feat: bridge_server livekit
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0Xiao0
194
main/test_client_wav.py
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194
main/test_client_wav.py
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import asyncio
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import os
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import sys
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import wave
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import time
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from dotenv import load_dotenv
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# Try to load credentials from .env.local
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load_dotenv(".env.local")
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from livekit import rtc
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async def publish_audio_from_wav(room: rtc.Room, wav_path: str):
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print(f"🎵 准备加载音频文件: {wav_path}")
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if not os.path.exists(wav_path):
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print(f"❌ 找不到文件 {wav_path}")
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return
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with wave.open(wav_path, "rb") as wf:
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sample_rate = wf.getframerate()
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num_channels = wf.getnchannels()
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sampwidth = wf.getsampwidth()
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if sampwidth != 2:
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print("❌ 错误：只支持 16-bit PCM (S16LE) 编码的 WAV 文件。")
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return
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print(f"📊 音频信息: 采样率 {sample_rate}Hz, 通道数 {num_channels}")
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source = rtc.AudioSource(sample_rate, num_channels)
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track = rtc.LocalAudioTrack.create_audio_track("agent_input_audio", source)
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options = rtc.TrackPublishOptions()
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options.source = rtc.TrackSource.SOURCE_MICROPHONE
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await room.local_participant.publish_track(track, options)
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print("✅ 成功发布麦克风音频轨道，开始推流...")
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# 每次读取并推送 20ms 的数据
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chunk_duration_ms = 20
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samples_per_chunk = int(sample_rate * (chunk_duration_ms / 1000.0))
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bytes_per_chunk = samples_per_chunk * num_channels * sampwidth
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while True:
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data = wf.readframes(samples_per_chunk)
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if not data:
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break
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# 根据已读取的数据计算出完整的采样数（最后一帧可能不足 20ms）
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frame_samples = len(data) // (num_channels * sampwidth)
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# 使用 LiveKit SDK 封装 AudioFrame
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try:
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# LiveKit Python SDK version >= 0.15
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audio_frame = rtc.AudioFrame(data, sample_rate, num_channels, frame_samples)
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await source.capture_frame(audio_frame)
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except TypeError:
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# 若抛出 TypeError，可能 SDK 版本有差异，尝试旧版本 API 或者直接 copy
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frame = rtc.AudioFrame.create(sample_rate, num_channels, frame_samples)
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frame.data[:] = data
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await source.capture_frame(frame)
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# 严格控制发送速率，避免瞬时把整个音频发过去而导致对面不识别（模拟真实发音）
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await asyncio.sleep(chunk_duration_ms / 1000.0)
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print("🎉 录音流推送完毕！等待 Agent 回复中...")
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async def save_audio_stream(track: rtc.RemoteAudioTrack):
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# 为避免旧文件冲突，加个时间戳
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filename = f"agent_response_{int(time.time())}.wav"
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print(f"🎙️ 正在将 Agent 的声音写入文件: {filename}")
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stream = rtc.AudioStream(track)
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wf = None
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try:
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async for event in stream:
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# 接收到第一帧时初始化 WAV 格式
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if wf is None:
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wf = wave.open(filename, "wb")
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wf.setnchannels(event.frame.num_channels)
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wf.setsampwidth(2) # 16-bit
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wf.setframerate(event.frame.sample_rate)
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# 写入当前帧音频数据 (16-bit PCM)
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wf.writeframes(bytes(event.frame.data))
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except Exception as e:
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print(f"音频流断开或出错: {e}")
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finally:
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if wf is not None:
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wf.close()
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print(f"💾 Agent 语音结果已成功保存到: {filename}")
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async def main(room_url: str, token: str, wav_path: str):
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room = rtc.Room()
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agent_ready = asyncio.Event()
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@room.on("connected")
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def on_connected():
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print("✅ 成功连接到 LiveKit 房间")
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# 如果连接时房间里已经有 Agent（远程参与者），直接准备触发
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if room.remote_participants:
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agent_ready.set()
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@room.on("participant_connected")
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def on_participant_connected(participant: rtc.RemoteParticipant):
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print(f"👋 Agent ({participant.identity}) 已加入房间")
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agent_ready.set()
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@room.on("data_received")
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def on_data_received(data_packet: rtc.DataPacket):
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identity = data_packet.participant.identity if data_packet.participant else "未知"
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try:
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print(f"📩 [数据接收 | {identity}]: {data_packet.data.decode('utf-8')}")
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except Exception:
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pass
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@room.on("transcription_received")
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def on_transcription_received(
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segments: list[rtc.TranscriptionSegment],
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participant: rtc.Participant,
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track_pub: rtc.TrackPublication
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):
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identity = participant.identity if participant else "未知"
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for segment in segments:
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status = "✅ 最终结果" if segment.final else "⏳ 正在思考/中间结果"
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print(f"🗣️ [{status} | {identity}]: {segment.text}")
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@room.on("track_subscribed")
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def on_track_subscribed(
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track: rtc.Track,
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publication: rtc.RemoteTrackPublication,
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participant: rtc.RemoteParticipant
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):
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# 当 Agent 发出新的声音（音频轨道）时，我们订阅并保存
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if track.kind == rtc.TrackKind.KIND_AUDIO:
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asyncio.create_task(save_audio_stream(track))
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print("⏳ 正在建立连接...")
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await room.connect(room_url, token)
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print("⏳ 等待 Agent 初始化并加入房间...")
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await agent_ready.wait()
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# 稍微延迟半秒钟，确保 Agent 侧面的准备（如模型加载等）一切就绪
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await asyncio.sleep(0.5)
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# 开始推送本地 wav 音频
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asyncio.create_task(publish_audio_from_wav(room, wav_path))
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try:
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await asyncio.Event().wait()
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except KeyboardInterrupt:
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print("\n断开连接中...")
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finally:
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await room.disconnect()
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if __name__ == "__main__":
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if len(sys.argv) < 2:
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print("❌ 用法: python test_client_wav.py <WAV文件路径> [LIVEKIT_URL] [LIVEKIT_TOKEN/API_KEY]")
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print("说明:\n1. 必须提供 WAV 路径。")
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print("2. 自动从 .env.local 读取 LIVEKIT_URL。并在没有提供 Token 时自动向 localhost:8000/getToken 请求。")
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sys.exit(1)
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wav_file = sys.argv[1]
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url = os.getenv("LIVEKIT_URL")
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token = None
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if len(sys.argv) >= 4:
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url = sys.argv[2]
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token = sys.argv[3]
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if not token:
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import urllib.request
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import json
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import random
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# 每次使用随机的测试房间，防止上一次没退出的 agent 堆积在同一个房间里导致多重回复
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unique_room = f"test-room-{random.randint(1000, 9999)}"
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print(f"🔄 正在通过本地服务获取 Token，请求加入全新独立房间: {unique_room} ...")
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try:
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req = urllib.request.urlopen(f"http://localhost:8000/getToken?room={unique_room}&identity=python_tester&agent_name=my-agent")
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res_body = req.read().decode('utf-8')
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data = json.loads(res_body)
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token = data.get("token")
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print("✅ 成功获取了包含了 Agent dispatch 的临时 Token！")
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except Exception as e:
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print(f"❌ 获取 Token 失败，错误信息: {e}")
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print("若本地 token 服务未启动，请手动提供有效的测试 token")
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sys.exit(1)
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if not url or not token:
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print("❌ 缺少 LiveKit URL 或 Token")
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sys.exit(1)
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asyncio.run(main(url, token, wav_file))
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