_port) {
randombytes_buf(udp_token_.data(), udp_token_.size());
}
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void ConnSession::set_io(SendFn send_fn, CloseFn close_fn) {
send_fn_ = std::move(send_fn);
close_fn_ = std::move(close_fn);
}
void ConnSession::begin() {
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
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touch_last_seen();
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// Nothing to do at TCP level — wait for ClientHello
}
void ConnSession::on_frame(std::vector<uint8_t> frame) {
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
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touch_last_seen(); // any inbound TCP activity keeps this session off the reaper's list
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voicecat::v1::Envelope env;
if (!protocol::decode_envelope(frame, env)) return;
auto st = state_.load(std::memory_order_acquire);
switch (env.body_case()) {
case voicecat::v1::Envelope::kClientHello:
if (st == State::WaitingHello)
handle_client_hello(env.request_id(), env.client_hello());
break;
case voicecat::v1::Envelope::kAuthRequest:
if (st == State::WaitingAuth)
handle_auth_request(env.request_id(), env.auth_request());
break;
case voicecat::v1::Envelope::kJoinChannel:
if (st == State::Authenticated)
handle_join_channel(env.request_id(), env.join_channel());
break;
case voicecat::v1::Envelope::kTextMessage:
if (st == State::Authenticated)
handle_text_message(env.text_message());
break;
case voicecat::v1::Envelope::kPing:
handle_ping(env.ping());
break;
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
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case voicecat::v1::Envelope::kDisconnect:
// Client-initiated graceful disconnect (code=0). Falls through to close() which
// broadcasts UserEvent::LEFT — same as a TCP drop, but immediate (no reaper/EOF
// wait). Gated on Authenticated so a pre-auth stray Disconnect can't skip cleanup.
if (st == State::Authenticated) handle_client_disconnect(env.disconnect());
break;
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case voicecat::v1::Envelope::kLeaveChannel:
if (st == State::Authenticated)
handle_leave_channel();
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break;
case voicecat::v1::Envelope::kUdpBinding:
if (st == State::Authenticated)
handle_udp_binding(env.request_id(), env.udp_binding());
break;
case voicecat::v1::Envelope::kStreamAnnounce:
if (st == State::Authenticated)
handle_stream_announce(env.request_id(), env.stream_announce());
break;
case voicecat::v1::Envelope::kStreamStop:
if (st == State::Authenticated)
handle_stream_stop(env.stream_stop());
break;
// ── M5 moderation / admin ─────────────────────────────────────────────
case voicecat::v1::Envelope::kKick:
if (st == State::Authenticated) handle_kick_request(env.request_id(), env.kick());
break;
case voicecat::v1::Envelope::kBan:
if (st == State::Authenticated) handle_ban_request(env.request_id(), env.ban());
break;
case voicecat::v1::Envelope::kSetPermission:
if (st == State::Authenticated)
handle_set_permission(env.request_id(), env.set_permission());
break;
case voicecat::v1::Envelope::kServerMute:
if (st == State::Authenticated)
handle_server_mute_request(env.request_id(), env.server_mute());
break;
case voicecat::v1::Envelope::kMoveUser:
if (st == State::Authenticated) handle_move_user(env.request_id(), env.move_user());
break;
case voicecat::v1::Envelope::kCreateChannel:
if (st == State::Authenticated)
handle_create_channel(env.request_id(), env.create_channel());
break;
case voicecat::v1::Envelope::kEditChannel:
if (st == State::Authenticated)
handle_edit_channel(env.request_id(), env.edit_channel());
break;
case voicecat::v1::Envelope::kDeleteChannel:
if (st == State::Authenticated)
handle_delete_channel(env.request_id(), env.delete_channel());
break;
case voicecat::v1::Envelope::kCreateAccount:
if (st == State::Authenticated)
handle_create_account(env.request_id(), env.create_account());
break;
case voicecat::v1::Envelope::kResetPassword:
if (st == State::Authenticated)
handle_reset_password(env.request_id(), env.reset_password());
break;
case voicecat::v1::Envelope::kDeleteAccount:
if (st == State::Authenticated)
handle_delete_account(env.request_id(), env.delete_account());
break;
case voicecat::v1::Envelope::kListAccounts:
if (st == State::Authenticated)
handle_list_accounts(env.request_id(), env.list_accounts());
break;
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default:
break;
}
}
void ConnSession::on_disconnect() { close(); }
void ConnSession::send_envelope(const voicecat::v1::Envelope& env) {
if (!send_fn_ || closed_.load()) return;
// Serialize to raw protobuf bytes; send_fn_ (→ TcpServerConn::send_frame)
// adds the [4-byte len] framing, so we must NOT pre-frame here.
