refactor(gateway): remove screen-share / GoLive feature entirely

Drop the Discord Go Live (screen share) stack across the discord-gateway:
- delete src/goLive/ (19 modules: Streamer, Demuxer, encoders, WebRTC wrapper, native loader, etc.)
- delete native/libdatachannel-min/ N-API binding + flake native build + LD_LIBRARY_PATH wiring
- delete screenShareController.ts and screen-share tests (goLive-port, golive-*, demuxerNut, screenShareInput)
- mediaSource.ts: remove Invidious helpers + downloadScreenInput (YouTube full-file download)
- mediaTypes.ts: drop ScreenShare* types, narrow MediaMode to 'music' and DiscordPlayerOwner to non-screen
- media.handler.ts: remove screen branch, screenController/screenPlayback, voice-disconnect/reconnect accessor
- commandHandler.ts: stop passing getVoiceStatus / setVoiceController into MediaHandler
- media handler now only handles music; music queue/playback/status untouched

Verification: tsc --noEmit clean, biome clean on touched files, no lingering goLive/screenShare refs in BE/FE/gateway.
This commit is contained in:
asepharyana
2026-08-15 21:20:20 +07:00
parent 9ae26b8ec9
commit 0164444dd7
42 changed files with 14 additions and 5957 deletions
@@ -1,3 +0,0 @@
node_modules/
build/
package-lock.json
@@ -1,526 +0,0 @@
// libdatachannel-min — minimal N-API binding to libdatachannel.
// Exposes ONLY what GMW GoLive needs:
// PeerConnection (offer/answer, ICE, SDP), DataChannel (signaling),
// Track send (added in media phase).
// Built against libdatachannel 0.24.0 (built from source in /tmp/ldc-build).
#include <napi.h>
#include <rtc/rtc.hpp>
#include <functional>
#include <memory>
#include <string>
#include <variant>
using namespace Napi;
namespace {
std::string stateToString(rtc::PeerConnection::State s) {
switch (s) {
case rtc::PeerConnection::State::New: return "new";
case rtc::PeerConnection::State::Connecting: return "connecting";
case rtc::PeerConnection::State::Connected: return "connected";
case rtc::PeerConnection::State::Disconnected: return "disconnected";
case rtc::PeerConnection::State::Failed: return "failed";
case rtc::PeerConnection::State::Closed: return "closed";
default: return "unknown";
}
}
std::string binaryToString(const rtc::binary& data) {
// rtc::binary is std::vector<std::byte> in libdatachannel >= 0.21
std::string msg(data.size(), '\0');
for (size_t i = 0; i < data.size(); i++) {
msg[i] = static_cast<char>(data[i]);
}
return msg;
}
// Holds a Napi::Promise::Deferred so it can be moved into TSFN lambdas
// without invalid copies (node-addon-api 8.x Deferred is not movable).
struct DeferredHolder {
Promise::Deferred deferred;
explicit DeferredHolder(Promise::Deferred d) : deferred(d) {}
};
class DataChannelWrap : public Napi::ObjectWrap<DataChannelWrap> {
public:
static Function Init(Napi::Env env) {
Function func = DefineClass(env, "DataChannel", {
InstanceMethod("send", &DataChannelWrap::Send),
InstanceMethod("isOpen", &DataChannelWrap::IsOpen),
InstanceMethod("close", &DataChannelWrap::Close),
InstanceMethod("onMessage", &DataChannelWrap::OnMessage),
InstanceMethod("onOpen", &DataChannelWrap::OnOpen),
});
dcConstructor = Napi::Persistent(func);
return func;
}
// Create a JS wrapper (calls the JS constructor, returns instance).
