werift 0.23.0 → 0.24.0

package/lib/webrtc/src/nonstandard/userMedia.js.map

@@ -1 +1 @@
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{ type ChildProcess, exec } from \"child_process\";\nimport { randomUUID } from \"crypto\";\nimport { createSocket } from \"dgram\";\nimport { setImmediate } from \"timers/promises\";\n\nimport { randomPort } from \"../imports/common\";\nimport { RtpPacket } from \"../imports/rtp\";\nimport { MediaStreamTrack } from \"../media/track\";\n\nexport const getUserMedia = async ({\n  path,\n  loop,\n  width,\n  height,\n}: {\n  path: string;\n  loop?: boolean;\n  width?: number;\n  height?: number;\n}) => {\n  const audioPort = await randomPort();\n  const videoPort = await randomPort();\n\n  if (path.endsWith(\".mp4\")) {\n    return new MediaPlayerMp4({\n      audioPort,\n      videoPort,\n      path,\n      loop,\n      width,\n      height,\n    });\n  } else {\n    return new MediaPlayerWebm({\n      audioPort,\n      videoPort,\n      path,\n      loop,\n      width,\n      height,\n    });\n  }\n};\n\nabstract class MediaPlayer {\n  protected streamId = randomUUID().toString();\n  audio = new MediaStreamTrack({ kind: \"audio\", streamId: this.streamId });\n  video = new MediaStreamTrack({ kind: \"video\", streamId: this.streamId });\n  protected process!: ChildProcess;\n  stopped = false;\n\n  constructor(\n    protected props: {\n      videoPort: number;\n      audioPort: number;\n      path: string;\n      loop?: boolean;\n      width?: number;\n      height?: number;\n    },\n  ) {\n    this.setupTrack(props.audioPort, this.audio);\n    this.setupTrack(props.videoPort, this.video);\n  }\n\n  private setupTrack = (port: number, track: MediaStreamTrack) => {\n    let payloadType = 0;\n\n    const socket = createSocket(\"udp4\");\n    socket.bind(port);\n    socket.on(\"message\", async (buf) => {\n      const rtp = RtpPacket.deSerialize(buf);\n      if (!payloadType) {\n        payloadType = rtp.header.payloadType;\n      }\n\n      // detect gStreamer restarted\n      if (payloadType !== rtp.header.payloadType) {\n        payloadType = rtp.header.payloadType;\n        track.onSourceChanged.execute(rtp.header);\n      }\n\n      track.writeRtp(buf);\n    });\n  };\n\n  stop() {\n    this.stopped = true;\n    this.process.kill(\"SIGINT\");\n  }\n}\n\nexport class MediaPlayerMp4 extends MediaPlayer {\n  async start() {\n    let payloadType = 96;\n    const run = async () => {\n      if (payloadType > 100) payloadType = 96;\n\n      let cmd = \"\";\n      if (this.props.width && this.props.height) {\n        cmd = `gst-launch-1.0 filesrc location= ${this.props.path} ! \\\ndecodebin ! videoscale ! video/x-raw,width=${this.props.width},height=${\n          this.props.height\n        } ! x264enc ! \\\nh264parse ! rtph264pay config-interval=10 pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort}`;\n      } else {\n        cmd = `gst-launch-1.0 filesrc location= ${this.props.path} ! \\\nqtdemux name=d ! \\\nqueue ! h264parse ! rtph264pay config-interval=10 pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort} d. ! \\\nqueue ! aacparse ! avdec_aac ! audioresample ! audioconvert ! opusenc ! rtpopuspay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.audioPort}`;\n      }\n      console.log(cmd);\n      this.process = exec(cmd);\n      this.process.on(\"error\", (e) => console.error(\"gst error\", e));\n\n      if (this.props.loop) {\n        await new Promise((r) => this.process.on(\"close\", r));\n        if (!this.stopped) {\n          run();\n        }\n      }\n    };\n    await setImmediate();\n    run();\n  }\n}\n\nexport class MediaPlayerWebm extends MediaPlayer {\n  async start() {\n    let payloadType = 96;\n    const run = async () => {\n      if (payloadType > 100) payloadType = 96;\n\n      const cmd = `gst-launch-1.0 filesrc location=${this.props.path} ! matroskademux name=d \\\nd.video_0 ! queue ! rtpvp8pay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.videoPort} \\\nd.audio_0 ! queue ! rtpopuspay pt=${payloadType++} ! \\\nudpsink host=127.0.0.1 port=${this.props.audioPort}`;\n      this.process = exec(cmd);\n\n      if (this.props.loop) {\n        await new Promise((r) => this.process.on(\"close\", r));\n        if (!this.stopped) {\n          run();\n        }\n      }\n    };\n    await setImmediate();\n    run();\n  }\n}\n"]}