std::string bytes;
if (!env.SerializeToString(&bytes)) return;
std::vector<uint8_t> raw(bytes.begin(), bytes.end());
send_fn_(std::move(raw));
}
void ConnSession::close() {
if (closed_.exchange(true)) return;
state_.store(State::Disconnecting, std::memory_order_release);
uint32_t uid = user_id_.load();
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
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if (uid) {
// Broadcast LEFT BEFORE erasing the user, so remaining clients (and the audio
// engine's remove_stream path on each peer) learn about the departure. This
// covers ungraceful disconnects (TCP drop, crash, network loss) that previously
// silently erased the user from the registry without notifying anyone — which
// left stale users in peer client lists and kept Opus PLC hissing forever on
// peers whose remove_stream was never triggered. Mirrors kick_user's first half
// (session_registry.cpp kick_user). broadcast_left releases its shared_lock
// before remove_user acquires the unique_lock, so no deadlock; and send_envelope
// on this session is a no-op now that closed_ is true.
registry_->broadcast_left(uid, "");
registry_->remove_user(uid);
}
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if (session_id_) registry_->unregister_session(session_id_);
if (close_fn_) close_fn_();
}
// ── M2: media crypto ─────────────────────────────────────────────────────────
void ConnSession::set_media_crypto(
std::unique_ptr<voicecat::crypto::SodiumMediaCrypto> send,
std::unique_ptr<voicecat::crypto::SodiumMediaCrypto> recv) {
std::lock_guard lk(crypto_mu_);
send_crypto_ = std::move(send);
recv_crypto_ = std::move(recv);
}
voicecat::crypto::SodiumMediaCrypto* ConnSession::send_crypto() {
std::lock_guard lk(crypto_mu_);
return send_crypto_.get();
}
voicecat::crypto::SodiumMediaCrypto* ConnSession::recv_crypto() {
std::lock_guard lk(crypto_mu_);
return recv_crypto_.get();
}
// ── M2: UDP endpoint ─────────────────────────────────────────────────────────
void ConnSession::set_udp_endpoint(asio::ip::udp::endpoint ep) {
{
std::lock_guard lk(udp_ep_mu_);
udp_ep_ = ep;
}
has_udp_ep_.store(true, std::memory_order_release);
registry_->register_udp_endpoint(ep, session_id_);
}
asio::ip::udp::endpoint ConnSession::udp_endpoint() const {
std::lock_guard lk(udp_ep_mu_);
return udp_ep_;
}
// ── Permission helpers ───────────────────────────────────────────────────────
bool ConnSession::has_permission(bool (voicecat::v1::Permissions::* getter)() const) const {
return (permissions_.*getter)();
}
void ConnSession::set_permissions(const voicecat::v1::Permissions& perms) {
permissions_ = perms;
registry_->set_session_permissions(session_id_, perms);
}
void ConnSession::send_generic_result(uint64_t req_id, bool ok, uint32_t code,
const std::string& message) {
auto env = make_env(req_id);
auto* gr = env.mutable_generic_result();
gr->set_ok(ok);
gr->set_code(code);
gr->set_message(message);
send_envelope(env);
}
// ── Handlers ─────────────────────────────────────────────────────────────────
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void ConnSession::handle_client_hello(uint64_t req_id, const voicecat::v1::ClientHello& msg) {
if (msg.proto_version() != 2) { // protocol v2: 64-bit voice seq (docs/voice.md §2)
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send_disconnect_and_close(1, "unsupported protocol version");
return;
}
auto env = make_env(req_id);
auto* hello = env.mutable_server_hello();
hello->set_proto_version(2);
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hello->set_server_name("VoiceCat Server");
hello->set_server_version("0.1.0");
if (allow_guests_) hello->add_auth_methods("guest");
hello->add_auth_methods("password");
hello->set_server_identity_fingerprint(server_fp_.data(), server_fp_.size());
if (udp_media_port_) hello->set_udp_port(udp_media_port_);
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send_envelope(env);
state_.store(State::WaitingAuth, std::memory_order_release);
}
void ConnSession::handle_auth_request(uint64_t req_id, const voicecat::v1::AuthRequest& msg) {
if (msg.has_guest()) {
finish_guest_auth(msg.guest(), req_id);
} else if (msg.has_password()) {
finish_password_auth(msg.password().username(), msg.password().password(), req_id);
} else {
auto env = make_env(req_id);
env.mutable_auth_result()->set_ok(false);
env.mutable_auth_result()->set_error("unknown auth method");
