Use persistent WebTransport frame stream

AGENTS.md

@@ -48,6 +48,8 @@ Audio is the playback clock. The server sends JPEG frames over WebSocket by defa
 - First 8 bytes: little-endian float64 timestamp in seconds.
 - Remaining bytes: one complete JPEG image.
 
+WebSocket sends one binary frame packet per message. WebTransport uses one persistent server-to-client unidirectional frame stream: the stream begins with the single byte `WT_STREAM_FRAME`, then repeats records of 4-byte little-endian uint32 packet length followed by one complete binary frame packet. Do not open one WebTransport stream per JPEG frame; that caused poor device performance at normal frame rates.
+
 The frontend decodes JPEGs with browser image APIs, queues frames, and paints frames whose timestamps are due relative to `audio.currentTime`. This means the browser decodes only audio and still images, not video.
 
 ## Why JPEG Frames

README.md

@@ -3,7 +3,7 @@
 A small web app that plays a remote video stream without using browser video decoding. The server uses `ffmpeg` to decode the input URL into:
 
 - an MP3 audio stream served to a normal `<audio>` element
-- timed JPEG image frames sent over a WebSocket, or optionally WebTransport/QUIC, and painted onto a `<canvas>`
+- timed JPEG image frames sent over WebSocket, or optionally WebTransport/QUIC, and painted onto a `<canvas>`
 
 This is meant for machines where image and audio decoding work but browser video decoding is unavailable or unreliable.
 
@@ -56,9 +56,9 @@ The app sets `FFMPEG_INPUT_SEEKABLE=0` by default so `ffmpeg` reads stream input
 
 YouTube URLs are resolved server-side with `yt-dlp` before they enter the existing ffmpeg pipeline. Recents and favorites keep the original YouTube URL, while the short-lived playback session uses the resolved media URL and headers returned by `yt-dlp`. Tune the selected format with `YT_DLP_FORMAT` and the resolver timeout with `YT_DLP_TIMEOUT_MS`.
 
-JPEG frames are dropped when the browser WebSocket falls behind instead of letting stale frames queue indefinitely. Tune the server-side backlog cap with `MAX_WS_BUFFER_BYTES`; the default is `2097152`.
+JPEG frames are dropped when the browser frame transport falls behind instead of letting stale frames queue indefinitely. Tune the server-side backlog cap with `MAX_WS_BUFFER_BYTES`; the default is `2097152`.
 
-WebSocket remains the default frame transport. To try WebTransport/QUIC for frame delivery, set `FRAME_TRANSPORT=webtransport` or `FRAME_TRANSPORT=auto` and expose the WebTransport UDP port. The default WebTransport port is `PORT + 1`; override it with `WEBTRANSPORT_PORT` and, when needed behind Docker or a proxy, `WEBTRANSPORT_PUBLIC_HOST` and `WEBTRANSPORT_PUBLIC_PORT`. The server generates a short-lived local ECDSA certificate under `data/` unless `WEBTRANSPORT_CERT_PATH` and `WEBTRANSPORT_KEY_PATH` are both set. Browser WebTransport requires a secure context: localhost is usually allowed, but plain HTTP over a LAN address may not expose the API. Browsers that cannot connect with WebTransport fall back to WebSocket.
+WebSocket remains the default frame transport. To try WebTransport/QUIC for frame delivery, set `FRAME_TRANSPORT=webtransport` or `FRAME_TRANSPORT=auto` and expose the WebTransport UDP port. The WebSocket path sends one binary frame packet per message. The WebTransport path uses one persistent server-to-client unidirectional stream with repeated length-prefixed frame packets. The default WebTransport port is `PORT + 1`; override it with `WEBTRANSPORT_PORT` and, when needed behind Docker or a proxy, `WEBTRANSPORT_PUBLIC_HOST` and `WEBTRANSPORT_PUBLIC_PORT`. The server generates a short-lived local ECDSA certificate under `data/` unless `WEBTRANSPORT_CERT_PATH` and `WEBTRANSPORT_KEY_PATH` are both set. Browser WebTransport requires a secure context: localhost is usually allowed, but plain HTTP over a LAN address may not expose the API. Browsers that cannot connect with WebTransport fall back to WebSocket.
 
 In single mode, audio output from `ffmpeg` is buffered before it is written to the browser so short HTTP backpressure pauses are less likely to stall frame generation. Tune the cap with `MAX_AUDIO_QUEUE_BYTES`; the default is `4194304`.
 

public/app.js

@@ -73,6 +73,8 @@ const METADATA_REFRESH_INTERVAL_MS = 650;
 const WT_STREAM_CONTROL_TO_CLIENT = 1;
 const WT_STREAM_FRAME = 2;
 const WT_STREAM_CONTROL_TO_SERVER = 3;
+const WT_FRAME_RECORD_HEADER_BYTES = 4;
+const MAX_FRAME_PACKET_BYTES = 16 * 1024 * 1024;
 const FRAME_CONNECTION_OPEN = 1;
 const FRAME_CONNECTION_CLOSING = 2;
 const FRAME_CONNECTION_CLOSED = 3;
@@ -741,35 +743,64 @@ async function readWebTransportControlStream(reader, initialPayload, events) {
 }
 
 async function readWebTransportFrameStream(reader, initialPayload, events) {
-  const chunks = [];
+  let pending = new Uint8Array(0);
-  let byteLength = 0;
 
   if (initialPayload.byteLength > 0) {
-    chunks.push(initialPayload);
+    pending = appendWebTransportFrameChunk(pending, initialPayload);
-    byteLength += initialPayload.byteLength;
+    pending = dispatchWebTransportFrameRecords(pending, events);
   }
 
   while (true) {
     const { value, done } = await reader.read();
 
     if (done) {
-      const packet = new Uint8Array(byteLength);
+      if (pending.byteLength > 0) {
-      let offset = 0;
+        throw new Error('WebTransport frame stream ended with a partial frame record.');
-
-      for (const chunk of chunks) {
-        packet.set(chunk, offset);
-        offset += chunk.byteLength;
       }
 