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{ randomUUID } from \"crypto\";\nimport { homedir } from \"os\";\nimport { isAbsolute, resolve } from \"path\";\nimport { Readable } from \"stream\";\nimport { readFile } from \"fs/promises\";\nimport {\n  setImmediate as onNextTurn,\n  setTimeout as wait,\n} from \"timers/promises\";\n\nimport {\n  BufferSource,\n  EncodedPacketSink,\n  Input,\n  type InputAudioTrack,\n  type InputTrack,\n  type InputVideoTrack,\n  MP4,\n  WEBM,\n} from \"mediabunny\";\n\nimport { Event } from \"../imports/common\";\nimport type { RTCRtpCodecParameters } from \"../media/parameters\";\nimport { MediaStreamTrack } from \"../media/track\";\nimport { codecParametersFromString } from \"../sdp\";\nimport {\n  type SupportedSourceCodec,\n  createPacketizer,\n  supportedSourceCodecs,\n  toSupportedMimeType,\n} from \"./userMedia/packetizer\";\n\ntype UserMediaSource =\n  | { path: string; buffer?: never; stream?: never }\n  | {\n      path?: never;\n      buffer: Buffer | ArrayBuffer | ArrayBufferView;\n      stream?: never;\n    }\n  | {\n      path?: never;\n      buffer?: never;\n      stream: Readable | ReadableStream<Uint8Array>;\n    };\n\nexport type FileMediaTrackSource = UserMediaSource & {\n  loop?: boolean;\n};\n\nexport const getUserMedia = async (options: FileMediaTrackSource) => {\n  assertSupportedOptions(options);\n  const replayableSource = await createReplayableSource(options);\n  const input = await createInput(replayableSource);\n  try {\n    if (!(await input.canRead())) {\n      throw new Error(\"Only MP4 and WebM file playback is supported\");\n    }\n\n    const streamId = randomUUID().toString();\n    const playbackTracks = await selectPlaybackTracks(input, streamId);\n    if (playbackTracks.length === 0) {\n      throw new Error(\n        \"The media source does not contain playable audio or video tracks\",\n      );\n    }\n\n    const mediaStartTimestamp = await input.getFirstTimestamp(\n      playbackTracks.map(({ inputTrack }) => inputTrack),\n    );\n\n    return new MediaPlayer({\n      source: replayableSource,\n      loop: options.loop ?? false,\n      streamId,\n      session: createPlaybackSession({\n        input,\n        mediaStartTimestamp,\n        playbackTracks,\n      }),\n    });\n  } catch (error) {\n    input.dispose();\n    throw error;\n  }\n};\n\nclass MediaPlayer {\n  readonly audio?: MediaStreamTrack;\n  readonly video?: MediaStreamTrack;\n  readonly onError = new Event<[Error]>();\n\n  private readonly source: ReplayableUserMediaSource;\n  private readonly loop: boolean;\n  private readonly streamId: string;\n  private session?: PlaybackSession;\n  private running?: Promise<void>;\n  private abortController?: AbortController;\n  private hasStartedPlayback = false;\n  private stopped = false;\n\n  constructor({\n    source,\n    loop,\n    streamId,\n    session,\n  }: {\n    source: ReplayableUserMediaSource;\n    loop: boolean;\n    streamId: string;\n    session: PlaybackSession;\n  }) {\n    this.source = source;\n    this.loop = loop;\n    this.streamId = streamId;\n    this.session = session;\n\n    this.audio = session.playbackTracks.find(\n      ({ track }) => track.kind === \"audio\",\n    )?.track;\n    this.video = session.playbackTracks.find(\n      ({ track }) => track.kind === \"video\",\n    )?.track;\n  }\n\n  async start() {\n    if (this.stopped) {\n      throw new Error(\"The media source has already been stopped\");\n    }\n    if (this.running) {\n      return;\n    }\n\n    const abortController = new AbortController();\n    this.abortController = abortController;\n\n    let startupPending = true;\n    let session: PlaybackSession | undefined;\n    const playback = (async () => {\n      session = await this.getOrCreateSession();\n      if (abortController.signal.aborted || this.stopped) {\n        return;\n      }\n\n      session.runners.forEach((runner) => runner.assertReady());\n\n      const initialSourceChanged = this.hasStartedPlayback;\n      this.hasStartedPlayback = true;\n      await this.run(session, abortController, initialSourceChanged);\n    })();\n    const observedPlayback = playback.catch((error) => {\n      if (abortController.signal.aborted) {\n        return;\n      }\n\n      this.onError.execute(toError(error));\n      if (startupPending) {\n        throw error;\n      }\n    });\n\n    this.running = observedPlayback.finally(() => {\n      if (this.abortController === abortController) {\n        this.abortController = undefined;\n      }\n      this.running = undefined;\n      this.disposeSession(session);\n    });\n\n    try {\n      await Promise.race([this.running, onNextTurn()]);\n    } finally {\n      startupPending = false;\n    }\n  }\n\n  stop() {\n    if (this.stopped) {\n      return;\n    }\n    // File playback mirrors the previous process-backed behavior: stop() is terminal.\n    this.stopped = true;\n    this.abortController?.abort();\n    if (!this.running) {\n      this.disposeSession();\n    }\n  }\n\n  private async run(\n    session: PlaybackSession,\n    abortController: AbortController,\n    initialSourceChanged: boolean,\n  ) {\n    let sourceChanged = initialSourceChanged;\n\n    do {\n      const startedAt = performance.now();\n      await Promise.all(\n        session.runners.map((runner) =>\n          runner.play({\n            startedAt,\n            signal: abortController.signal,\n            sourceChanged,\n          }),\n        ),\n      );\n      if (!this.loop || abortController.signal.aborted) {\n        break;\n      }\n      sourceChanged = true;\n    } while (!abortController.signal.aborted);\n  }\n\n  private async getOrCreateSession() {\n    if (this.session) {\n      return this.session;\n    }\n\n    const input = await createInput(this.source);\n    try {\n      if (!