ylhan
74a30b11d7
新增实验代码 - services/bot_server.py：Pipecat SmallWebRTC 服务端， 把 ConversationEngine 包成一个 FrameProcessor（检验 A 方案的核心假设） - client/test_client.py：自写 aiortc 客户端——Pipecat 的客户端 SDK 覆盖浏览器/移动端/C++/ESP32，没有适配"Linux x86 盒子接 USB 阵列"这种形态 实验结果 - ✅ 传输层跑通：信令（47–280ms 往返）、ICE 连通、data channel、 双向媒体轨（客户端音轨送达服务端，服务端音轨客户端收到） - ❌ 帧路由未通：自定义处理器的 process_frame 调用次数为 0， 连自己注入的 StartFrame 都没进去，引擎从未运行 - 现象：连接全对、音轨协商成功、助手永远不说话，日志里一条错误都没有 对架构决策的影响 A 方案（Pipecat 只做传输、逻辑归自有引擎）的核心假设不是开箱即用的。 B 方案按框架原生 service 重构，绕开了这个接缝。 根因未定位：接线错误（接缝浅，A 可行）vs 模型不兼容（接缝深，应转 B）。 顺带挖出 8 个 Pipecat 集成坑，每个都表现为"什么都没发生"，已记入 README： 客户端需发 data channel ping、offer 无媒体则不收集 ICE 候选、 run() 要 WorkerParams、TaskManager 需在事件循环内建、 后台任务异常被静默吞掉等。 其他 - bot_server.py 保留 [诊断] 前缀日志，供下一轮排查，定位后应清理 - 客户端默认地址改用 tailscale 域名（局域网已于当日重新编址， 从 192.168.0.x 变为 172.16.1.x） - 新增依赖 aiortc（含 av 17.x，注意 18.x 与 aiortc 1.15 不兼容）
339 lines
13 KiB
Python
339 lines
13 KiB
Python
#!/usr/bin/env python3
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"""最小 aiortc 客户端（实验用）。
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**为什么需要它**：Pipecat 的官方客户端 SDK 覆盖浏览器 / 移动端 / C++ / ESP32，
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唯独没有适配"跑 Linux 的 x86 盒子接 USB 麦克风阵列"——也就是 N5095 的形态。
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所以客户端必须自己写，这个文件是雏形。
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它同时是验证工具：不依赖浏览器就能测服务端。用法：
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# 发一段 wav，收回应答音频
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python client/test_client.py --server http://<服务器>:7860 \
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--wav p0/stt_tts_bench/wavs/stt_1.wav --out /tmp/reply.wav
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# 只做连接测试，不发音频
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python client/test_client.py --server http://<服务器>:7860 --connect-only
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**它还不是什么**：不做 AEC、不做播客参考、不做打断（那些要真实声卡和硬件），
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只用文件音频验证信令与媒体通路。
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"""
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from __future__ import annotations
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import argparse
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import asyncio
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import fractions
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import json
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import logging
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import sys
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import time
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import urllib.request
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import wave
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from pathlib import Path
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import numpy as np
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from aiortc import (
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RTCConfiguration,
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RTCPeerConnection,
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RTCSessionDescription,
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)
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from aiortc.mediastreams import AudioStreamTrack, MediaStreamError
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logging.basicConfig(level=logging.INFO, format="%(asctime)s %(levelname)s %(message)s")
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log = logging.getLogger("client")
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class SilentTrack(AudioStreamTrack):
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"""静音音轨。只为触发 ICE 候选收集，不发真实音频。"""
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kind = "audio"
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def __init__(self, sample_rate: int = 16000):
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super().__init__()
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self.sample_rate = sample_rate
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self._n = 0
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async def recv(self):
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import fractions as _fr
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from av import AudioFrame as _AF
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await asyncio.sleep(0.02)
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pcm = np.zeros(self.sample_rate // 50, dtype=np.int16)
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f = _AF(format="s16", layout="mono", samples=len(pcm))
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f.planes[0].update(pcm.tobytes())
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f.sample_rate = self.sample_rate
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f.pts = self._n
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f.time_base = _fr.Fraction(1, self.sample_rate)
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self._n += len(pcm)
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return f
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class FileAudioTrack(AudioStreamTrack):
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"""把 wav 文件当作麦克风音源，按实时速率吐出来。
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实时速率很重要——如果一次性灌完，服务端的 VAD 会把它当成
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一段超长的连续语音，测出来的延迟也不真实。
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"""
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kind = "audio"
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def __init__(self, wav_path: Path, *, sample_rate: int = 16000):
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super().__init__()
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with wave.open(str(wav_path)) as w:
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assert w.getnchannels() == 1, "需要单声道"
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self.src_rate = w.getframerate()
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pcm = np.frombuffer(w.readframes(w.getnframes()), dtype=np.int16)
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self.samples = pcm.astype(np.float32) / 32768.0
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self.sample_rate = sample_rate
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if self.src_rate != sample_rate:
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n = int(len(self.samples) * sample_rate / self.src_rate)
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self.samples = np.interp(np.linspace(0, len(self.samples) - 1, n),
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np.arange(len(self.samples)), self.samples)
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self._pos = 0
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self._t0: float | None = None
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self._frame_samples = sample_rate // 100 # 10ms
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async def recv(self):
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"""按 10ms 一块、以真实时间节奏产出。"""
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if self._t0 is None:
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self._t0 = time.perf_counter()
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self._pos = 0
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# 音频放完后持续吐静音，让服务端的 VAD 能判到"说完"
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if self._pos >= len(self.samples):
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audio = np.zeros(self._frame_samples, dtype=np.float32)
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else:
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audio = self.samples[self._pos:self._pos + self._frame_samples]
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if len(audio) < self._frame_samples:
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audio = np.pad(audio, (0, self._frame_samples - len(audio)))
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self._pos += self._frame_samples
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# 实时节奏
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target = self._t0 + self._pos / self.sample_rate
