mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-31 03:43:57 +08:00
resolve merge conflict
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@@ -9,7 +9,6 @@ import asyncio
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import traceback
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import threading
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import opuslib_next
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import concurrent.futures
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from abc import ABC, abstractmethod
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from config.logger import setup_logging
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from typing import Optional, Tuple, List
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@@ -118,121 +117,89 @@ class ASRProviderBase(ABC):
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# 接收音频
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async def receive_audio(self, conn, audio, audio_have_voice):
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if conn.client_listen_mode == "auto" or conn.client_listen_mode == "realtime":
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have_voice = audio_have_voice
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if conn.client_listen_mode == "manual":
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# 手动模式:缓存音频用于ASR识别
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conn.asr_audio.append(audio)
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else:
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have_voice = conn.client_have_voice
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conn.asr_audio.append(audio)
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if not have_voice and not conn.client_have_voice:
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conn.asr_audio = conn.asr_audio[-10:]
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return
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# 自动/实时模式:使用VAD检测
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have_voice = audio_have_voice
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if conn.client_voice_stop:
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asr_audio_task = conn.asr_audio.copy()
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conn.asr_audio.clear()
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conn.reset_vad_states()
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conn.asr_audio.append(audio)
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if not have_voice and not conn.client_have_voice:
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conn.asr_audio = conn.asr_audio[-10:]
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return
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if len(asr_audio_task) > 15:
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await self.handle_voice_stop(conn, asr_audio_task)
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# 自动模式下通过VAD检测到语音停止时触发识别
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if conn.client_voice_stop:
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asr_audio_task = conn.asr_audio.copy()
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conn.asr_audio.clear()
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conn.reset_vad_states()
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if len(asr_audio_task) > 15:
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await self.handle_voice_stop(conn, asr_audio_task)
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# 处理语音停止
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async def handle_voice_stop(self, conn, asr_audio_task: List[bytes]):
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"""并行处理ASR和声纹识别"""
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try:
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total_start_time = time.monotonic()
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# 准备音频数据
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if conn.audio_format == "pcm":
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pcm_data = asr_audio_task
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else:
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pcm_data = self.decode_opus(asr_audio_task)
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combined_pcm_data = b"".join(pcm_data)
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# 预先准备WAV数据
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wav_data = None
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if conn.voiceprint_provider and combined_pcm_data:
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wav_data = self._pcm_to_wav(combined_pcm_data)
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# 定义ASR任务
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def run_asr():
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start_time = time.monotonic()
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try:
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loop = asyncio.new_event_loop()
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asyncio.set_event_loop(loop)
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try:
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result = loop.run_until_complete(
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self.speech_to_text(asr_audio_task, conn.session_id, conn.audio_format)
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)
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end_time = time.monotonic()
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logger.bind(tag=TAG).info(f"ASR耗时: {end_time - start_time:.3f}s")
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return result
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finally:
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loop.close()
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except Exception as e:
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end_time = time.monotonic()
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logger.bind(tag=TAG).error(f"ASR失败: {e}")
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return ("", None)
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# 定义声纹识别任务
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def run_voiceprint():
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if not wav_data:
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return None
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try:
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loop = asyncio.new_event_loop()
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asyncio.set_event_loop(loop)
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try:
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# 使用连接的声纹识别提供者
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result = loop.run_until_complete(
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conn.voiceprint_provider.identify_speaker(wav_data, conn.session_id)
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)
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return result
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finally:
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loop.close()
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except Exception as e:
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logger.bind(tag=TAG).error(f"声纹识别失败: {e}")
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return None
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# 使用线程池执行器并行运行
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with concurrent.futures.ThreadPoolExecutor(max_workers=2) as thread_executor:
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asr_future = thread_executor.submit(run_asr)
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if conn.voiceprint_provider and wav_data:
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voiceprint_future = thread_executor.submit(run_voiceprint)
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# 等待两个线程都完成
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asr_result = asr_future.result(timeout=15)
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voiceprint_result = voiceprint_future.result(timeout=15)
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results = {"asr": asr_result, "voiceprint": voiceprint_result}
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else:
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asr_result = asr_future.result(timeout=15)
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results = {"asr": asr_result, "voiceprint": None}
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# 处理结果
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raw_text, _ = results.get("asr", ("", None))
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speaker_name = results.get("voiceprint", None)
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# 记录识别结果
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asr_task = self.speech_to_text(asr_audio_task, conn.session_id, conn.audio_format)
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if conn.voiceprint_provider and wav_data:
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voiceprint_task = conn.voiceprint_provider.identify_speaker(wav_data, conn.session_id)
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# 并发等待两个结果
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asr_result, voiceprint_result = await asyncio.gather(
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asr_task, voiceprint_task, return_exceptions=True
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)
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else:
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asr_result = await asr_task
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voiceprint_result = None
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# 记录识别结果 - 检查是否为异常
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if isinstance(asr_result, Exception):
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logger.bind(tag=TAG).error(f"ASR识别失败: {asr_result}")
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raw_text = ""
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else:
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raw_text, _ = asr_result
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if isinstance(voiceprint_result, Exception):
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logger.bind(tag=TAG).error(f"声纹识别失败: {voiceprint_result}")
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speaker_name = ""
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else:
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speaker_name = voiceprint_result
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if raw_text:
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logger.bind(tag=TAG).info(f"识别文本: {raw_text}")
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if speaker_name:
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logger.bind(tag=TAG).info(f"识别说话人: {speaker_name}")
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# 性能监控
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total_time = time.monotonic() - total_start_time
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logger.bind(tag=TAG).info(f"总处理耗时: {total_time:.3f}s")
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logger.bind(tag=TAG).debug(f"总处理耗时: {total_time:.3f}s")
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# 检查文本长度
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text_len, _ = remove_punctuation_and_length(raw_text)
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self.stop_ws_connection()
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if text_len > 0:
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# 构建包含说话人信息的JSON字符串
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enhanced_text = self._build_enhanced_text(raw_text, speaker_name)
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# 使用自定义模块进行上报
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await startToChat(conn, enhanced_text)
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enqueue_asr_report(conn, enhanced_text, asr_audio_task)
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@@ -306,6 +273,7 @@ class ASRProviderBase(ABC):
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@staticmethod
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def decode_opus(opus_data: List[bytes]) -> List[bytes]:
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"""将Opus音频数据解码为PCM数据"""
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decoder = None
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try:
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decoder = opuslib_next.Decoder(16000, 1)
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pcm_data = []
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@@ -330,3 +298,9 @@ class ASRProviderBase(ABC):
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except Exception as e:
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logger.bind(tag=TAG).error(f"音频解码过程发生错误: {e}")
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return []
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finally:
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if decoder is not None:
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try:
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del decoder
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except Exception as e:
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logger.bind(tag=TAG).debug(f"释放decoder资源时出错: {e}")
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