(await input.canRead())) {\n        throw new Error(\"Only MP4 and WebM file playback is supported\");\n      }\n\n      const playbackTracks = await selectPlaybackTracks(\n        input,\n        this.streamId,\n        buildExistingTrackMap(this.audio, this.video),\n      );\n      const mediaStartTimestamp = await input.getFirstTimestamp(\n        playbackTracks.map(({ inputTrack }) => inputTrack),\n      );\n\n      const session = createPlaybackSession({\n        input,\n        mediaStartTimestamp,\n        playbackTracks,\n      });\n      this.session = session;\n      return session;\n    } catch (error) {\n      input.dispose();\n      throw error;\n    }\n  }\n\n  private disposeSession(session = this.session) {\n    if (!session) {\n      return;\n    }\n\n    session.input.dispose();\n    if (this.session === session) {\n      this.session = undefined;\n    }\n  }\n}\n\nclass TrackPlaybackRunner {\n  constructor(\n    private readonly props: PlaybackTrack & {\n      mediaStartTimestamp: number;\n    },\n  ) {}\n\n  async play({\n    startedAt,\n    signal,\n    sourceChanged,\n  }: {\n    startedAt: number;\n    signal: AbortSignal;\n    sourceChanged: boolean;\n  }) {\n    const codec = this.requireNegotiatedCodec();\n    const packetizer = createPacketizer({\n      codec,\n      sourceCodec: this.props.sourceCodec,\n      decoderDescription: this.props.decoderDescription,\n    });\n    const packetSink = new EncodedPacketSink(this.props.inputTrack);\n    let sourceChangeNotified = false;\n\n    try {\n      for await (const packet of packetSink.packets(undefined, undefined, {\n        verifyKeyPackets: this.props.track.kind === \"video\",\n      })) {\n        signal.throwIfAborted?.();\n\n        const dueAtMs =\n          (packet.timestamp - this.props.mediaStartTimestamp) * 1_000;\n        const remainingMs = dueAtMs - (performance.now() - startedAt);\n        if (remainingMs > 0) {\n          await wait(remainingMs, undefined, { signal });\n        }\n\n        const rtpTimestamp = toRtpTimestamp(\n          packet.timestamp,\n          this.props.mediaStartTimestamp,\n          codec.clockRate,\n        );\n        const packets = packetizer.packetize(packet, rtpTimestamp);\n        if (packets.length === 0) {\n          continue;\n        }\n\n        if (sourceChanged && !sourceChangeNotified) {\n          this.props.track.onSourceChanged.execute(packets[0].header);\n          sourceChangeNotified = true;\n        }\n\n        for (const rtpPacket of packets) {\n          this.props.track.writeRtp(rtpPacket);\n        }\n      }\n    } catch (error) {\n      if (signal.aborted || isAbortError(error)) {\n        return;\n      }\n      throw error;\n    }\n  }\n\n  assertReady() {\n    const codec = this.requireNegotiatedCodec();\n    createPacketizer({\n      codec,\n      sourceCodec: this.props.sourceCodec,\n      decoderDescription: this.props.decoderDescription,\n    });\n  }\n\n  private requireNegotiatedCodec() {\n    const codec = this.props.track.codec;\n    if (!codec) {\n      throw new Error(\n        `Cannot start ${this.props.track.kind} playback before the track is attached to an RTCRtpSender`,\n      );\n    }\n\n    const expectedMimeType = toSupportedMimeType(this.props.sourceCodec);\n    const negotiatedMimeType = normalizeMimeType(codec.mimeType);\n    if (negotiatedMimeType !== expectedMimeType) {\n      throw new Error(\n        `Input ${this.props.track.kind} codec ${expectedMimeType} does not match negotiated codec ${codec.mimeType}`,\n      );\n    }\n\n    const expectedClockRate =\n      this.props.track.kind === \"audio\" ? 48_000 : 90_000;\n    if (codec.clockRate !== expectedClockRate) {\n      throw new Error(\n        `Input ${this.props.track.kind} clock rate ${expectedClockRate} does not match negotiated clock rate ${codec.clockRate}`,\n      );\n    }\n\n    assertCodecParameters({\n      trackKind: this.props.track.kind === \"audio\" ? \"audio\" : \"video\",\n      codec,\n      sourceCodec: this.props.sourceCodec,\n      decoderDescription: this.props.decoderDescription,\n      sourceChannels: this.props.sourceChannels,\n    });\n\n    return codec;\n  }\n}\n\ninterface PlaybackTrack {\n  inputTrack: InputTrack;\n  track: MediaStreamTrack;\n  sourceCodec: SupportedSourceCodec;\n  decoderDescription?: ArrayBuffer | ArrayBufferView | null;\n  sourceChannels?: number;\n}\n\ninterface PlaybackSession {\n  input: Input;\n  mediaStartTimestamp: number;\n  playbackTracks: PlaybackTrack[];\n  runners: TrackPlaybackRunner[];\n}\n\nasync function selectPlaybackTracks(\n  input: Input,\n  streamId: string,\n  existingTracks: Partial<Record<\"audio\" | \"video\", MediaStreamTrack>> = {},\n) {\n  const primaryVideoTrack = await input.getPrimaryVideoTrack();\n  const primaryAudioTrack = primaryVideoTrack\n    ? ((await primaryVideoTrack.getPrimaryPairableAudioTrack()) ??