{
  "context": "This criteria evaluates how effectively the engineer uses aiortc's RTCRtpSender API and related WebRTC components to transmit audio streams, focusing on proper sender lifecycle management, track replacement, statistics retrieval, and capability querying.",
  "type": "weighted_checklist",
  "checklist": [
    {
      "name": "RTCPeerConnection initialization",
      "description": "Creates an RTCPeerConnection instance to establish the WebRTC peer connection for audio transmission",
      "max_score": 10
    },
    {
      "name": "addTrack usage",
      "description": "Uses RTCPeerConnection.addTrack() method to add the audio track and obtain an RTCRtpSender instance",
      "max_score": 15
    },
    {
      "name": "SDP offer creation",
      "description": "Calls createOffer() and setLocalDescription() to generate and set the local SDP for connection negotiation",
      "max_score": 10
    },
    {
      "name": "Sender reference storage",
      "description": "Stores reference to the RTCRtpSender returned by addTrack() for later operations like track replacement and statistics retrieval",
      "max_score": 10
    },
    {
      "name": "replaceTrack method",
      "description": "Uses RTCRtpSender.replaceTrack() method to dynamically replace the audio track without renegotiation",
      "max_score": 20
    },
    {
      "name": "getStats method",
      "description": "Calls RTCRtpSender.getStats() method to retrieve transmission statistics",
      "max_score": 15
    },
    {
      "name": "Statistics extraction",
      "description": "Extracts packetsSent and bytesSent from the RTCStatsReport returned by getStats() to provide transmission metrics",
      "max_score": 10
    },
    {
      "name": "getCapabilities method",
      "description": "Uses RTCRtpSender.getCapabilities('audio') class method to query supported audio codecs and capabilities",
      "max_score": 10
    }
  ]
}