send_envelope(env);
}
}
void ConnSession::finish_guest_auth(const voicecat::v1::GuestAuth& guest, uint64_t req_id) {
if (!allow_guests_) {
auto env = make_env(req_id);
env.mutable_auth_result()->set_ok(false);
env.mutable_auth_result()->set_error("guest login not permitted");
send_envelope(env);
return;
}
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voicecat::v1::User user;
user.set_nickname(guest.nickname().empty() ? "Guest" : guest.nickname());
user.set_is_guest(true);
user.set_channel_id(1);
uint32_t uid = registry_->add_user(session_id_, user);
user.set_id(uid);
user_id_.store(uid, std::memory_order_relaxed);
state_.store(State::Authenticated, std::memory_order_release);
permissions_ = no_permissions();
registry_->set_session_permissions(session_id_, permissions_);
registry_->register_udp_token(udp_token_, session_id_);
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{
auto env = make_env(req_id);
auto* res = env.mutable_auth_result();
res->set_ok(true);
res->set_session_id(session_id_);
*res->mutable_self() = user;
*res->mutable_permissions() = permissions_;
res->set_udp_token(udp_token_.data(), udp_token_.size());
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send_envelope(env);
}
broadcast_user_joined(user);
send_state_snapshot();
}
void ConnSession::finish_password_auth(const std::string& username,
const std::string& password, uint64_t req_id) {
// Argon2id runs on the worker pool (deliberately slow).
auto self = shared_from_this();
workers_->post([self, username, password, req_id] {
// M5: check username bans before verifying password.
if (self->db_->ban_check("username", username)) {
auto env = make_env(req_id);
env.mutable_auth_result()->set_ok(false);
env.mutable_auth_result()->set_error("account banned");
self->send_envelope(env);
return;
}
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auto acc = self->db_->authenticate(username, password);
if (!acc) {
auto env = make_env(req_id);
env.mutable_auth_result()->set_ok(false);
env.mutable_auth_result()->set_error("invalid credentials");
self->send_envelope(env);
return;
}
voicecat::v1::User user;
user.set_nickname(acc->username);
user.set_is_guest(false);
user.set_channel_id(1);
uint32_t uid = self->registry_->add_user(self->session_id_, user);
user.set_id(uid);
self->user_id_.store(uid, std::memory_order_relaxed);
self->state_.store(State::Authenticated, std::memory_order_release);
voicecat::v1::Permissions perms = acc->is_admin ? all_permissions() : no_permissions();
self->permissions_ = perms;
self->registry_->set_session_permissions(self->session_id_, perms);
self->registry_->register_udp_token(self->udp_token_, self->session_id_);
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{
auto env = make_env(req_id);
auto* res = env.mutable_auth_result();
res->set_ok(true);
res->set_session_id(self->session_id_);
*res->mutable_self() = user;
*res->mutable_permissions() = perms;
res->set_udp_token(self->udp_token_.data(), self->udp_token_.size());
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self->send_envelope(env);
}
self->broadcast_user_joined(user);
self->send_state_snapshot();
});
}
void ConnSession::send_state_snapshot() {
auto env = make_env();
auto* snap = env.mutable_server_state();
for (auto& ch : registry_->channel_snapshot()) *snap->add_channels() = ch;
for (auto& u : registry_->user_snapshot()) *snap->add_users() = u;
send_envelope(env);
}
void ConnSession::broadcast_user_joined(const voicecat::v1::User& user) {
auto bcast = make_env();
auto* ue = bcast.mutable_user_event();
ue->set_kind(voicecat::v1::UserEvent::JOINED);
*ue->mutable_user() = user;
registry_->broadcast(bcast, session_id_);
}
void ConnSession::handle_join_channel(uint64_t req_id,
const voicecat::v1::JoinChannelRequest& msg) {
uint32_t uid = user_id_.load();
auto ch = registry_->get_channel(msg.channel_id());
if (!ch) {
auto env = make_env(req_id);
auto* res = env.mutable_join_channel_result();
res->set_ok(false);
res->set_error("channel not found");
send_envelope(env);
return;
}
if (ch->password_protected()) {
if (!registry_->check_channel_password(msg.channel_id(), msg.password())) {
auto env = make_env(req_id);
auto* res = env.mutable_join_channel_result();
res->set_ok(false);
res->set_error("invalid channel password");
send_envelope(env);
return;
}
}
if (ch->max_users() > 0) {
auto members = registry_->find_channel_sessions(msg.channel_id(), session_id_);