-      events.dispatchEvent(new MessageEvent('message', { data: packet.buffer }));
       return;
     }
 
-    chunks.push(value);
+    pending = appendWebTransportFrameChunk(pending, value);
-    byteLength += value.byteLength;
+    pending = dispatchWebTransportFrameRecords(pending, events);
   }
 }
 
+function appendWebTransportFrameChunk(pending, chunk) {
+  if (pending.byteLength === 0) {
+    return chunk;
+  }
+
+  const merged = new Uint8Array(pending.byteLength + chunk.byteLength);
+  merged.set(pending, 0);
+  merged.set(chunk, pending.byteLength);
+  return merged;
+}
+
+function dispatchWebTransportFrameRecords(buffer, events) {
+  let offset = 0;
+
+  while (buffer.byteLength - offset >= WT_FRAME_RECORD_HEADER_BYTES) {
+    const packetLength = new DataView(buffer.buffer, buffer.byteOffset + offset, WT_FRAME_RECORD_HEADER_BYTES).getUint32(0, true);
+
+    if (packetLength <= 8 || packetLength > MAX_FRAME_PACKET_BYTES) {
+      throw new Error(`Invalid WebTransport frame packet length: ${packetLength}`);
+    }
+
+    const recordBytes = WT_FRAME_RECORD_HEADER_BYTES + packetLength;
+
+    if (buffer.byteLength - offset < recordBytes) {
+      break;
+    }
+
+    const packet = buffer.slice(offset + WT_FRAME_RECORD_HEADER_BYTES, offset + recordBytes);
+    events.dispatchEvent(new MessageEvent('message', { data: packet.buffer }));
+    offset += recordBytes;
+  }
+
+  return offset === buffer.byteLength ? new Uint8Array(0) : buffer.slice(offset);
+}
+
 function dispatchWebTransportControlLines(text, events) {
   const lines = text.split(/\n/);
   const pending = lines.pop() ?? '';

server/index.js

@@ -65,6 +65,7 @@ const JPEG_EOI = Buffer.from([0xff, 0xd9]);
 const WT_STREAM_CONTROL_TO_CLIENT = 1;
 const WT_STREAM_FRAME = 2;
 const WT_STREAM_CONTROL_TO_SERVER = 3;
+const WT_FRAME_RECORD_HEADER_BYTES = 4;
 const BEST_EFFORT_RESUME_MAX_SECONDS = 30 * 24 * 60 * 60;
 const RECORDED_MEDIA_EXTENSIONS = new Set([
   '.avi',
@@ -2524,6 +2525,8 @@ class WebTransportFrameConnection extends EventEmitter {
     this._closedEmitted = false;
     this._controlStreamPromise = this.openControlStream();
     this._controlWrite = Promise.resolve();
+    this._frameStreamPromise = null;
+    this._frameWrite = Promise.resolve();
 
     this.session.closed().then((info) => {
       this.markClosed(info?.closeCode ?? 1000, info?.reason ?? '');
@@ -2554,7 +2557,11 @@ class WebTransportFrameConnection extends EventEmitter {
     }
 
     const packet = Buffer.isBuffer(data) ? data : Buffer.from(data);
-    this.writeFrame(packet).then(() => {
+    this._frameWrite = this._frameWrite
+      .catch(() => {})
+      .then(() => this.writeFrame(packet));
+
+    this._frameWrite.then(() => {
       done?.();
     }).catch((error) => {
       this.emitError(error);
@@ -2583,6 +2590,17 @@ class WebTransportFrameConnection extends EventEmitter {
     return stream;
   }
 
+  async openFrameStream() {
+    const stream = await this.session.openUni();
+    await stream.write(Buffer.from([WT_STREAM_FRAME]));
+    return stream;
+  }
+
+  getFrameStream() {
+    this._frameStreamPromise ??= this.openFrameStream();
+    return this._frameStreamPromise;
+  }
+
   async writeControl(data) {
     if (this.readyState !== WebSocket.OPEN) {
       return;
@@ -2597,17 +2615,20 @@ class WebTransportFrameConnection extends EventEmitter {
       throw new Error('WebTransport frame connection is closed.');
     }
 
-    this._bufferedAmount += packet.length;
+    if (packet.length > 0xffffffff) {
+      throw new Error('WebTransport frame packet is too large.');
+    }
+
+    const recordHeader = Buffer.allocUnsafe(WT_FRAME_RECORD_HEADER_BYTES);
+    recordHeader.writeUInt32LE(packet.length, 0);
+    this._bufferedAmount += WT_FRAME_RECORD_HEADER_BYTES + packet.length;
 
     try {
-      const stream = await this.session.openUni();
+      const stream = await this.getFrameStream();
-      const payload = Buffer.allocUnsafe(packet.length + 1);
+      await stream.write(recordHeader);
-      payload[0] = WT_STREAM_FRAME;
+      await stream.write(packet);
-      packet.copy(payload, 1);
-      await stream.write(payload);
-      await stream.finish();
     } finally {
-      this._bufferedAmount = Math.max(0, this._bufferedAmount - packet.length);
+      this._bufferedAmount = Math.max(0, this._bufferedAmount - WT_FRAME_RECORD_HEADER_BYTES - packet.length);
     }
   }