\n      (await input.getPrimaryAudioTrack()))\n    : await input.getPrimaryAudioTrack();\n\n  const playbackTracks: PlaybackTrack[] = [];\n  if (primaryAudioTrack) {\n    playbackTracks.push(\n      await createPlaybackTrack(\n        primaryAudioTrack,\n        streamId,\n        existingTracks.audio,\n      ),\n    );\n  }\n  if (primaryVideoTrack) {\n    playbackTracks.push(\n      await createPlaybackTrack(\n        primaryVideoTrack,\n        streamId,\n        existingTracks.video,\n      ),\n    );\n  }\n\n  if (!primaryVideoTrack && !primaryAudioTrack) {\n    const fallbackAudioTrack = await input.getPrimaryAudioTrack();\n    if (fallbackAudioTrack) {\n      playbackTracks.push(\n        await createPlaybackTrack(\n          fallbackAudioTrack,\n          streamId,\n          existingTracks.audio,\n        ),\n      );\n    }\n  }\n\n  return playbackTracks;\n}\n\nasync function createPlaybackTrack(\n  inputTrack: InputAudioTrack | InputVideoTrack,\n  streamId: string,\n  track?: MediaStreamTrack,\n): Promise<PlaybackTrack> {\n  const sourceCodec = await requireSupportedCodec(inputTrack);\n  const decoderConfig = await inputTrack.getDecoderConfig();\n  const kind = inputTrack.isAudioTrack() ? \"audio\" : \"video\";\n\n  return {\n    inputTrack,\n    sourceCodec,\n    decoderDescription:\n      decoderConfig && \"description\" in decoderConfig\n        ? (decoderConfig.description ?? null)\n        : null,\n    sourceChannels:\n      decoderConfig && \"numberOfChannels\" in decoderConfig\n        ? decoderConfig.numberOfChannels\n        : undefined,\n    track:\n      track ??\n      new MediaStreamTrack({\n        kind,\n        streamId,\n      }),\n  };\n}\n\nasync function requireSupportedCodec(inputTrack: InputTrack) {\n  const codec = await inputTrack.getCodec();\n  if (codec && supportedSourceCodecs.includes(codec as SupportedSourceCodec)) {\n    return codec as SupportedSourceCodec;\n  }\n\n  const codecLabel =\n    codec ??\n    (await inputTrack.getCodecParameterString()) ??\n    String((await inputTrack.getInternalCodecId()) ?? \"unknown\");\n\n  throw new Error(\n    `Unsupported ${inputTrack.type} codec \"${codecLabel}\". File playback supports H264, VP8, VP9, AV1, and Opus only.`,\n  );\n}\n\ntype ReplayableUserMediaSource =\n  | { path: string; buffer?: never }\n  | { path?: never; buffer: Buffer };\n\nasync function createReplayableSource(\n  options: UserMediaSource,\n): Promise<ReplayableUserMediaSource> {\n  if (typeof options.path === \"string\") {\n    return { path: options.path };\n  }\n\n  const buffer =\n    options.buffer != undefined\n      ? toBuffer(options.buffer)\n      : await readStreamToBuffer(options.stream);\n\n  return { buffer };\n}\n\nasync function createInput(options: ReplayableUserMediaSource) {\n  if (typeof options.path === \"string\") {\n    const buffer = await readFile(resolveUserMediaPath(options.path));\n    return new Input({\n      source: new BufferSource(buffer),\n      formats: [MP4, WEBM],\n    });\n  }\n\n  return new Input({\n    source: new BufferSource(options.buffer),\n    formats: [MP4, WEBM],\n  });\n}\n\nfunction resolveUserMediaPath(path: string) {\n  if (path === \"~\") {\n    return homedir();\n  }\n  if (path.startsWith(\"~/\")) {\n    return resolve(homedir(), path.slice(2));\n  }\n  return isAbsolute(path) ? path : resolve(path);\n}\n\nasync function readStreamToBuffer(\n  stream: Readable | ReadableStream<Uint8Array>,\n) {\n  const chunks: Buffer[] = [];\n\n  if (stream instanceof Readable) {\n    for await (const chunk of stream) {\n      chunks.push(toBuffer(chunk));\n    }\n    return Buffer.concat(chunks);\n  }\n\n  const reader = stream.getReader();\n  try {\n    for (;;) {\n      const { done, value } = await reader.read();\n      if (done) {\n        break;\n      }\n      if (value) {\n        chunks.push(\n          Buffer.from(value.buffer, value.byteOffset, value.byteLength),\n        );\n      }\n    }\n  } finally {\n    reader.releaseLock();\n  }\n\n  return Buffer.concat(chunks);\n}\n\nfunction toBuffer(value: Buffer | ArrayBuffer | ArrayBufferView | string) {\n  if (Buffer.isBuffer(value)) {\n    return value;\n  }\n  if (typeof value === \"string\") {\n    return Buffer.from(value);\n  }\n  if (ArrayBuffer.isView(value)) {\n    return Buffer.from(value.buffer, value.byteOffset, value.byteLength);\n  }\n  return Buffer.from(new Uint8Array(value));\n}\n\nfunction normalizeMimeType(mimeType: string) {\n  return mimeType.toLowerCase();\n}\n\nfunction assertSupportedOptions(options: FileMediaTrackSource) {\n  const legacyOptions = options as FileMediaTrackSource & {\n    width?: unknown;\n    height?: unknown;\n  };\n\n  if (legacyOptions.width != undefined || legacyOptions.height != undefined) {\n    throw new Error(\n      \"getUserMedia({ width, height }) is no longer supported for file playback. Resize or re-encode the source before calling getUserMedia().\",\n    );\n  }\n}\n\nfunction createPlaybackSession({\n  input,\n  mediaStartTimestamp,\n  playbackTracks,\n}: {\n  input: Input;\n  mediaStartTimestamp: number;\n  playbackTracks: PlaybackTrack[];\n}): PlaybackSession {\n  return {\n    input,\n    mediaStartTimestamp,\n    playbackTracks,\n    runners: playbackTracks.map(\n      (playbackTrack) =>\n        new TrackPlaybackRunner({\n          mediaStartTimestamp,\n          ...playbackTrack,\n        }),\n    ),\n  };\n}\n\nfunction buildExistingTrackMap(\n  audio?: MediaStreamTrack,\n  video?: MediaStreamTrack,\n): Partial<Record<\"audio\" | \"video\", MediaStreamTrack>> {\n  return {\n    ...