import asyncio
import requests
from livekit import rtc

import wave
import numpy as np
from livekit.rtc import AudioSource, AudioFrame, LocalAudioTrack

TOKEN_URL = "http://localhost:8000/getToken"
WS_URL = "wss://esp32-vt80c4y6.livekit.cloud"  # 你的 LiveKit Server 地址

ROOM_NAME = "test-room20"
import uuid
IDENTITY = f"uv-{uuid.uuid4().hex[:6]}"
# IDENTITY = "test-user0"


def get_token():
    resp = requests.get(
        TOKEN_URL,
        params={
            "room": ROOM_NAME,
            "identity": IDENTITY,
            "agent_name": "my-agent",  # 关键！！！
        },
    )
    data = resp.json()
    return data["token"]


async def main():
    token = get_token()

    room = rtc.Room()

    @room.on("participant_connected")
    def on_participant_connected(participant):
        print(f"✅ 有人加入房间: {participant.identity}")

    @room.on("participant_disconnected")
    def on_participant_disconnected(participant):
        print(f"❌ 有人离开房间: {participant.identity}")

    print("🔌 正在连接房间...")
    await room.connect(WS_URL, token)

    print("✅ 已连接房间:", ROOM_NAME)
    print("当前房间成员：")
    for p in room.remote_participants.values():
        print(" -", p.identity)

    @room.on("data_received")
    def on_data_received(data, participant, kind, topic):
        try:
            msg = data.decode()
            print(f"📩 来自 {participant.identity}: {msg}")
        except:
            print("📩 收到二进制数据")

    @room.on("track_subscribed")
    def on_track_subscribed(track, publication, participant):
        print(f"🎧 订阅轨道: {participant.identity}")

        if track.kind == rtc.TrackKind.KIND_AUDIO:
            print("👉 TTS 音频来了")

    # 等一下确保连接稳定
    await asyncio.sleep(1)
    await room.local_participant.publish_data(
        b"hello",
        reliable=True,
        topic="chat"
    )
    # 上传 wav
    await publish_wav(room, "2food.wav")

    await room.disconnect()


async def publish_wav(room, wav_path):
    print("🎵 开始上传本地 wav:", wav_path)

    wf = wave.open(wav_path, "rb")

    sample_rate = wf.getframerate()
    num_channels = wf.getnchannels()
    sample_width = wf.getsampwidth()

    print(f"📊 WAV信息: {sample_rate}Hz, {num_channels}ch, {sample_width*8}bit")

    # 创建音频源
    source = AudioSource(sample_rate, num_channels)

    # 创建本地音轨
    track = LocalAudioTrack.create_audio_track("mic", source)

    # 发布轨道
    await room.local_participant.publish_track(track)
    print("📡 已发布音轨")

    frame_duration = 0.02  # 20ms
    samples_per_frame = int(sample_rate * frame_duration)

    while True:
        data = wf.readframes(samples_per_frame)
        if not data:
            break

        # 用于计算长度
        audio = np.frombuffer(data, dtype=np.int16)

        if len(audio) == 0:
            continue

        samples_per_channel = len(audio) // num_channels

        frame = AudioFrame(
            data=data,  # ✅ 关键：用 bytes
            sample_rate=sample_rate,
            num_channels=num_channels,
            samples_per_channel=samples_per_channel,
        )

        await source.capture_frame(frame)
        await asyncio.sleep(frame_duration)
    print("✅ wav 推流结束")


if __name__ == "__main__":
    asyncio.run(main())