130 lines
3.3 KiB
Python
130 lines
3.3 KiB
Python
import asyncio
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import requests
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from livekit import rtc
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import wave
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import numpy as np
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from livekit.rtc import AudioSource, AudioFrame, LocalAudioTrack
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TOKEN_URL = "http://localhost:8000/getToken"
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WS_URL = "wss://esp32-vt80c4y6.livekit.cloud" # 你的 LiveKit Server 地址
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ROOM_NAME = "test-room20"
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import uuid
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IDENTITY = f"uv-{uuid.uuid4().hex[:6]}"
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# IDENTITY = "test-user0"
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def get_token():
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resp = requests.get(
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TOKEN_URL,
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params={
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"room": ROOM_NAME,
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"identity": IDENTITY,
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"agent_name": "my-agent", # 关键！！！
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},
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)
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data = resp.json()
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return data["token"]
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async def main():
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token = get_token()
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room = rtc.Room()
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@room.on("participant_connected")
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def on_participant_connected(participant):
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print(f"✅ 有人加入房间: {participant.identity}")
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@room.on("participant_disconnected")
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def on_participant_disconnected(participant):
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print(f"❌ 有人离开房间: {participant.identity}")
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print("🔌 正在连接房间...")
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await room.connect(WS_URL, token)
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print("✅ 已连接房间:", ROOM_NAME)
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print("当前房间成员：")
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for p in room.remote_participants.values():
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print(" -", p.identity)
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@room.on("data_received")
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def on_data_received(data, participant, kind, topic):
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try:
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msg = data.decode()
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print(f"📩 来自 {participant.identity}: {msg}")
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except:
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print("📩 收到二进制数据")
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@room.on("track_subscribed")
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def on_track_subscribed(track, publication, participant):
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print(f"🎧 订阅轨道: {participant.identity}")
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if track.kind == rtc.TrackKind.KIND_AUDIO:
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print("👉 TTS 音频来了")
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# 等一下确保连接稳定
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await asyncio.sleep(1)
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await room.local_participant.publish_data(
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b"hello",
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reliable=True,
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topic="chat"
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)
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# 上传 wav
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await publish_wav(room, "2food.wav")
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await room.disconnect()
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async def publish_wav(room, wav_path):
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print("🎵 开始上传本地 wav:", wav_path)
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wf = wave.open(wav_path, "rb")
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sample_rate = wf.getframerate()
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num_channels = wf.getnchannels()
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sample_width = wf.getsampwidth()
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print(f"📊 WAV信息: {sample_rate}Hz, {num_channels}ch, {sample_width*8}bit")
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# 创建音频源
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source = AudioSource(sample_rate, num_channels)
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# 创建本地音轨
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track = LocalAudioTrack.create_audio_track("mic", source)
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# 发布轨道
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await room.local_participant.publish_track(track)
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print("📡 已发布音轨")
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frame_duration = 0.02 # 20ms
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samples_per_frame = int(sample_rate * frame_duration)
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while True:
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data = wf.readframes(samples_per_frame)
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if not data:
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break
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# 用于计算长度
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audio = np.frombuffer(data, dtype=np.int16)
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if len(audio) == 0:
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continue
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samples_per_channel = len(audio) // num_channels
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frame = AudioFrame(
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data=data, # ✅ 关键：用 bytes
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sample_rate=sample_rate,
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num_channels=num_channels,
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samples_per_channel=samples_per_channel,
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)
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await source.capture_frame(frame)
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await asyncio.sleep(frame_duration)
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print("✅ wav 推流结束")
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if __name__ == "__main__":
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asyncio.run(main()) |