if (members.size() >= ch->max_users()) {
auto env = make_env(req_id);
auto* res = env.mutable_join_channel_result();
res->set_ok(false);
res->set_error("channel is full");
send_envelope(env);
return;
}
}
bool ok = registry_->set_user_channel(uid, msg.channel_id());
2026-06-15 23:48:44 +02:00
auto env = make_env(req_id);
auto* res = env.mutable_join_channel_result();
res->set_ok(ok);
if (!ok) {
res->set_error("channel not found");
} else {
res->set_channel_id(msg.channel_id());
*res->mutable_audio() = ch->audio();
fix(protocol): deliver self-initiated state changes to the actor too A connected Windows client would randomly snap from its joined channel back to Lobby. Root cause was a state-sync inconsistency, not a drop: the server delivered self-initiated state changes (channel join/leave, stream announce/stop) only as a private *Result to the actor and broadcast the authoritative UserEvent::UPDATED to everyone else. The core never applied the result to its SessionModel, so vc_list_users() kept self in the old channel; the Windows HandleUserUpdated rebuilds _currentChannelId from vc_list_users() on any user's UPDATED event, so the next unrelated event surfaced the stale self-channel. Fix, per the response-vs-broadcast contract now documented in docs/protocol.md §6: the *Result is pure ack/correlation/actor-private payload; the resulting state change is broadcast to every client INCLUDING the actor, and clients apply it to their local model rather than re-deriving own state from a *Result. - server: join/leave/stream announce+stop broadcast with exclude=0 - server: text fan-out includes the sender (channel + private echo) - core: response handlers no longer mutate session_model_ - windows: drop optimistic text echo; render own message via the relay - docs/protocol.md §6: document the response-vs-broadcast contract Registry-level admin broadcasts (move/mute/kick/channel CRUD) already used exclude=0 and were correct. ctest build/m1-dev 18/18 green; VoiceCat.App builds 0 warnings. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-17 20:48:50 +02:00
// Broadcast that this user changed channel — to everyone INCLUDING the mover.
// The JoinChannelResult only acks the request; this UserEvent is the authoritative
// state change every client (mover included) applies to its local model. Excluding
// the mover here is what made it read a stale self-channel. See docs/protocol.md §6.
if (auto updated_user = registry_->user_snapshot_user(uid)) {
auto bcast = make_env();
auto* ue = bcast.mutable_user_event();
ue->set_kind(voicecat::v1::UserEvent::UPDATED);
*ue->mutable_user() = *updated_user;
fix(protocol): deliver self-initiated state changes to the actor too A connected Windows client would randomly snap from its joined channel back to Lobby. Root cause was a state-sync inconsistency, not a drop: the server delivered self-initiated state changes (channel join/leave, stream announce/stop) only as a private *Result to the actor and broadcast the authoritative UserEvent::UPDATED to everyone else. The core never applied the result to its SessionModel, so vc_list_users() kept self in the old channel; the Windows HandleUserUpdated rebuilds _currentChannelId from vc_list_users() on any user's UPDATED event, so the next unrelated event surfaced the stale self-channel. Fix, per the response-vs-broadcast contract now documented in docs/protocol.md §6: the *Result is pure ack/correlation/actor-private payload; the resulting state change is broadcast to every client INCLUDING the actor, and clients apply it to their local model rather than re-deriving own state from a *Result. - server: join/leave/stream announce+stop broadcast with exclude=0 - server: text fan-out includes the sender (channel + private echo) - core: response handlers no longer mutate session_model_ - windows: drop optimistic text echo; render own message via the relay - docs/protocol.md §6: document the response-vs-broadcast contract Registry-level admin broadcasts (move/mute/kick/channel CRUD) already used exclude=0 and were correct. ctest build/m1-dev 18/18 green; VoiceCat.App builds 0 warnings. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-17 20:48:50 +02:00
registry_->broadcast(bcast, /*exclude*/ 0);
}
}
2026-06-15 23:48:44 +02:00
send_envelope(env);
}
void ConnSession::handle_leave_channel() {
uint32_t uid = user_id_.load();
if (!uid) return;