static Object NewInstance(Napi::Env env) {
return dcConstructor.New({});
}
DataChannelWrap(const Napi::CallbackInfo& info)
: Napi::ObjectWrap<DataChannelWrap>(info) {}
void Init(std::shared_ptr<rtc::DataChannel> dc) {
dc_ = dc;
dc_->onMessage([this](rtc::message_variant data) {
std::string msg;
if (std::holds_alternative<rtc::binary>(data)) {
msg = binaryToString(std::get<rtc::binary>(data));
} else {
msg = std::get<std::string>(data);
}
if (msgCb_) {
msgCb_->BlockingCall([msg](Napi::Env env, Function cb) {
cb.Call({String::New(env, msg)});
});
}
});
dc_->onOpen([this]() {
if (openCb_) {
openCb_->BlockingCall([](Napi::Env env, Function cb) {
cb.Call({});
});
}
});
}
private:
static FunctionReference dcConstructor;
std::shared_ptr<rtc::DataChannel> dc_;
std::shared_ptr<ThreadSafeFunction> msgCb_;
std::shared_ptr<ThreadSafeFunction> openCb_;
void Send(const Napi::CallbackInfo& info) {
std::string msg = info[0].As<String>().Utf8Value();
if (dc_) dc_->send(msg);
}
Napi::Value IsOpen(const Napi::CallbackInfo& info) {
bool open = dc_ && dc_->isOpen();
return Boolean::New(info.Env(), open);
}
void Close(const Napi::CallbackInfo& info) {
if (dc_) dc_->close();
}
void OnMessage(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
msgCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "dc-message", 0, 1));
}
void OnOpen(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
openCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "dc-open", 0, 1));
}
};
class TrackWrap : public Napi::ObjectWrap<TrackWrap> {
public:
static Function Init(Napi::Env env) {
Function func = DefineClass(env, "Track", {
InstanceMethod("send", &TrackWrap::Send),
InstanceMethod("isOpen", &TrackWrap::IsOpen),
InstanceMethod("close", &TrackWrap::Close),
InstanceMethod("setPacketizer", &TrackWrap::SetPacketizer),
InstanceMethod("sendFrame", &TrackWrap::SendFrame),
InstanceMethod("addTimestamp", &TrackWrap::AddTimestamp),
});
trackConstructor = Napi::Persistent(func);
return func;
}
static Object NewInstance(Napi::Env env) {
return trackConstructor.New({});
}
TrackWrap(const Napi::CallbackInfo& info)
: Napi::ObjectWrap<TrackWrap>(info) {}
void Init(std::shared_ptr<rtc::Track> track, Napi::Env env) {
track_ = track;
(void)env;
}
private:
static FunctionReference trackConstructor;
std::shared_ptr<rtc::Track> track_;
std::shared_ptr<rtc::RtpPacketizationConfig> rtpConfig_;
void Send(const Napi::CallbackInfo& info) {
Buffer<uint8_t> buf = info[0].As<Buffer<uint8_t>>();
if (!track_) return;
rtc::binary data(buf.Length());
for (size_t i = 0; i < buf.Length(); i++) data[i] = (std::byte)buf[i];
try {
track_->send(data);
} catch (const std::exception& e) {
fprintf(stderr, "[binding] track.send THREW: %s\n", e.what());
}
}
// setPacketizer(kind, ssrc, payloadType, clockRate, playoutDelayId,
// playoutDelayMin, playoutDelayMax)
// kind: "audio" | "h264" | "h265" | "av1"
// Builds the media-handler chain (packetizer → RTCP SR → NACK → pacing for
// video) exactly like @dank074's WebRtcWrapper does via node-datachannel.
void SetPacketizer(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
if (!track_) throw Error::New(env, "track closed");
std::string kind = info[0].As<String>().Utf8Value();
uint32_t ssrc = info[1].As<Number>().Uint32Value();
uint8_t pt = (uint8_t)info[2].As<Number>().Uint32Value();
uint32_t clockRate = info[3].As<Number>().Uint32Value();
uint8_t playoutDelayId = (uint8_t)info[4].As<Number>().Uint32Value();
uint16_t playoutDelayMin = (uint16_t)info[5].As<Number>().Uint32Value();
uint16_t playoutDelayMax = (uint16_t)info[6].As<Number>().Uint32Value();
try {
auto cfg = std::make_shared<rtc::RtpPacketizationConfig>(
ssrc, "", pt, clockRate);
cfg->playoutDelayId = playoutDelayId;
cfg->playoutDelayMin = playoutDelayMin;
cfg->playoutDelayMax = playoutDelayMax;
std::shared_ptr<rtc::MediaHandler> handler;
if (kind == "audio") {
handler = std::make_shared<rtc::OpusRtpPacketizer>(cfg);
} else if (kind == "h264") {
handler = std::make_shared<rtc::H264RtpPacketizer>(
rtc::NalUnit::Separator::StartSequence, cfg);
} else if (kind == "h265") {
handler = std::make_shared<rtc::H265RtpPacketizer>(
rtc::NalUnit::Separator::StartSequence, cfg);
} else if (kind == "av1") {
handler = std::make_shared<rtc::AV1RtpPacketizer>(
rtc::AV1RtpPacketizer::Packetization::Obu, cfg);
} else {
throw std::runtime_error("unknown packetizer kind: " + kind);
}
handler->addToChain(std::make_shared<rtc::RtcpSrReporter>(cfg));
handler->addToChain(std::make_shared<rtc::RtcpNackResponder>());
if (kind != "audio") {
handler->addToChain(std::make_shared<rtc::PacingHandler>(
25.0 * 1000 * 1000, std::chrono::milliseconds(1)));
}
track_->setMediaHandler(handler);
rtpConfig_ = cfg;
} catch (const std::exception& e) {
fprintf(stderr, "[binding] setPacketizer THREW: %s\n", e.what());
throw Error::New(env, e.what());
}
}
// sendFrame(buffer) — sends an ENCODED frame (AnnexB H264 / raw opus /
// OBU AV1). The media-handler chain packetizes it into RTP.
