Silent Emergency Live-Audio

// ─── MODUL 1: WebRTC VOICE-STREAM · Stealth-Mode ────────────────
// Latenz-Ziel: < 250ms End-to-End
// STUN/TURN: SEC.SI eigene Infrastruktur (keine Google-Server)

MODULE SilentWebRTCStream:

  CONST ICE_SERVERS = [
    { urls: "stun:stun.sec-si.de:3478" },
    { urls: "turn:turn.sec-si.de:3478",
      username: SESSION.turn_user,
      credential: SESSION.turn_pass }
  ]

  CONST AUDIO_CONSTRAINTS = {
    sampleRate:       16000,
    channelCount:     1,            // Mono
    echoCancellation: false,        // Keine Echo-Änderung → originaler Ton
    noiseSuppression: false,        // Eigener Filter aus Modul 2
    autoGainControl:  false,        // Konstante Lautstärke für Beweise
    latency:          0.01          // 10ms Puffer-Anforderung
  }

  STATE pc          = null          // RTCPeerConnection
  STATE mediaStream = null
  STATE statsTimer  = null

  FUNCTION init(sessionToken):
    // getUserMedia im Hintergrund — kein UI-Indicator
    mediaStream = await navigator.mediaDevices.getUserMedia({
      audio: AUDIO_CONSTRAINTS,
      video: false                  // Kein Video — nur Audio
    })

    pc = new RTCPeerConnection({ iceServers: ICE_SERVERS })

    // STEALTH: MediaStream Track unsichtbar hinzufügen
    track = mediaStream.getAudioTracks()[0]
    track.contentHint = "speech"    // Optimierungshinweis für Codec
    pc.addTrack(track, mediaStream)

    // Adaptive Bitrate vorbereiten (Modul 2)
    sender = pc.getSenders()[0]
    AdaptiveBitrate.attach(sender)

    // SDP Offer → SEC.SI Signaling-Server
    offer  = await pc.createOffer({ offerToReceiveAudio: false })
    await pc.setLocalDescription(offer)

    answer = await SignalingServer.exchange(offer, sessionToken)
    await pc.setRemoteDescription(answer)

    // Stats-Monitoring starten
    statsTimer = setInterval(() => AdaptiveBitrate.monitor(pc), 2000)

  FUNCTION injectPreBuffer(pcmData: Float32Array):
    // Pre-Trigger-Audio als erste Pakete in den Stream einspeisen
    // via AudioContext → MediaStreamDestination → RTCPeerConnection
    ctx      = new AudioContext({ sampleRate: 16000 })
    buffer   = ctx.createBuffer(1, pcmData.length, 16000)
    buffer.copyToChannel(pcmData, 0)

    source   = ctx.createBufferSource()
    source.buffer = buffer

    dest     = ctx.createMediaStreamDestination()
    source.connect(dest)
    source.start(0)

    // Track austauschen: Pre-Buffer → Live-Mic
    newTrack = dest.stream.getAudioTracks()[0]
    sender   = pc.getSenders()[0]
    await sender.replaceTrack(newTrack)

    // Nach Pre-Buffer-Abspielen: zurück zu Live-Mic
    source.onended = () => sender.replaceTrack(
      mediaStream.getAudioTracks()[0]
    )

  FUNCTION stop():
    clearInterval(statsTimer)
    mediaStream?.getTracks().forEach(t => t.stop())
    pc?.close()
    pc = null