if (!registry_->set_user_channel(uid, 1)) return;
if (auto updated_user = registry_->user_snapshot_user(uid)) {
auto bcast = make_env();
auto* ue = bcast.mutable_user_event();
ue->set_kind(voicecat::v1::UserEvent::UPDATED);
*ue->mutable_user() = *updated_user;
fix(protocol): deliver self-initiated state changes to the actor too A connected Windows client would randomly snap from its joined channel back to Lobby. Root cause was a state-sync inconsistency, not a drop: the server delivered self-initiated state changes (channel join/leave, stream announce/stop) only as a private *Result to the actor and broadcast the authoritative UserEvent::UPDATED to everyone else. The core never applied the result to its SessionModel, so vc_list_users() kept self in the old channel; the Windows HandleUserUpdated rebuilds _currentChannelId from vc_list_users() on any user's UPDATED event, so the next unrelated event surfaced the stale self-channel. Fix, per the response-vs-broadcast contract now documented in docs/protocol.md §6: the *Result is pure ack/correlation/actor-private payload; the resulting state change is broadcast to every client INCLUDING the actor, and clients apply it to their local model rather than re-deriving own state from a *Result. - server: join/leave/stream announce+stop broadcast with exclude=0 - server: text fan-out includes the sender (channel + private echo) - core: response handlers no longer mutate session_model_ - windows: drop optimistic text echo; render own message via the relay - docs/protocol.md §6: document the response-vs-broadcast contract Registry-level admin broadcasts (move/mute/kick/channel CRUD) already used exclude=0 and were correct. ctest build/m1-dev 18/18 green; VoiceCat.App builds 0 warnings. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-17 20:48:50 +02:00
registry_->broadcast(bcast, /*exclude*/ 0); // include the leaver — same as join
}
}
2026-06-15 23:48:44 +02:00
void ConnSession::handle_text_message(const voicecat::v1::TextMessage& msg) {
using namespace std::chrono;
int64_t now_ms = duration_cast<milliseconds>(
system_clock::now().time_since_epoch()).count();
voicecat::v1::TextMessage relay = msg;
relay.set_sender_id(user_id_.load(std::memory_order_relaxed));
relay.set_sent_at_unix_ms(now_ms);
voicecat::v1::Envelope fwd;
*fwd.mutable_text_message() = relay;
auto targets = registry_->resolve_text_targets(session_id_, msg.scope(), msg.target_id());
for (auto& t : targets) t->send_envelope(fwd);
// Ack
auto env = make_env();
auto* ack = env.mutable_text_message_ack();
ack->set_client_msg_id(msg.client_msg_id());
ack->set_ok(true);
send_envelope(env);
}
void ConnSession::handle_ping(const voicecat::v1::Ping& msg) {
auto env = make_env();
env.mutable_pong()->set_nonce(msg.nonce());
send_envelope(env);
}
fix(net): broadcast LEFT on disconnect, add keepalive/reaper, cap PLC hiss Three reported bugs traced to one root cause plus two missing designed features: 1. Stale users + eternal PLC hiss (root cause): ConnSession::close() silently erased dropped users without broadcasting UserEvent::LEFT, so peers never learned the user left and their audio engines never called remove_stream — Opus PLC synthesized comfort noise forever. Fix: broadcast_left() helper + close() broadcasts LEFT before erasing. 2. PLC cap (defense-in-depth): on_playback now caps pure PLC at ~2s, then emits digital silence so a stale stream can never hiss forever even if remove_stream is skipped. Resets automatically on fresh packets. 3. No timeout / no ping: client never sent Ping, server had no last_seen / reaper, so half-open connections (NAT timeout, wifi loss, sleep) left ghost users forever. Fix: client Ping every 15s with RTT measurement, ConnSession::last_seen bumped on every inbound TCP/UDP frame, steady_timer reaper sweeps every 15s and drops sessions older than 45s (configurable via server::Config). 4. UDP KEEPALIVE: client sends plaintext kFrameKeepalive every 5s; server bumps last_seen + echoes back. Keeps NAT bindings alive and lets media activity defer the reaper independently of TCP. 5. Graceful client disconnect: vc_disconnect() sends Disconnect{code=0} via a flag-based io-thread exit (no double-close race); server handles client-sent Disconnect with immediate close() + LEFT broadcast. 3 new tests: disconnect_left, plc_cap, reaper_timeout. 21/21 ctest green. Docs: protocol.md §6/§7, voice.md §6, architecture.md §5, PROGRESS.md.