void SendFrame(const Napi::CallbackInfo& info) {
Buffer<uint8_t> buf = info[0].As<Buffer<uint8_t>>();
if (!track_) return;
rtc::binary data(buf.Length());
for (size_t i = 0; i < buf.Length(); i++) data[i] = (std::byte)buf[i];
try {
track_->send(data);
} catch (const std::exception& e) {
fprintf(stderr, "[binding] track.sendFrame THREW: %s\n", e.what());
}
}
// addTimestamp(delta) — advances the packetizer RTP timestamp by delta
// (clock-rate units). Called by JS after each frame, matching the
// node-datachannel contract (WebRtcWrapper does the same increment).
void AddTimestamp(const Napi::CallbackInfo& info) {
uint32_t delta = info[0].As<Number>().Uint32Value();
if (rtpConfig_) rtpConfig_->timestamp += delta;
}
Napi::Value IsOpen(const Napi::CallbackInfo& info) {
bool open = track_ && track_->isOpen();
return Boolean::New(info.Env(), open);
}
void Close(const Napi::CallbackInfo& info) {
if (track_) track_->close();
}
void OnStateChange(const Napi::CallbackInfo& info) {
// libdatachannel Track has no state-change callback; kept for API parity.
(void)info;
}
};
class PeerConnectionWrap : public Napi::ObjectWrap<PeerConnectionWrap> {
public:
static Function Init(Napi::Env env) {
Function func = DefineClass(env, "PeerConnection", {
InstanceMethod("state", &PeerConnectionWrap::State),
InstanceMethod("createOffer", &PeerConnectionWrap::CreateOffer),
InstanceMethod("createAnswer", &PeerConnectionWrap::CreateAnswer),
InstanceMethod("setRemoteDescription",
&PeerConnectionWrap::SetRemoteDescription),
InstanceMethod("close", &PeerConnectionWrap::Close),
InstanceMethod("onStateChange", &PeerConnectionWrap::OnStateChange),
InstanceMethod("createDataChannel", &PeerConnectionWrap::CreateDataChannel),
InstanceMethod("onDataChannel", &PeerConnectionWrap::OnDataChannel),
InstanceMethod("addTrack", &PeerConnectionWrap::AddTrack),
});
return func;
}
PeerConnectionWrap(const Napi::CallbackInfo& info)
: Napi::ObjectWrap<PeerConnectionWrap>(info) {
Napi::Env env = info.Env();
if (!info[0].IsObject()) {
throw TypeError::New(env, "config object required");
}
Object config = info[0].As<Object>();
rtc::Configuration rtcConfig;
if (config.Has("iceServers")) {
Array servers = config.Get("iceServers").As<Array>();
for (uint32_t i = 0; i < servers.Length(); i++) {
std::string url = servers.Get(i).As<String>().Utf8Value();
rtcConfig.iceServers.emplace_back(url);
}
}
pc_ = std::make_shared<rtc::PeerConnection>(rtcConfig);
// IMPORTANT: register description/gathering callbacks HERE (constructor),
// BEFORE any createDataChannel call. libdatachannel only fires
// onLocalDescription for negotiations that start AFTER the callback is
// registered — if createDataChannel runs first, the offer callback never
// fires (verified in C++ spike: test3 vs test2).
pc_->onLocalDescription([this](rtc::Description desc) {
latestLocalDesc_ = std::string(desc);
fprintf(stderr, "[binding] trickle desc, %zu bytes\n",
latestLocalDesc_.size());
});
pc_->onGatheringStateChange([this](rtc::PeerConnection::GatheringState gs) {
fprintf(stderr, "[binding] gathering state: %d\n", (int)gs);
if (gs == rtc::PeerConnection::GatheringState::Complete) {
// Use the getter — it returns the FULL SDP including candidates after
// gathering (trickle callbacks only carry the initial fragment).