(audio ? { audio } : {}),\n    ...(video ? { video } : {}),\n  };\n}\n\nfunction assertCodecParameters({\n  trackKind,\n  codec,\n  sourceCodec,\n  decoderDescription,\n  sourceChannels,\n}: {\n  trackKind: \"audio\" | \"video\";\n  codec: RTCRtpCodecParameters;\n  sourceCodec: SupportedSourceCodec;\n  decoderDescription?: ArrayBuffer | ArrayBufferView | null;\n  sourceChannels?: number;\n}) {\n  if (sourceCodec === \"opus\" && sourceChannels && codec.channels != undefined) {\n    if (codec.channels !== sourceChannels) {\n      throw new Error(\n        `Input ${trackKind} channels ${sourceChannels} do not match negotiated channels ${codec.channels}`,\n      );\n    }\n    return;\n  }\n\n  if (sourceCodec !== \"avc\") {\n    // mediabunny exposes codec-specific decoder configuration for H264 only here.\n    return;\n  }\n\n  const negotiatedParameters = codecParametersFromString(\n    codec.parameters ?? \"\",\n  );\n  const negotiatedPacketizationMode = Number(\n    negotiatedParameters[\"packetization-mode\"] ?? 1,\n  );\n  if (negotiatedPacketizationMode !== 1) {\n    throw new Error(\n      `Input ${trackKind} packetization-mode 1 does not match negotiated packetization-mode ${negotiatedPacketizationMode}`,\n    );\n  }\n\n  const sourceProfileLevelId = getH264ProfileLevelId(decoderDescription);\n  const negotiatedProfileLevelId = normalizeOptionalHexParameter(\n    negotiatedParameters[\"profile-level-id\"],\n  );\n\n  if (!sourceProfileLevelId) {\n    return;\n  }\n\n  if (!negotiatedProfileLevelId) {\n    throw new Error(\n      `Input ${trackKind} profile-level-id ${sourceProfileLevelId} does not match negotiated profile-level-id <missing>`,\n    );\n  }\n\n  if (sourceProfileLevelId !== negotiatedProfileLevelId) {\n    throw new Error(\n      `Input ${trackKind} profile-level-id ${sourceProfileLevelId} does not match negotiated profile-level-id ${negotiatedProfileLevelId}`,\n    );\n  }\n}\n\nfunction getH264ProfileLevelId(\n  decoderDescription?: ArrayBuffer | ArrayBufferView | null,\n) {\n  if (!decoderDescription) {\n    return undefined;\n  }\n\n  const description = toDescriptionBuffer(decoderDescription);\n  if (description.length < 4) {\n    throw new Error(\"invalid H264 decoder configuration\");\n  }\n\n  return description.subarray(1, 4).toString(\"hex\");\n}\n\nfunction normalizeOptionalHexParameter(value: unknown) {\n  return typeof value === \"string\" ? value.toLowerCase() : undefined;\n}\n\nfunction toRtpTimestamp(\n  packetTimestamp: number,\n  mediaStartTimestamp: number,\n  clockRate: number,\n) {\n  const relativeSeconds = Math.max(0, packetTimestamp - mediaStartTimestamp);\n  return Math.round(relativeSeconds * clockRate) >>> 0;\n}\n\nfunction isAbortError(error: unknown) {\n  return (\n    error instanceof Error &&\n    (error.name === \"AbortError\" ||\n      error.message === \"The operation was aborted\")\n  );\n}\n\nfunction toError(error: unknown) {\n  return error instanceof Error ? error : new Error(String(error));\n}\n\nfunction toDescriptionBuffer(\n  decoderDescription: ArrayBuffer | ArrayBufferView,\n) {\n  if (ArrayBuffer.isView(decoderDescription)) {\n    return Buffer.from(\n      decoderDescription.buffer,\n      decoderDescription.byteOffset,\n      decoderDescription.byteLength,\n    );\n  }\n\n  return Buffer.from(decoderDescription);\n}\n"]}

package/lib/webrtc/src/peerConnection.d.ts

@@ -1,16 +1,41 @@
 import type { RTCDataChannel } from "./dataChannel";
 import { EventTarget } from "./helper";
-import { type Address, Event, type InterfaceAddresses } from "./imports/common";
+import { type Address, Event, type InterfaceAddresses, type TlsConnectionOptions } from "./imports/common";
 import type { CandidatePair, Message, Protocol } from "./imports/ice";
 import { type MediaStream, type MediaStreamTrack, type RTCRtpCodecParameters, type RTCRtpHeaderExtensionParameters, type RTCRtpReceiver, type RTCRtpSender, type RTCRtpTransceiver, type TransceiverOptions } from "./media";
-import { RTCStatsReport } from "./media/stats";
-import { type BundlePolicy, SessionDescription } from "./sdp";
+import { type RTCStatsReport } from "./media/stats";
+import { type BundlePolicy, type RTCSessionDescription, SessionDescription } from "./sdp";
 import { type RTCSessionDescriptionInit } from "./sdpManager";
-import type { DtlsKeys, RTCDtlsTransport } from "./transport/dtls";
+import type { DtlsKeys, RTCCertificate, RTCDtlsTransport } from "./transport/dtls";
 import type { RTCIceCandidate, RTCIceCandidateInit, RTCIceTransport } from "./transport/ice";
+import { type RTCSctpTransport } from "./transport/sctp";
 import type { Kind, RTCSignalingState } from "./types/domain";
 import type { Callback, CallbackWithValue } from "./types/util";
+/**
+ * W3C compatibility notes kept near the public RTCPeerConnection surface so the
+ * reviewable diff does not depend on external PR text.