2026-06-18 01:18:33 +02:00
void ConnSession::handle_client_disconnect(const voicecat::v1::Disconnect& /*msg*/) {
// Graceful client-initiated disconnect (code=0). close() broadcasts UserEvent::LEFT and
// cleans up the registry — identical to the TCP-drop path, but triggered by the client's
// explicit goodbye so peers learn about the departure immediately (no reaper timeout,
// no EOF detection delay). Idempotent: the closed_ guard in close() handles the
// inevitable follow-up TCP EOF when the client closes its socket.
close();
}
void ConnSession::handle_udp_binding(uint64_t req_id, const voicecat::v1::UdpBinding& msg) {
if (msg.ack()) return; // server→client direction; ignore if echoed back
const std::string& tok = msg.udp_token();
if (tok.size() != 16 || std::memcmp(tok.data(), udp_token_.data(), 16) != 0) {
// Bad token — silently ignore (don't leak timing information)
return;
}
// Ack over TCP; MediaRelay will set the UDP endpoint when the UDP binding packet arrives.
auto env = make_env(req_id);
env.mutable_udp_binding()->set_ack(true);
send_envelope(env);
}
void ConnSession::handle_stream_announce(uint64_t req_id,
const voicecat::v1::StreamAnnounce& msg) {
uint32_t ssrc = registry_->assign_ssrc(session_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
uint32_t stream_id = next_stream_id_++;
auto env = make_env(req_id);
auto* res = env.mutable_stream_announce_result();
res->set_ok(true);
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
res->set_stream_id(stream_id);
res->set_ssrc(ssrc);
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
// Per-channel AudioConfig is authoritative (docs/voice.md §3): the channel's mode/
// sample_rate/frame_ms/application/fec/dtx/complexity/expected_packet_loss apply to every
// stream announced into it, regardless of kind. bitrate_bps is clamped (not overridden) to
// the channel's ceiling so a client may still request less. sample_rate is taken from the
// channel verbatim (copied via *eff below) — the codec always runs at 48kHz internally, but
// a sub-48k value caps the encoder's audio bandwidth (narrowband/wideband; see
// OpusEncoder::init), so it's a channel policy, not a client choice.
auto* eff = res->mutable_effective_audio();
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
auto chan_cfg = registry_->channel_audio_config(registry_->user_channel(user_id_.load()));
uint32_t requested_bps =
msg.has_requested_audio() ? msg.requested_audio().bitrate_bps() : 0;
if (chan_cfg) {
*eff = *chan_cfg;
eff->set_bitrate_bps(requested_bps > 0 ? std::min(requested_bps, chan_cfg->bitrate_bps())
: chan_cfg->bitrate_bps());
} else if (msg.has_requested_audio()) {
*eff = msg.requested_audio();
} else {
eff->set_codec(0); // OPUS
eff->set_sample_rate(48000);
eff->set_bitrate_bps(24000);
eff->set_frame_ms(20);
eff->set_fec(true);
}
if (eff->sample_rate() == 0) eff->set_sample_rate(48000);
if (eff->bitrate_bps() == 0) eff->set_bitrate_bps(24000);
if (eff->frame_ms() == 0) eff->set_frame_ms(20);
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
announced_stream_ids_.push_back(stream_id);
voicecat::v1::StreamInfo info;
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
info.set_stream_id(stream_id);
info.set_ssrc(ssrc);
info.set_kind(msg.kind());
*info.mutable_audio() = *eff;
info.set_label(msg.label());
send_envelope(env);
auto updated = registry_->set_user_stream(user_id_.load(), info);
if (updated) {
auto bcast = make_env();
auto* ue = bcast.mutable_user_event();
ue->set_kind(voicecat::v1::UserEvent::UPDATED);
*ue->mutable_user() = *updated;