auto ld = pc_->localDescription();
if (ld) {
latestLocalDesc_ = std::string(*ld);
fprintf(stderr, "[binding] final desc, %zu bytes\n",
latestLocalDesc_.size());
}
resolvePendingLocalDesc_();
}
});
}
private:
std::shared_ptr<rtc::PeerConnection> pc_;
std::shared_ptr<ThreadSafeFunction> stateCb_;
std::shared_ptr<ThreadSafeFunction> dcCb_;
std::string latestLocalDesc_;
std::shared_ptr<DeferredHolder> pendingDescDeferred_;
std::shared_ptr<ThreadSafeFunction> pendingDescTsfn_;
void resolvePendingLocalDesc_() {
if (!pendingDescDeferred_ || !pendingDescTsfn_) return;
auto holder = pendingDescDeferred_;
auto tsfn = pendingDescTsfn_;
pendingDescDeferred_.reset();
pendingDescTsfn_.reset();
std::string sdp = latestLocalDesc_;
tsfn->BlockingCall([sdp, holder](Napi::Env e, Function) {
holder->deferred.Resolve(String::New(e, sdp));
});
}
Napi::Value State(const Napi::CallbackInfo& info) {
return String::New(info.Env(),
pc_ ? stateToString(pc_->state()) : "closed");
}
// createOffer() -> Promise<string> — sets local description, waits for
// ICE gathering to complete (so candidates are in the SDP), resolves SDP.
Napi::Value CreateOffer(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
auto holder = std::make_shared<DeferredHolder>(Promise::Deferred::New(env));
if (!pc_) {
holder->deferred.Reject(Error::New(env, "peer closed").Value());
return holder->deferred.Promise();
}
// createDataChannel already triggers negotiation in libdatachannel 0.24 —
// if gathering already completed, resolve immediately from the cached SDP.
if (!latestLocalDesc_.empty()) {
auto tsfn = std::make_shared<ThreadSafeFunction>(ThreadSafeFunction::New(
env, Function::New(env, [](const CallbackInfo&) {}), "desc", 0, 1));
std::string sdp = latestLocalDesc_;
tsfn->BlockingCall([sdp, holder](Napi::Env e, Function) {
holder->deferred.Resolve(String::New(e, sdp));
});
return holder->deferred.Promise();
}
if (pendingDescDeferred_) {
pendingDescDeferred_->deferred.Reject(
Error::New(env, "previous negotiation still pending").Value());
}
pendingDescDeferred_ = holder;
pendingDescTsfn_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(env, Function::New(env, [](const CallbackInfo&) {}),
"desc", 0, 1));
fprintf(stderr, "[binding] calling setLocalDescription(Offer)\n");
try {
pc_->setLocalDescription(rtc::Description::Type::Offer);
fprintf(stderr, "[binding] setLocalDescription returned OK\n");
} catch (const std::exception& e) {
pendingDescDeferred_.reset();
fprintf(stderr, "[binding] setLocalDescription THREW: %s\n", e.what());
throw Error::New(env, e.what());
}
return holder->deferred.Promise();
}
// createAnswer(offerSdp: string) -> Promise<string>
Napi::Value CreateAnswer(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
std::string offer = info[0].As<String>().Utf8Value();
auto holder = std::make_shared<DeferredHolder>(Promise::Deferred::New(env));
if (!pc_) {
holder->deferred.Reject(Error::New(env, "peer closed").Value());
return holder->deferred.Promise();
}
if (pendingDescDeferred_) {
pendingDescDeferred_->deferred.Reject(
Error::New(env, "previous negotiation still pending").Value());
}
pendingDescDeferred_ = holder;
pendingDescTsfn_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(env, Function::New(env, [](const CallbackInfo&) {}),
"desc", 0, 1));
try {
pc_->setRemoteDescription(
rtc::Description(offer, rtc::Description::Type::Offer));
fprintf(stderr, "[binding] answer: setRemoteDescription OK\n");
// libdatachannel 0.24 AUTO-GENERATES the answer when a remote offer is
// applied (verified in C++ spike test8/9: B desc type=Answer fires
// immediately with a=setup:active). Calling setLocalDescription() again
// would OVERWRITE it with a role=actpass SDP, which A rejects with
// "Illegal role actpass in remote answer description". So we do NOT call
// setLocalDescription here — we just wait for gathering complete and
// resolve with the auto-generated answer. This also matches @dank074's
// Discord voice flow.