+ *
+ * - `current/pending*Description`, `canTrickleIceCandidates`, `sctp`,
+ *   `addIceCandidate(null)`, and `RTCConfiguration` round-trip behavior are
+ *   implemented here and covered by `tests/wpt/peerConnectionApiCompatibility.test.ts`.
+ * - `addIceCandidate()` also validates `sdpMid` / `sdpMLineIndex` /
+ *   `usernameFragment` against the applied remote description and appends
+ *   candidates or end-of-candidates markers to the corresponding m-section.
+ *   The public API keeps werift's historical pre-SRD buffering behavior, while
+ *   the WPT runner wraps the class to exercise strict spec rejection.
+ * - `bundlePolicy: "balanced"` is accepted for input compatibility but is
+ *   normalized to werift's `"max-compat"` behavior, so `getConfiguration()`
+ *   returns the normalized value.
+ * - `setLocalDescription()` keeps the historical `SessionDescription` return
+ *   value for non-rollback calls, while `{ type: "rollback" }` resolves `void`
+ *   to match the actual behavior without pretending to return a description.
+ * - API reference markdown is regenerated with `cd packages/webrtc && npm run doc`.
+ *   The generated output lives under `packages/webrtc/doc/`; compatibility
+ *   notes remain here and in the package README so review context is visible
+ *   even when generated docs are not committed in the same change.
+ */
 export declare class RTCPeerConnection extends EventTarget {
+    readonly id: string;
     readonly cname: string;
     config: Required<PeerConfig>;
     signalingState: RTCSignalingState;
@@ -23,41 +48,59 @@ export declare class RTCPeerConnection extends EventTarget {
     private readonly secureManager;
     private isClosed;
     private shouldNegotiationneeded;
-    private remoteBundleNegotiated;
+    private lastCreatedAnswer?;
+    private lastCreatedOffer?;
+    private readonly pendingRemoteCandidates;
     readonly iceGatheringStateChange: Event<["complete" | "new" | "gathering"]>;
-    readonly iceConnectionStateChange: Event<["disconnected" | "closed" | "completed" | "new" | "connected" | "failed" | "checking"]>;
+    readonly iceConnectionStateChange: Event<["closed" | "disconnected" | "completed" | "new" | "connected" | "failed" | "checking"]>;
     readonly signalingStateChange: Event<["closed" | "stable" | "have-local-offer" | "have-remote-offer" | "have-local-pranswer" | "have-remote-pranswer"]>;
-    readonly connectionStateChange: Event<["disconnected" | "closed" | "new" | "connected" | "connecting" | "failed"]>;
+    readonly connectionStateChange: Event<["closed" | "disconnected" | "new" | "connected" | "connecting" | "failed"]>;
     readonly onDataChannel: Event<[RTCDataChannel]>;
     readonly onRemoteTransceiverAdded: Event<[RTCRtpTransceiver]>;
     readonly onTransceiverAdded: Event<[RTCRtpTransceiver]>;
     readonly onIceCandidate: Event<[RTCIceCandidate | undefined]>;
     readonly onNegotiationneeded: Event<[]>;
     readonly onTrack: Event<[MediaStreamTrack]>;
-    ondatachannel?: CallbackWithValue<RTCDataChannelEvent>;
-    onicecandidate?: CallbackWithValue<RTCPeerConnectionIceEvent>;
-    onicecandidateerror?: CallbackWithValue<any>;
-    onicegatheringstatechange?: CallbackWithValue<any>;
-    onnegotiationneeded?: CallbackWithValue<any>;
-    onsignalingstatechange?: CallbackWithValue<any>;
-    ontrack?: CallbackWithValue<RTCTrackEvent>;
-    onconnectionstatechange?: Callback;
-    oniceconnectionstatechange?: Callback;
-    constructor(config?: Partial<PeerConfig>);
-    get connectionState(): "disconnected" | "closed" | "new" | "connected" | "connecting" | "failed";
-    get iceConnectionState(): "disconnected" | "closed" | "completed" | "new" | "connected" | "failed" | "checking";
+    private readonly eventHandlers;
+    get ondatachannel(): CallbackWithValue<RTCDataChannelEvent> | null;
+    set ondatachannel(value: CallbackWithValue<RTCDataChannelEvent> | null);
+    get onicecandidate(): CallbackWithValue<RTCPeerConnectionIceEvent> | null;
+    set onicecandidate(value: CallbackWithValue<RTCPeerConnectionIceEvent> | null);
+    get onicecandidateerror(): CallbackWithValue<any> | null;
+    set onicecandidateerror(value: CallbackWithValue<any> | null);
+    get onicegatheringstatechange(): CallbackWithValue<any> | null;
+    set onicegatheringstatechange(value: CallbackWithValue<any> | null);
+    get onnegotiationneeded(): CallbackWithValue<any> | null;
+    set onnegotiationneeded(value: CallbackWithValue<any> | null);
+    get onsignalingstatechange(): CallbackWithValue<any> | null;
+    set onsignalingstatechange(value: CallbackWithValue<any> | null);
+    get ontrack(): CallbackWithValue<RTCTrackEvent> | null;
+    set ontrack(value: CallbackWithValue<RTCTrackEvent> | null);
+    get onconnectionstatechange(): Callback | null;
+    set onconnectionstatechange(value: Callback | null);
+    get oniceconnectionstatechange(): Callback | null;
+    set oniceconnectionstatechange(value: Callback | null);
+    constructor(config?: RTCPeerConnectionConfig);
