spike: expose libdatachannel media packetizer chain via Track
Track.setPacketizer(kind, ssrc, pt, clockRate, ...) builds the same media-handler chain node-datachannel does for @dank074: RtpPacketizer (Opus | H264 | H265 | AV1) → RtcpSrReporter → RtcpNackResponder → PacingHandler(25Mbps, 1ms) for video Track.sendFrame(encodedFrame) packetizes into RTP; addTimestamp(delta) advances the RTP timestamp (node-datachannel contract). Verified test-packetizer.js: two peers connected over tracks, real opus frames + AnnexB H264 (SPS/PPS/IDR) flow through the chain without crash. This removes the need for a JS RTP packetizer entirely — libdatachannel 0.24 has the full media stack built in.
This commit is contained in:
asepharyana
@@ -130,7 +130,9 @@ class TrackWrap : public Napi::ObjectWrap<TrackWrap> {
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InstanceMethod("send", &TrackWrap::Send),
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InstanceMethod("isOpen", &TrackWrap::IsOpen),
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InstanceMethod("close", &TrackWrap::Close),
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InstanceMethod("onStateChange", &TrackWrap::OnStateChange),
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InstanceMethod("setPacketizer", &TrackWrap::SetPacketizer),
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InstanceMethod("sendFrame", &TrackWrap::SendFrame),
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InstanceMethod("addTimestamp", &TrackWrap::AddTimestamp),
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});
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trackConstructor = Napi::Persistent(func);
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return func;
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@@ -151,6 +153,7 @@ class TrackWrap : public Napi::ObjectWrap<TrackWrap> {
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private:
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static FunctionReference trackConstructor;
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std::shared_ptr<rtc::Track> track_;
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std::shared_ptr<rtc::RtpPacketizationConfig> rtpConfig_;
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void Send(const Napi::CallbackInfo& info) {
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Buffer<uint8_t> buf = info[0].As<Buffer<uint8_t>>();
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@@ -164,6 +167,78 @@ class TrackWrap : public Napi::ObjectWrap<TrackWrap> {
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}
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}
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// setPacketizer(kind, ssrc, payloadType, clockRate, playoutDelayId,
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// playoutDelayMin, playoutDelayMax)
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// kind: "audio" | "h264" | "h265" | "av1"
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// Builds the media-handler chain (packetizer → RTCP SR → NACK → pacing for
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// video) exactly like @dank074's WebRtcWrapper does via node-datachannel.
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void SetPacketizer(const Napi::CallbackInfo& info) {
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Napi::Env env = info.Env();
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if (!track_) throw Error::New(env, "track closed");
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std::string kind = info[0].As<String>().Utf8Value();
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uint32_t ssrc = info[1].As<Number>().Uint32Value();
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uint8_t pt = (uint8_t)info[2].As<Number>().Uint32Value();
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uint32_t clockRate = info[3].As<Number>().Uint32Value();
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uint8_t playoutDelayId = (uint8_t)info[4].As<Number>().Uint32Value();
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uint16_t playoutDelayMin = (uint16_t)info[5].As<Number>().Uint32Value();
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uint16_t playoutDelayMax = (uint16_t)info[6].As<Number>().Uint32Value();
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try {
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auto cfg = std::make_shared<rtc::RtpPacketizationConfig>(
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ssrc, "", pt, clockRate);
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cfg->playoutDelayId = playoutDelayId;
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cfg->playoutDelayMin = playoutDelayMin;
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cfg->playoutDelayMax = playoutDelayMax;
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std::shared_ptr<rtc::MediaHandler> handler;
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if (kind == "audio") {
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handler = std::make_shared<rtc::OpusRtpPacketizer>(cfg);
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} else if (kind == "h264") {
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handler = std::make_shared<rtc::H264RtpPacketizer>(
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rtc::NalUnit::Separator::StartSequence, cfg);
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} else if (kind == "h265") {
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handler = std::make_shared<rtc::H265RtpPacketizer>(
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rtc::NalUnit::Separator::StartSequence, cfg);
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} else if (kind == "av1") {
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handler = std::make_shared<rtc::AV1RtpPacketizer>(
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rtc::AV1RtpPacketizer::Packetization::Obu, cfg);
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} else {
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throw std::runtime_error("unknown packetizer kind: " + kind);
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}
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handler->addToChain(std::make_shared<rtc::RtcpSrReporter>(cfg));
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handler->addToChain(std::make_shared<rtc::RtcpNackResponder>());
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if (kind != "audio") {
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handler->addToChain(std::make_shared<rtc::PacingHandler>(
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25.0 * 1000 * 1000, std::chrono::milliseconds(1)));
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}
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track_->setMediaHandler(handler);
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rtpConfig_ = cfg;
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} catch (const std::exception& e) {
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fprintf(stderr, "[binding] setPacketizer THREW: %s\n", e.what());
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throw Error::New(env, e.what());
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}
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}
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// sendFrame(buffer) — sends an ENCODED frame (AnnexB H264 / raw opus /
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// OBU AV1). The media-handler chain packetizes it into RTP.