fix(protocol): deliver self-initiated state changes to the actor too A connected Windows client would randomly snap from its joined channel back to Lobby. Root cause was a state-sync inconsistency, not a drop: the server delivered self-initiated state changes (channel join/leave, stream announce/stop) only as a private *Result to the actor and broadcast the authoritative UserEvent::UPDATED to everyone else. The core never applied the result to its SessionModel, so vc_list_users() kept self in the old channel; the Windows HandleUserUpdated rebuilds _currentChannelId from vc_list_users() on any user's UPDATED event, so the next unrelated event surfaced the stale self-channel. Fix, per the response-vs-broadcast contract now documented in docs/protocol.md §6: the *Result is pure ack/correlation/actor-private payload; the resulting state change is broadcast to every client INCLUDING the actor, and clients apply it to their local model rather than re-deriving own state from a *Result. - server: join/leave/stream announce+stop broadcast with exclude=0 - server: text fan-out includes the sender (channel + private echo) - core: response handlers no longer mutate session_model_ - windows: drop optimistic text echo; render own message via the relay - docs/protocol.md §6: document the response-vs-broadcast contract Registry-level admin broadcasts (move/mute/kick/channel CRUD) already used exclude=0 and were correct. ctest build/m1-dev 18/18 green; VoiceCat.App builds 0 warnings. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-17 20:48:50 +02:00
registry_->broadcast(bcast, /*exclude*/ 0); // include announcer; result only acks
}
}
void ConnSession::handle_stream_stop(const voicecat::v1::StreamStop& 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
auto it = std::find(announced_stream_ids_.begin(), announced_stream_ids_.end(),
msg.stream_id());
if (it == announced_stream_ids_.end()) return; // not ours — ignore (no spoofed stops)
announced_stream_ids_.erase(it);
auto updated = registry_->clear_user_stream(user_id_.load(), msg.stream_id());
if (updated) {
auto bcast = make_env();
auto* ue = bcast.mutable_user_event();
ue->set_kind(voicecat::v1::UserEvent::UPDATED);
*ue->mutable_user() = *updated;
fix(protocol): deliver self-initiated state changes to the actor too A connected Windows client would randomly snap from its joined channel back to Lobby. Root cause was a state-sync inconsistency, not a drop: the server delivered self-initiated state changes (channel join/leave, stream announce/stop) only as a private *Result to the actor and broadcast the authoritative UserEvent::UPDATED to everyone else. The core never applied the result to its SessionModel, so vc_list_users() kept self in the old channel; the Windows HandleUserUpdated rebuilds _currentChannelId from vc_list_users() on any user's UPDATED event, so the next unrelated event surfaced the stale self-channel. Fix, per the response-vs-broadcast contract now documented in docs/protocol.md §6: the *Result is pure ack/correlation/actor-private payload; the resulting state change is broadcast to every client INCLUDING the actor, and clients apply it to their local model rather than re-deriving own state from a *Result. - server: join/leave/stream announce+stop broadcast with exclude=0 - server: text fan-out includes the sender (channel + private echo) - core: response handlers no longer mutate session_model_ - windows: drop optimistic text echo; render own message via the relay - docs/protocol.md §6: document the response-vs-broadcast contract Registry-level admin broadcasts (move/mute/kick/channel CRUD) already used exclude=0 and were correct. ctest build/m1-dev 18/18 green; VoiceCat.App builds 0 warnings. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-17 20:48:50 +02:00
registry_->broadcast(bcast, /*exclude*/ 0); // include the stopper
}
}
// ── M5 handlers ──────────────────────────────────────────────────────────────
void ConnSession::handle_kick_request(uint64_t req_id, const voicecat::v1::KickRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_kick)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
bool ok = registry_->kick_user(msg.user_id(), msg.reason());
send_generic_result(req_id, ok, ok ? 0 : 3, ok ? "" : "user not found");
}
void ConnSession::handle_ban_request(uint64_t req_id, const voicecat::v1::BanRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_ban)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
// Ban by username for persistence (nickname == username for password users).
// Also ban by runtime user_id for immediate effect.