+    get connectionState(): "closed" | "disconnected" | "new" | "connected" | "connecting" | "failed";
+    get iceConnectionState(): "closed" | "disconnected" | "completed" | "new" | "connected" | "failed" | "checking";
     get iceGathererState(): "complete" | "new" | "gathering";
     get iceGatheringState(): "complete" | "new" | "gathering";
     get dtlsTransports(): RTCDtlsTransport[];
-    get sctpTransport(): import("./transport/sctp").RTCSctpTransport | undefined;
+    get sctpTransport(): RTCSctpTransport | undefined;
+    get sctp(): RTCSctpTransport | null;
     get sctpRemotePort(): number | undefined;
     get iceTransports(): RTCIceTransport[];
     get extIdUriMap(): {
         [id: number]: string;
     };
     get iceGeneration(): number;
-    get localDescription(): import("./sdp").RTCSessionDescription | undefined;
-    get remoteDescription(): import("./sdp").RTCSessionDescription | undefined;
+    get localDescription(): RTCSessionDescription | null;
+    get currentLocalDescription(): RTCSessionDescription | null;
+    get pendingLocalDescription(): RTCSessionDescription | null;
+    get remoteDescription(): RTCSessionDescription | null;
+    get currentRemoteDescription(): RTCSessionDescription | null;
+    get pendingRemoteDescription(): RTCSessionDescription | null;
+    get canTrickleIceCandidates(): boolean | null;
     get remoteIsBundled(): import("./sdp").GroupDescription | undefined;
     /**@private */
     get _localDescription(): SessionDescription | undefined;
@@ -66,11 +109,62 @@ export declare class RTCPeerConnection extends EventTarget {
     getTransceivers(): RTCRtpTransceiver[];
     getSenders(): RTCRtpSender[];
     getReceivers(): RTCRtpReceiver[];
-    setConfiguration(config: Partial<PeerConfig>): void;
-    getConfiguration(): Required<PeerConfig>;
+    setConfiguration(config: RTCPeerConnectionConfig): void;
+    getConfiguration(): {
+        codecs: {
+            audio: RTCRtpCodecParameters[] | undefined;
+            video: RTCRtpCodecParameters[] | undefined;
+        };
+        headerExtensions: {
+            audio: RTCRtpHeaderExtensionParameters[] | undefined;
+            video: RTCRtpHeaderExtensionParameters[] | undefined;
+        };
+        iceServers: {
+            urls: string | string[];
+            username?: string;
+            credential?: string;
+        }[];
+        icePortRange: [number, number] | undefined;
+        iceAdditionalHostAddresses: string[] | undefined;
+        dtls: {
+            keys?: DtlsKeys | undefined;
+        };
+        certificates: RTCCertificate[];
+        debug: {
+            inboundPacketLoss?: number | undefined;
+            outboundPacketLoss?: number | undefined;
+            receiverReportDelay?: number | undefined;
+            disableSendNack?: boolean | undefined;
+            disableRecvRetransmit?: boolean | undefined;
+        };
+        iceTransportPolicy: "all" | "relay";
+        /** Advertise local ICE lite and operate in the controlled role. */
+        iceLite: boolean;
+        iceInterfaceAddresses: InterfaceAddresses | undefined;
+        iceUseIpv4: boolean;
+        iceUseIpv6: boolean;
+        iceUseTcp: boolean;
+        turnTransport: "udp" | "tcp" | "tls" | undefined;
+        turnTlsOptions: TlsConnectionOptions | undefined;
+        /** @deprecated Prefer turn URL transport parameters or turnTransport. */
+        forceTurnTCP: boolean;
+        /** such as google cloud run */
+        iceUseLinkLocalAddress: boolean | undefined;
+        /** If provided, is called on each STUN request.
+         * Return `true` if a STUN response should be sent, false if it should be skipped. */
+        iceFilterStunResponse: ((message: Message, addr: Address, protocol: Protocol) => boolean) | undefined;
+        iceFilterCandidatePair: ((pair: CandidatePair) => boolean) | undefined;
+        icePasswordPrefix: string | undefined;
+        bundlePolicy: BundlePolicy;
+        rtcpMuxPolicy: "require";
+        iceCandidatePoolSize: number;
+        midSuffix: boolean;
+        /** Advertised local SCTP max-message-size in SDP. Use 0 for unlimited. */
+        maxMessageSize: number;
+    };
     createOffer({ iceRestart }?: {
         iceRestart?: boolean;
-    }): Promise<import("./sdp").RTCSessionDescription>;
+    }): Promise<RTCSessionDescription>;
     private createSctpTransport;
     createDataChannel(label: string, options?: Partial<{
         maxPacketLifeTime?: number;
@@ -82,27 +176,39 @@ export declare class RTCPeerConnection extends EventTarget {
     }>): RTCDataChannel;
     removeTrack(sender: RTCRtpSender): void;
     private needNegotiation;
+    private invalidateLastCreatedDescriptions;
+    private waitForPendingDescriptionTask;
     private findOrCreateTransport;
-    setLocalDescription(sessionDescription?: {
-        type: "offer" | "answer";
-        sdp: string;
-    }): Promise<SessionDescription>;
+    setLocalDescription(sessionDescription: {
+        type: "rollback";
+    }): Promise<void>;