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delay = target - time.perf_counter()
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if delay > 0:
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await asyncio.sleep(delay)
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pcm = (np.clip(audio, -1, 1) * 32767).astype(np.int16)
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from av import AudioFrame
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f = AudioFrame(format="s16", layout="mono", samples=len(pcm))
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f.planes[0].update(pcm.tobytes())
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f.sample_rate = self.sample_rate
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f.pts = int(self._pos * 90000 / self.sample_rate)
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f.time_base = fractions.Fraction(1, 90000)
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return f
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@property
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def finished(self) -> bool:
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return self._pos >= len(self.samples)
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class ReplyRecorder:
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"""收服务端音频，写 wav，并记录首音延迟。"""
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def __init__(self, out_path: Path, sample_rate: int = 24000):
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self.out_path = out_path
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self.sample_rate = sample_rate
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self.chunks: list[np.ndarray] = []
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self.first_at: float | None = None
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self.t0 = time.perf_counter()
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def feed(self, frame) -> None:
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if self.first_at is None:
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self.first_at = time.perf_counter()
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arr = frame.to_ndarray()
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# 多声道取第一路
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if arr.shape[0] > 1:
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arr = arr[0:1]
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self.chunks.append(arr.astype(np.float32).reshape(-1) / 32768.0)
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def save(self) -> float:
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if not self.chunks:
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return 0.0
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audio = np.concatenate(self.chunks)
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pcm = (np.clip(audio, -1, 1) * 32767).astype(np.int16)
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self.out_path.parent.mkdir(parents=True, exist_ok=True)
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with wave.open(str(self.out_path), "wb") as w:
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w.setnchannels(1)
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w.setsampwidth(2)
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w.setframerate(self.sample_rate)
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w.writeframes(pcm.tobytes())
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return len(audio) / self.sample_rate
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@property
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def first_audio_ms(self) -> float | None:
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return None if self.first_at is None else (self.first_at - self.t0) * 1000
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def post_offer(server: str, sdp: str, type_: str) -> dict:
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"""把 offer 发给服务端，取回 answer。"""
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payload = json.dumps({"sdp": sdp, "type": type_, "pc_id": None,
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"restart_pc": False, "request_data": None}).encode()
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req = urllib.request.Request(f"{server.rstrip('/')}/api/offer", data=payload,
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headers={"Content-Type": "application/json"},
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method="POST")
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with urllib.request.urlopen(req, timeout=30) as r:
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return json.loads(r.read().decode())
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async def run(server: str, wav: Path | None, out: Path, *, connect_only: bool,
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timeout_s: float) -> int:
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pc = RTCPeerConnection(RTCConfiguration(iceServers=[]))
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recorder = ReplyRecorder(out)
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track = None
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got_audio = asyncio.Event()
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@pc.on("connectionstatechange")
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async def _state():
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log.info("连接状态: %s", pc.connectionState)
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if pc.connectionState in ("failed", "closed"):
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got_audio.set()
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@pc.on("track")
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def _track(t):
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log.info("收到服务端音轨: %s", t.kind)
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async def _pump():
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try:
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while True:
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frame = await t.recv()
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recorder.feed(frame)
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if not got_audio.is_set() and recorder.chunks:
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pass
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except (MediaStreamError, asyncio.CancelledError):
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pass
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finally:
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got_audio.set()
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asyncio.ensure_future(_pump())
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if wav and not connect_only:
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track = FileAudioTrack(wav)
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pc.addTrack(track)
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log.info("音源: %s（%d 采样点）", wav.name, len(track.samples))
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else:
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# --connect-only 也要加一条（静音）音轨。
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# **不能省**：aiortc 的 offer 里没有媒体时完全不收集 ICE 候选，
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# 现象是"信令成功、SDP 交换正常、但 ICE 永远停在 new"。
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pc.addTrack(SilentTrack())
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log.info("连接测试模式：已加静音音轨（仅为触发 ICE 收集）")
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# **必须建 data channel 并定期发 "ping"。**
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# Pipecat 的 SmallWebRTC 用 data channel 上的消息来判断"客户端还活着"：
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# if isinstance(message, str) and message.startswith("ping"):
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# self._last_received_time = time.time()
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# 而 connection 的 connected 事件、进而 on_client_connected 回调、
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# 进而 StartFrame（音频推流的开关）都依赖它。