} catch (const std::exception& e) {
pendingDescDeferred_.reset();
fprintf(stderr, "[binding] answer THREW: %s\n", e.what());
holder->deferred.Reject(Error::New(env, e.what()).Value());
}
return holder->deferred.Promise();
}
void SetRemoteDescription(const Napi::CallbackInfo& info) {
std::string sdp = info[0].As<String>().Utf8Value();
std::string type = info[1].As<String>().Utf8Value();
rtc::Description::Type t = (type == "answer")
? rtc::Description::Type::Answer
: rtc::Description::Type::Offer;
if (pc_) pc_->setRemoteDescription(rtc::Description(sdp, t));
}
void Close(const Napi::CallbackInfo& info) {
if (pc_) pc_->close();
}
void OnStateChange(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
stateCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "pc-state", 0, 1));
std::shared_ptr<rtc::PeerConnection> pc = pc_;
pc->onStateChange([this](rtc::PeerConnection::State state) {
if (stateCb_) {
std::string s = stateToString(state);
stateCb_->BlockingCall([s](Napi::Env env, Function cb) {
cb.Call({String::New(env, s)});
});
}
});
}
Napi::Value CreateDataChannel(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
std::string label = info[0].As<String>().Utf8Value();
fprintf(stderr, "[binding] createDataChannel(%s)\n", label.c_str());
auto dc = pc_->createDataChannel(label);
Object obj = DataChannelWrap::NewInstance(env);
DataChannelWrap::Unwrap(obj)->Init(dc);
return obj;
}
Napi::Value AddTrack(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
std::string mid = info[0].As<String>().Utf8Value();
std::string kind = info[1].As<String>().Utf8Value();
if (!pc_) throw Error::New(env, "peer closed");
fprintf(stderr, "[binding] addTrack(%s, %s) start\n", mid.c_str(), kind.c_str());
try {
std::shared_ptr<rtc::Track> track;
if (kind == "audio") {
// Opus payload type 120 (matches @dank074 CodecPayloadType.opus)
auto desc = rtc::Description::Audio(mid);
desc.addOpusCodec(120);
track = pc_->addTrack(desc);
} else {
// All video codecs with their payload types, matching WebRtcWrapper:
// H264 101/102, H265 103/104, VP8 105/106, VP9 107/108, AV1 109/110
auto desc = rtc::Description::Video(mid);
desc.addH264Codec(101);
desc.addRtxCodec(102, 101, 90000);
desc.addH265Codec(103);
desc.addRtxCodec(104, 103, 90000);
desc.addVP8Codec(105);
desc.addRtxCodec(106, 105, 90000);
desc.addVP9Codec(107);
desc.addRtxCodec(108, 107, 90000);
desc.addAV1Codec(109);
desc.addRtxCodec(110, 109, 90000);
track = pc_->addTrack(desc);
}
Object obj = TrackWrap::NewInstance(env);
TrackWrap::Unwrap(obj)->Init(track, env);
return obj;
} catch (const std::exception& e) {
fprintf(stderr, "[binding] addTrack THREW: %s\n", e.what());
throw Error::New(env, e.what());
}
}
void OnDataChannel(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
dcCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "dc", 0, 1));
std::shared_ptr<rtc::PeerConnection> pc = pc_;
pc->onDataChannel([this](std::shared_ptr<rtc::DataChannel> dc) {
if (dcCb_) {
auto dcPtr = dc;
dcCb_->BlockingCall([dcPtr](Napi::Env env, Function cb) {
Object obj = DataChannelWrap::NewInstance(env);
DataChannelWrap::Unwrap(obj)->Init(dcPtr);
cb.Call({obj});
});
}
});
}
};
Object InitAll(Napi::Env env, Object exports) {
exports.Set("PeerConnection", PeerConnectionWrap::Init(env));
exports.Set("DataChannel", DataChannelWrap::Init(env));
exports.Set("Track", TrackWrap::Init(env));
return exports;
}
NODE_API_MODULE(libdatachannel_min, InitAll)
// Definition for the static constructor references.
