view dictation/webrtc_smoke.py @ 279:b3b547563ec7

Add Google connector service and agent wiki Implement the C/Seobeo Google Drive and Gmail connector with encrypted OAuth storage, Zenbu authentication, browser testing, AI tool discovery, chunked HTTP decoding, and Bazel coverage. Consolidate repository guidance into progressive wiki documentation and enforce arena-first allocation for new first-party C code. Co-authored-by: Copilot <[email protected]> Copilot-Session: 84c338fd-0939-4bb3-b7f3-1062eb213e5d
author MrJuneJune <me@mrjunejune.com>
date Mon, 17 Aug 2026 22:22:36 -0700
parents 78699f810817
children
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from __future__ import annotations

import argparse
import asyncio
import json

from aiortc import RTCPeerConnection, RTCSessionDescription
from aiortc.contrib.media import MediaPlayer
import httpx


async def run(server_url: str, audio_path: str, timeout: float) -> None:
    peer = RTCPeerConnection()
    channel = peer.createDataChannel("transcripts")
    player = MediaPlayer(audio_path)
    if player.audio is None:
        raise RuntimeError("Input file does not contain an audio track")
    peer.addTrack(player.audio)
    final_received = asyncio.Event()
    session_id = None

    @channel.on("message")
    def on_message(message):
        event = json.loads(message)
        print(json.dumps(event, ensure_ascii=False), flush=True)
        if event.get("type") == "transcript.final":
            final_received.set()

    try:
        offer = await peer.createOffer()
        await peer.setLocalDescription(offer)
        async with httpx.AsyncClient(timeout=timeout) as client:
            response = await client.post(
                f"{server_url}/api/webrtc/offer",
                json={
                    "sdp": peer.localDescription.sdp,
                    "type": peer.localDescription.type,
                },
            )
            response.raise_for_status()
            answer = response.json()
            session_id = answer["sessionId"]
            await peer.setRemoteDescription(
                RTCSessionDescription(
                    sdp=answer["sdp"],
                    type=answer["type"],
                )
            )
            await asyncio.wait_for(final_received.wait(), timeout=timeout)
            await client.post(
                f"{server_url}/api/webrtc/session/{session_id}/close"
            )
    finally:
        await peer.close()


def main() -> None:
    parser = argparse.ArgumentParser()
    parser.add_argument("audio")
    parser.add_argument(
        "--server",
        default="http://127.0.0.1:8090",
    )
    parser.add_argument("--timeout", type=float, default=60.0)
    args = parser.parse_args()
    asyncio.run(run(args.server.rstrip("/"), args.audio, args.timeout))


if __name__ == "__main__":
    main()