+    setLocalDescription(sessionDescription?: RTCLocalSessionDescriptionInit): Promise<SessionDescription>;
     private gatherCandidates;
-    addIceCandidate(candidateMessage: RTCIceCandidate | RTCIceCandidateInit): Promise<void>;
+    addIceCandidate(candidateMessage?: RTCIceCandidate | RTCIceCandidateInit | null): Promise<void>;
+    private applyRemoteIceCandidate;
+    private flushPendingRemoteCandidates;
     private connect;
     restartIce(): void;
     setRemoteDescription(sessionDescription: RTCSessionDescriptionInit): Promise<void>;
     addTransceiver(trackOrKind: Kind | MediaStreamTrack, options?: Partial<TransceiverOptions>): RTCRtpTransceiver;
-    addTrack(track: MediaStreamTrack, 
-    /**todo impl */
-    ms?: MediaStream): RTCRtpSender;
-    createAnswer(): Promise<import("./sdp").RTCSessionDescription>;
+    addTrack(track: MediaStreamTrack, ...streams: MediaStream[]): RTCRtpSender;
+    createAnswer(): Promise<RTCSessionDescription>;
     private assertNotClosed;
     private setSignalingState;
     private createPeerConnectionStats;
     getStats(selector?: MediaStreamTrack | null): Promise<RTCStatsReport>;
     close(): Promise<void>;
 }
+export type DebugConfig = Partial<{
+    /**% */
+    inboundPacketLoss: number;
+    /**% */
+    outboundPacketLoss: number;
+    /**ms */
+    receiverReportDelay: number;
+    disableSendNack: boolean;
+    disableRecvRetransmit: boolean;
+}>;
 export interface PeerConfig {
     codecs: Partial<{
         /**
@@ -117,6 +223,8 @@ export interface PeerConfig {
         video: RTCRtpHeaderExtensionParameters[];
     }>;
     iceTransportPolicy: "all" | "relay";
+    /** Advertise local ICE lite and operate in the controlled role. */
+    iceLite: boolean;
     iceServers: RTCIceServer[];
     /**Minimum port and Maximum port must not be the same value */
     icePortRange: [number, number] | undefined;
@@ -127,6 +235,10 @@ export interface PeerConfig {
     iceAdditionalHostAddresses: string[] | undefined;
     iceUseIpv4: boolean;
     iceUseIpv6: boolean;
+    iceUseTcp: boolean;
+    turnTransport: "udp" | "tcp" | "tls" | undefined;
+    turnTlsOptions: TlsConnectionOptions | undefined;
+    /** @deprecated Prefer turn URL transport parameters or turnTransport. */
     forceTurnTCP: boolean;
     /** such as google cloud run */
     iceUseLinkLocalAddress: boolean | undefined;
@@ -139,36 +251,57 @@ export interface PeerConfig {
     }>;
     icePasswordPrefix: string | undefined;
     bundlePolicy: BundlePolicy;
-    debug: Partial<{
-        /**% */
-        inboundPacketLoss: number;
-        /**% */
-        outboundPacketLoss: number;
-        /**ms */
-        receiverReportDelay: number;
-        disableSendNack: boolean;
-        disableRecvRetransmit: boolean;
-    }>;
+    rtcpMuxPolicy: "require";
+    iceCandidatePoolSize: number;
+    certificates: RTCCertificate[];
+    debug: DebugConfig;
     midSuffix: boolean;
     /** Advertised local SCTP max-message-size in SDP. Use 0 for unlimited. */
     maxMessageSize: number;
 }
 export declare const findCodecByMimeType: (codecs: RTCRtpCodecParameters[], target: RTCRtpCodecParameters) => RTCRtpCodecParameters | undefined;
 export type RTCIceServer = {
-    urls: string;
+    urls: string | string[];
     username?: string;
     credential?: string;
 };
+export type RTCBundlePolicy = "balanced" | "max-compat" | "max-bundle";
+export type RTCRtcpMuxPolicy = "require";
+export interface RTCConfiguration {
+    iceServers?: RTCIceServer[];
+    iceTransportPolicy?: PeerConfig["iceTransportPolicy"];
+    bundlePolicy?: RTCBundlePolicy;
+    rtcpMuxPolicy?: RTCRtcpMuxPolicy;
+    iceCandidatePoolSize?: number;
+    certificates?: RTCCertificate[];
+}
+export interface RTCLocalSessionDescriptionInit extends RTCSessionDescriptionInit {
+    type?: Exclude<RTCSessionDescriptionInit["type"], "rollback"> | "rollback";
+}
+type RTCPeerConnectionRTCConfiguration = Omit<RTCConfiguration, "bundlePolicy"> & {
+    bundlePolicy?: PeerConfig["bundlePolicy"] | RTCBundlePolicy;
+};
+export type RTCPeerConnectionConfig = Partial<Omit<PeerConfig, "bundlePolicy" | "rtcpMuxPolicy" | "iceCandidatePoolSize" | "certificates">> & RTCPeerConnectionRTCConfiguration;
 export declare const defaultPeerConfig: PeerConfig;
-export interface RTCTrackEvent {
-    track: MediaStreamTrack;
-    streams: MediaStream[];
-    transceiver: RTCRtpTransceiver;
-    receiver: RTCRtpReceiver;
+export declare class RTCTrackEvent {
+    readonly type = "track";
+    readonly track: MediaStreamTrack;
+    readonly streams: MediaStream[];
+    readonly transceiver: RTCRtpTransceiver;
+    readonly receiver: RTCRtpReceiver;
+    constructor(init: {
+        track: MediaStreamTrack;
+        streams: MediaStream[];
+        transceiver: RTCRtpTransceiver;
+        receiver: RTCRtpReceiver;
+    });
 }
 export interface RTCDataChannelEvent {
+    type?: "datachannel";
     channel: RTCDataChannel;
 }
 export interface RTCPeerConnectionIceEvent {
+    type?: "icecandidate";
     candidate?: RTCIceCandidate;
 }
+export {};