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# 不发 ping 的现象是：ICE 连通、音轨协商成功、但**服务端一帧音频都不推**、
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# 助手永远不说话，而且日志里只有一句 "Data channel not established within 10s"。
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channel = pc.createDataChannel("pipecat")
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dc_state = {"ok": False}
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@channel.on("open")
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def _dc_open():
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dc_state["ok"] = True
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log.info("data channel 已打开，开始发心跳")
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async def _keepalive():
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while pc.connectionState not in ("closed", "failed"):
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if dc_state["ok"]:
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try:
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channel.send("ping")
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except Exception: # noqa: BLE001
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pass
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await asyncio.sleep(1.0)
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asyncio.ensure_future(_keepalive())
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offer = await pc.createOffer()
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await pc.setLocalDescription(offer)
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# 等 ICE 收集完成再发 offer——**这步不能省**。
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# 不等的话 SDP 里没有任何候选地址，服务端拿到的是一份"空地址"的 offer，
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# 信令看着成功，但 ICE 永远连不上（状态停在 new）。
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# 这叫非 trickle ICE：把候选都塞进 SDP 一次发完。
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# 真实产品会用 trickle（边收集边发候选），但那样需要额外的信令端点。
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for _ in range(100):
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if pc.iceGatheringState == "complete":
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break
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await asyncio.sleep(0.1)
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n_cand = pc.localDescription.sdp.count("candidate:")
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log.info("ICE 收集: %s，SDP 含 %d 个候选", pc.iceGatheringState, n_cand)
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if n_cand == 0:
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log.warning("SDP 里没有候选地址，ICE 很可能连不上")
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log.info("本地 SDP 已生成，发给服务端…")
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t_send = time.perf_counter()
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answer = post_offer(server, pc.localDescription.sdp, pc.localDescription.type)
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if answer.get("type") not in ("answer", None):
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log.error("服务端返回异常: %s", answer)
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return 1
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if answer.get("sdp"):
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await pc.setRemoteDescription(RTCSessionDescription(
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sdp=answer["sdp"], type=answer["type"]))
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log.info("远端 SDP 已设置（往返 %.0f ms）", (time.perf_counter() - t_send) * 1000)
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else:
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log.info("服务端未返回 SDP（可能是 pc_id 复用）: %s", answer)
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# 等 ICE 连通
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for _ in range(100):
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if pc.connectionState in ("connected", "completed"):
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break
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if pc.connectionState in ("failed", "closed"):
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log.error("连接失败")
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await pc.close()
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return 1
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await asyncio.sleep(0.1)
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log.info("✅ ICE 已连通（%s）", pc.connectionState)
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print(f" ✅ 连接建立，状态 {pc.connectionState}")
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if track:
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# 等音频放完 + 留出响应时间
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t0 = time.perf_counter()
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while not track.finished and time.perf_counter() - t0 < timeout_s:
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await asyncio.sleep(0.2)
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log.info("音频已发完，等待回复…")
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# 等首个回复音频
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t_wait = time.perf_counter()
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while time.perf_counter() - t_wait < timeout_s:
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if recorder.chunks:
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break
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await asyncio.sleep(0.1)
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# 再收一会儿，让回复说完
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await asyncio.sleep(6)
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dur = recorder.save()
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await pc.close()
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got_audio.set()
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print()
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print("=== 结果 ===")
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print(f" 连接: {'✅ 成功' if dur > 0 else '⚠️ 无音频返回'}")
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if dur:
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print(f" 收到音频: {dur:.2f} 秒 → {out}")
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fa = recorder.first_audio_ms
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if fa:
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print(f" 首音延迟（发完到收到首段）: {fa:.0f} ms")
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return 0 if dur > 0 else 1
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def main() -> int:
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p = argparse.ArgumentParser()
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p.add_argument("--server", default="http://cadadmin-server.taild0f91d.ts.net:7860",
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help="服务端地址。默认用 tailscale 域名——LAN 地址会被重新编址"
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"（2026-09-15 实测从 192.168.0.x 变成 172.16.1.x）")
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p.add_argument("--wav", type=Path)
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p.add_argument("--out", type=Path, default=Path("/tmp/client_reply.wav"))
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p.add_argument("--connect-only", action="store_true")
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p.add_argument("--timeout", type=float, default=30.0)
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a = p.parse_args()
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if not a.connect_only and not a.wav:
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print("需要 --wav 或 --connect-only", file=sys.stderr)
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return 2
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return asyncio.run(run(a.server, a.wav, a.out,
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connect_only=a.connect_only, timeout_s=a.timeout))
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if __name__ == "__main__":
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sys.exit(main())
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