FunctionReference DataChannelWrap::dcConstructor;
FunctionReference TrackWrap::trackConstructor;
} // namespace
@@ -1,21 +0,0 @@
{
"targets": [
{
"target_name": "libdatachannel_min",
"sources": ["binding.cpp"],
"include_dirs": [
"<!(node -e \"console.log(process.env.NAPI_INCLUDE || (() => { try { return require('node-addon-api').include; } catch { return '/nonexistent'; } })())\")",
"<!(node -e \"const s=process.env.LDC_INCLUDE||'/nix/store/39a85gpfjqy3h3k8jwrwh7m9yc3inqw7-source';console.log(s+'/include')\")"
],
"libraries": [
"<!(node -e \"console.log(process.env.LDC_LIB || '/tmp/ldc-build/libdatachannel.so.0.24.0')\")"
],
"cflags": ["-std=c++17", "-fexceptions"],
"cflags_cc": ["-std=c++17", "-fexceptions"],
"defines": ["NAPI_CPP_EXCEPTIONS"],
"conditions": [
["OS=='linux'", { "cflags": ["-fvisibility=hidden"] }]
]
}
]
}
@@ -1,3 +0,0 @@
// libdatachannel-min — JS entry.
const native = require("./build/Release/datachannel_min.node");
module.exports = native;
@@ -1,17 +0,0 @@
{
"name": "libdatachannel-min",
"version": "0.1.0",
"description": "Minimal N-API binding to libdatachannel — PeerConnection, DataChannel, ICE, SDP (+ media tracks for GoLive)",
"main": "index.js",
"gypfile": true,
"scripts": {
"build": "node-gyp rebuild",
"test": "node test-handshake.js"
},
"dependencies": {
"node-addon-api": "^8.3.0"
},
"devDependencies": {
"node-gyp": "^11.5.0"
}
}
@@ -1,82 +0,0 @@
// Phase 0 spike: prove the minimal binding can do a full WebRTC handshake
// (offer/answer + ICE + DataChannel) between two local PeerConnections.
"use strict";
const { PeerConnection } = require("./build/Release/datachannel_min.node");
function log(...args) {
console.log("[spike]", ...args);
}
async function main() {
const pcA = new PeerConnection({ iceServers: [] });
const pcB = new PeerConnection({ iceServers: [] });
const stateLog = [];
pcA.onStateChange((s) => {
stateLog.push(`A:${s}`);
log("A state:", s);
});
pcB.onStateChange((s) => {
stateLog.push(`B:${s}`);
log("B state:", s);
});
// B waits for incoming DataChannel
const received = new Promise((resolve) => {
pcB.onDataChannel((dc) => {
log("B got incoming DataChannel");
dc.onOpen(() => log("B DataChannel open"));
dc.onMessage((msg) => {
log("B received message:", msg);
dc.send("pong from B");
resolve(msg);
});
});
});
// A creates an outgoing DataChannel
const dcA = pcA.createDataChannel("test");
dcA.onOpen(() => {
log("A DataChannel open — sending hello");
dcA.send("hello from A");
});
dcA.onMessage((msg) => {
log("A received reply:", msg);
});
// Offer/answer dance
log("A createOffer...");
const offer = await pcA.createOffer();
log("Offer SDP bytes:", offer.length);
log("B createAnswer...");
const answer = await pcB.createAnswer(offer);
log("Answer SDP bytes:", answer.length);
const setupMatch = answer.match(/a=setup:(\S+)/);
log("Answer setup role:", setupMatch ? setupMatch[1] : "NONE");
pcA.setRemoteDescription(answer, "answer");
// Wait for message roundtrip
const msg = await Promise.race([
received,
new Promise((_, rej) => setTimeout(() => rej(new Error("TIMEOUT waiting for datachannel message")), 15000)),
]);
log("ROUNDTRIP OK — B got:", msg);
log("States:", stateLog.join(" | "));
const aState = pcA.state();
const bState = pcB.state();
log("Final states — A:", aState, "B:", bState);
pcA.close();
pcB.close();
if (msg !== "hello from A") throw new Error("wrong message");
if (aState !== "connected" && aState !== "disconnected") throw new Error("A not connected: " + aState);
log("SPIKE PASSED ✅");
}
main().catch((e) => {
console.error("SPIKE FAILED:", e.message);
process.exit(1);
});
@@ -1,80 +0,0 @@
// Verify setPacketizer + sendFrame: two peers connect, audio+video tracks
// packetize real encoded frames (opus + AnnexB H264), RTP flows without crash.