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void SendFrame(const Napi::CallbackInfo& info) {
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Buffer<uint8_t> buf = info[0].As<Buffer<uint8_t>>();
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if (!track_) return;
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rtc::binary data(buf.Length());
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for (size_t i = 0; i < buf.Length(); i++) data[i] = (std::byte)buf[i];
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try {
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track_->send(data);
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} catch (const std::exception& e) {
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fprintf(stderr, "[binding] track.sendFrame THREW: %s\n", e.what());
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}
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}
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// addTimestamp(delta) — advances the packetizer RTP timestamp by delta
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// (clock-rate units). Called by JS after each frame, matching the
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// node-datachannel contract (WebRtcWrapper does the same increment).
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void AddTimestamp(const Napi::CallbackInfo& info) {
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uint32_t delta = info[0].As<Number>().Uint32Value();
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if (rtpConfig_) rtpConfig_->timestamp += delta;
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}
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Napi::Value IsOpen(const Napi::CallbackInfo& info) {
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bool open = track_ && track_->isOpen();
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return Boolean::New(info.Env(), open);
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@@ -178,7 +253,6 @@ class TrackWrap : public Napi::ObjectWrap<TrackWrap> {
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(void)info;
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}
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};
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class PeerConnectionWrap : public Napi::ObjectWrap<PeerConnectionWrap> {
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public:
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static Function Init(Napi::Env env) {
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@@ -0,0 +1,80 @@
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// Verify setPacketizer + sendFrame: two peers connect, audio+video tracks
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// packetize real encoded frames (opus + AnnexB H264), RTP flows without crash.
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"use strict";
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const { PeerConnection } = require("./build/Release/datachannel_min.node");
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function sleep(ms) { return new Promise((r) => setTimeout(r, ms)); }
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async function main() {
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const pcA = new PeerConnection({ iceServers: [] });
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const pcB = new PeerConnection({ iceServers: [] });
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const aAudio = pcA.addTrack("0", "audio");
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const aVideo = pcA.addTrack("1", "video");
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pcB.addTrack("0", "audio");
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pcB.addTrack("1", "video");
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let states = { a: "", b: "" };
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pcA.onStateChange((s) => (states.a = s));
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pcB.onStateChange((s) => (states.b = s));
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// A: offer (createDataChannel not needed — tracks trigger negotiation)
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const offer = await pcA.createOffer();
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pcB.setRemoteDescription(offer, "offer");
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const answer = await pcB.createAnswer(offer);
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pcA.setRemoteDescription(answer, "answer");
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// Wait for connected
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for (let i = 0; i < 50; i++) {
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if (states.a === "connected" && states.b === "connected") break;
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await sleep(100);
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}
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console.log("[pkt] states:", states.a, states.b);
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if (states.a !== "connected" || states.b !== "connected") {
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console.log("PKT TEST FAILED: not connected");
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process.exit(1);
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}
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// Setup packetizers on A (sender)
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aAudio.setPacketizer("audio", 1234, 120, 48000, 5, 0, 1);
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aVideo.setPacketizer("h264", 5678, 101, 90000, 5, 0, 10);
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// Fake opus frame (20ms @48kHz stereo — payload can be any bytes)
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const opusFrame = Buffer.alloc(160);
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for (let i = 0; i < 160; i++) opusFrame[i] = i & 0xff;
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// Fake AnnexB H264 frame: SPS + PPS + IDR slice
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const sps = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x67, 0x42, 0xc0, 0x1e, 0xd9, 0x01, 0x40, 0x7e]);
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const pps = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x68, 0xce, 0x3c, 0x80]);
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const idr = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x65, 0x88, 0x84, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07]);
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const h264Frame = Buffer.concat([sps, pps, idr]);
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// Send 10 audio frames (20ms each) + 3 video frames (33ms each)
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for (let i = 0; i < 10; i++) {
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aAudio.sendFrame(opusFrame);
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aAudio.addTimestamp(960); // 20ms @ 48kHz
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}
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for (let i = 0; i < 3; i++) {
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aVideo.sendFrame(h264Frame);
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aVideo.addTimestamp(3000); // 33ms @ 90kHz
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}
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await sleep(500);
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console.log("[pkt] after send: states:", states.a, states.b);
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console.log("[pkt] audio track open:", aAudio.isOpen(), "| video track open:", aVideo.isOpen());
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const ok = states.a === "connected" && aAudio.isOpen() && aVideo.isOpen();
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console.log(ok ? "PKT TEST PASSED" : "PKT TEST FAILED");
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pcA.close();
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pcB.close();
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process.exit(ok ? 0 : 1);
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}
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main().catch((e) => {
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console.error("[pkt] FAILED:", e.message);
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process.exit(1);
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});
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setTimeout(() => {
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console.error("[pkt] TIMEOUT");
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process.exit(1);
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}, 25000);
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