if (auto target = registry_->find_session_by_user_id(msg.user_id())) {
if (!target->user_id()) {
send_generic_result(req_id, false, 3, "user not found");
return;
}
if (auto nick = registry_->user_nickname(target->user_id())) {
std::string err;
db_->ban_create("username", *nick, msg.reason(),
static_cast<int64_t>(msg.expires_unix_ms()), err);
}
}
bool ok = registry_->ban_user(msg.user_id(), msg.reason(),
static_cast<int64_t>(msg.expires_unix_ms()));
send_generic_result(req_id, ok, ok ? 0 : 3, ok ? "" : "user not found");
}
void ConnSession::handle_set_permission(uint64_t req_id,
const voicecat::v1::SetPermissionRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_admin_accounts)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
auto target = registry_->find_session_by_user_id(msg.user_id());
if (!target) {
send_generic_result(req_id, false, 3, "user not online");
return;
}
target->set_permissions(msg.permissions());
send_generic_result(req_id, true, 0, "");
}
void ConnSession::handle_server_mute_request(uint64_t req_id,
const voicecat::v1::ServerMuteRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_kick)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
bool ok = registry_->set_server_mute(msg.user_id(), msg.muted(), msg.deafened());
send_generic_result(req_id, ok, ok ? 0 : 3, ok ? "" : "user not found");
}
void ConnSession::handle_move_user(uint64_t req_id, const voicecat::v1::MoveUserRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_move_users)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
bool ok = registry_->move_user(msg.user_id(), msg.channel_id());
send_generic_result(req_id, ok, ok ? 0 : 3, ok ? "" : "user or channel not found");
}
void ConnSession::handle_create_channel(uint64_t req_id,
const voicecat::v1::CreateChannelRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_create_temp_channel)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
std::string error;
uint32_t id = registry_->create_channel(msg.channel(), msg.password(), error);
send_generic_result(req_id, id != 0, id != 0 ? 0 : 3,
id != 0 ? "" : (error.empty() ? "create failed" : error));
}
void ConnSession::handle_edit_channel(uint64_t req_id,
const voicecat::v1::EditChannelRequest& msg) {
if (!is_admin()) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
std::string error;
bool ok = registry_->update_channel(msg.channel(), msg.password(), error);
send_generic_result(req_id, ok, ok ? 0 : 3,
ok ? "" : (error.empty() ? "update failed" : error));
}
void ConnSession::handle_delete_channel(uint64_t req_id,
const voicecat::v1::DeleteChannelRequest& msg) {
if (!is_admin()) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
std::string error;
bool ok = registry_->delete_channel(msg.channel_id(), error);
send_generic_result(req_id, ok, ok ? 0 : 3,
ok ? "" : (error.empty() ? "delete failed" : error));
}
void ConnSession::handle_create_account(uint64_t req_id,
const voicecat::v1::CreateAccountRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_admin_accounts)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
auto self = shared_from_this();
workers_->post([self, req_id, msg]() mutable {
std::string error;
auto acc = self->db_->create_account(msg.username(), msg.password(), false, error);
self->send_generic_result(req_id, acc.has_value(), acc.has_value() ? 0 : 3,
acc.has_value() ? "" : error);
});
}
void ConnSession::handle_reset_password(uint64_t req_id,
const voicecat::v1::ResetPasswordRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_admin_accounts)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
auto self = shared_from_this();
workers_->post([self, req_id, msg]() mutable {
std::string error;
bool ok = self->db_->reset_password(msg.username(), msg.new_password(), error);
self->send_generic_result(req_id, ok, ok ? 0 : 3, ok ? "" : error);
});
}
void ConnSession::handle_delete_account(uint64_t req_id,
const voicecat::v1::DeleteAccountRequest& msg) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_admin_accounts)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
std::string error;
bool ok = db_->delete_account(msg.username(), error);
send_generic_result(req_id, ok, ok ? 0 : 3, ok ? "" : error);
}
void ConnSession::handle_list_accounts(uint64_t req_id,
const voicecat::v1::ListAccountsRequest& /*msg*/) {
if (!is_admin() && !has_permission(&voicecat::v1::Permissions::can_admin_accounts)) {
send_generic_result(req_id, false, 6, "permission denied");
return;
}
auto env = make_env(req_id);
auto* lr = env.mutable_list_accounts_result();
for (const auto& acc : db_->list_accounts()) {
auto* e = lr->add_accounts();
e->set_username(acc.username);
e->set_is_admin(acc.is_admin);
e->set_created_at_unix_ms(static_cast<uint64_t>(acc.created_at) * 1000);
e->set_last_login_unix_ms(static_cast<uint64_t>(acc.last_login) * 1000);
}
send_envelope(env);
}
2026-06-15 23:48:44 +02:00
void ConnSession::send_disconnect_and_close(uint32_t code, const std::string& reason) {
auto env = make_env();
auto* d = env.mutable_disconnect();
d->set_code(code);
d->set_reason(reason);
send_envelope(env);
close();
}
} // namespace voicecat::server
#endif // VOICECAT_HAS_NET