"use strict";
const { PeerConnection } = require("./build/Release/datachannel_min.node");
function sleep(ms) { return new Promise((r) => setTimeout(r, ms)); }
async function main() {
const pcA = new PeerConnection({ iceServers: [] });
const pcB = new PeerConnection({ iceServers: [] });
const aAudio = pcA.addTrack("0", "audio");
const aVideo = pcA.addTrack("1", "video");
pcB.addTrack("0", "audio");
pcB.addTrack("1", "video");
let states = { a: "", b: "" };
pcA.onStateChange((s) => (states.a = s));
pcB.onStateChange((s) => (states.b = s));
// A: offer (createDataChannel not needed — tracks trigger negotiation)
const offer = await pcA.createOffer();
pcB.setRemoteDescription(offer, "offer");
const answer = await pcB.createAnswer(offer);
pcA.setRemoteDescription(answer, "answer");
// Wait for connected
for (let i = 0; i < 50; i++) {
if (states.a === "connected" && states.b === "connected") break;
await sleep(100);
}
console.log("[pkt] states:", states.a, states.b);
if (states.a !== "connected" || states.b !== "connected") {
console.log("PKT TEST FAILED: not connected");
process.exit(1);
}
// Setup packetizers on A (sender)
aAudio.setPacketizer("audio", 1234, 120, 48000, 5, 0, 1);
aVideo.setPacketizer("h264", 5678, 101, 90000, 5, 0, 10);
// Fake opus frame (20ms @48kHz stereo — payload can be any bytes)
const opusFrame = Buffer.alloc(160);
for (let i = 0; i < 160; i++) opusFrame[i] = i & 0xff;
// Fake AnnexB H264 frame: SPS + PPS + IDR slice
const sps = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x67, 0x42, 0xc0, 0x1e, 0xd9, 0x01, 0x40, 0x7e]);
const pps = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x68, 0xce, 0x3c, 0x80]);
const idr = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x65, 0x88, 0x84, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07]);
const h264Frame = Buffer.concat([sps, pps, idr]);
// Send 10 audio frames (20ms each) + 3 video frames (33ms each)
for (let i = 0; i < 10; i++) {
aAudio.sendFrame(opusFrame);
aAudio.addTimestamp(960); // 20ms @ 48kHz
}
for (let i = 0; i < 3; i++) {
aVideo.sendFrame(h264Frame);
aVideo.addTimestamp(3000); // 33ms @ 90kHz
}
await sleep(500);
console.log("[pkt] after send: states:", states.a, states.b);
console.log("[pkt] audio track open:", aAudio.isOpen(), "| video track open:", aVideo.isOpen());
const ok = states.a === "connected" && aAudio.isOpen() && aVideo.isOpen();
console.log(ok ? "PKT TEST PASSED" : "PKT TEST FAILED");
pcA.close();
pcB.close();
process.exit(ok ? 0 : 1);
}
main().catch((e) => {
console.error("[pkt] FAILED:", e.message);
process.exit(1);
});
setTimeout(() => {
console.error("[pkt] TIMEOUT");
process.exit(1);
}, 25000);
@@ -1,138 +0,0 @@
// Two-peer RTP capture test: A sends REAL H264 frames through the binding's
// packetizer chain to B over localhost. tcpdump (run externally on lo) captures
// the RTP; a Python script reassembles AnnexB and ffmpeg decodes it.
//
// Usage:
// node test-rtp-capture.js <mode> mode = "a" (sender) | "b" (receiver)
// Sender writes the negotiated SDP pieces to /tmp/rtp-a.sdp /tmp/rtp-b.sdp
// Receiver listens and keeps alive.
"use strict";
const { PeerConnection } = require("./build/Release/datachannel_min.node");
const fs = require("fs");
const mode = process.argv[2] || "a";
const sleep = (ms) => new Promise((r) => setTimeout(r, ms));
async function main() {
const pc = new PeerConnection({ iceServers: [] });
const audio = pc.addTrack("0", "audio");
const video = pc.addTrack("1", "video");
if (mode === "a") {
// Sender: generate offer, hand to B via files, get B's answer
const offer = await pc.createOffer();
fs.writeFileSync("/tmp/rtp-offer.sdp", offer);
console.log("[a] offer written", offer.length, "bytes");
// wait for B to write its answer
for (let i = 0; i < 300; i++) {
if (fs.existsSync("/tmp/rtp-answer.sdp")) break;
await sleep(200);
}
const answer = fs.readFileSync("/tmp/rtp-answer.sdp", "utf8");
pc.setRemoteDescription(answer, "answer");
// wait connected
for (let i = 0; i < 50; i++) {
if (pc.state() === "connected") break;
await sleep(100);
}
console.log("[a] state:", pc.state());
if (pc.state() !== "connected") {
console.log("[a] FAILED not connected");
process.exit(1);
}
// Setup packetizer like the gateway does
video.setPacketizer("h264", 5678, 101, 90000, 5, 0, 10);
// Real H264 AnnexB (baseline) — read from file created by ffmpeg
const data = fs.readFileSync("/tmp/rtp-input.h264");
console.log("[a] input h264 bytes:", data.length);
// Split into NAL units by start codes, then group into access units
// the same way Demuxer does (param sets + one slice per frame).
const start3 = Buffer.from([0, 0, 1]);
const start4 = Buffer.from([0, 0, 0, 1]);
const nals = [];
let i = 0;
while (i < data.length) {
let start = -1;
let startLen = 0;
for (let j = i; j < data.length - 3; j++) {
if (data[j] === 0 && data[j + 1] === 0 && data[j + 2] === 1) {
start = j;
startLen = 3;
if (j > 0 && data[j - 1] === 0) {
start = j - 1;
startLen = 4;
}
break;
}
}
if (start === -1) break;
if (start > i) {
nals.push(data.subarray(i, start));
}
i = start + startLen;
}
console.log("[a] NALs:", nals.length);
// Group: buffer param sets, flush on slice (like Demuxer.flushAccessUnit)
let pending = [];
let frameCount = 0;
const flush = () => {
if (pending.length === 0) return;
const parts = pending.map((n) => Buffer.concat([start4, n]));
const au = Buffer.concat(parts);
pending = [];
video.sendFrame(au);
video.addTimestamp(3000); // 30fps @ 90kHz
frameCount++;
};
for (const n of nals) {
const t = n[0] & 0x1f;
if (t === 1 || t === 5) {
flush(); // previous AU closed by this slice
pending.push(n);
} else {
pending.push(n); // param set / SEI
}
}
flush();
console.log("[a] sent frames:", frameCount);
await sleep(3000); // let packets flow
console.log("[a] done");
pc.close();
process.exit(0);
} else {
// Receiver: read offer, answer, keep alive
for (let i = 0; i < 300; i++) {
if (fs.existsSync("/tmp/rtp-offer.sdp")) break;
await sleep(200);
}
const offer = fs.readFileSync("/tmp/rtp-offer.sdp", "utf8");
pc.setRemoteDescription(offer, "offer");
const answer = await pc.createAnswer(offer);
fs.writeFileSync("/tmp/rtp-answer.sdp", answer);
console.log("[b] answer written");
for (let i = 0; i < 50; i++) {
if (pc.state() === "connected") break;
await sleep(100);
}
console.log("[b] state:", pc.state());
await sleep(10000); // hold while sender streams
console.log("[b] done");
pc.close();
process.exit(0);
}
}
main().catch((e) => {
console.error("FAILED:", e.message);
process.exit(1);
});
setTimeout(() => {
console.error("TIMEOUT");
process.exit(1);
}, 30000);
@@ -1,33 +0,0 @@
// Verify addTrack produces SDP with audio+video media sections.
"use strict";
const { PeerConnection } = require("./build/Release/datachannel_min.node");
const pc = new PeerConnection({ iceServers: [] });
const audioTrack = pc.addTrack("0", "audio");
const videoTrack = pc.addTrack("1", "video");
pc.onStateChange((s) => console.log("[test-track] state:", s));
pc.createOffer().then((sdp) => {
const hasAudio = /^m=audio\s/m.test(sdp);
const hasVideo = /^m=video\s/m.test(sdp);
const audioPts = sdp.match(/a=rtpmap:(\d+) opus/g) || [];
const videoPts = sdp.match(/a=rtpmap:(\d+) H264/g) || [];
console.log("[test-track] SDP bytes:", sdp.length);
console.log("[test-track] m=audio:", hasAudio, "| m=video:", hasVideo);
console.log("[test-track] opus pt:", audioPts, "| H264 pt:", videoPts);
console.log("[test-track] audio track send ok:", typeof audioTrack.send === "function");
console.log("[test-track] video track send ok:", typeof videoTrack.send === "function");
const ok = hasAudio && hasVideo && audioPts.length > 0 && videoPts.length > 0;
console.log(ok ? "TRACK TEST PASSED" : "TRACK TEST FAILED");
pc.close();
process.exit(ok ? 0 : 1);
}).catch((e) => {
console.error("[test-track] FAILED:", e.message);
process.exit(1);
});
setTimeout(() => {
console.error("[test-track] TIMEOUT");
process.exit(1);
}, 20000);