update: 增加唤醒时声纹处理 ,1秒内发送至大模型 【需优化唤醒锁机制,中途会遭受打断(偶发),考虑忽略检测】

This commit is contained in:
Sakura-RanChen
2025-08-28 17:58:23 +08:00
parent 91cd843cfe
commit 41b8fac3aa
10 changed files with 126 additions and 519 deletions
+88 -60
View File
@@ -1,14 +1,14 @@
import os
import io
import wave
import uuid
import json
import time
import queue
import asyncio
import traceback
import threading
import opuslib_next
import json
import io
import time
import concurrent.futures
from abc import ABC, abstractmethod
from config.logger import setup_logging
@@ -53,6 +53,10 @@ class ASRProviderBase(ABC):
# 接收音频
async def receive_audio(self, conn, audio, audio_have_voice):
# 检查是否在唤醒处理锁定期内
if getattr(conn, 'wakeup_processing_lock', 0) > time.monotonic():
return
if conn.client_listen_mode == "auto" or conn.client_listen_mode == "realtime":
have_voice = audio_have_voice
else:
@@ -62,6 +66,14 @@ class ASRProviderBase(ABC):
if not have_voice and not conn.client_have_voice:
conn.asr_audio = conn.asr_audio[-10:]
return
# 检查是否处于唤醒模式
if getattr(conn, 'wakeup_mode', False) and len(conn.asr_audio) >= 10:
asr_audio_task = conn.asr_audio.copy()
conn.reset_vad_states()
conn.asr_audio.clear()
await self.handle_voice_stop(conn, asr_audio_task)
if conn.client_voice_stop:
asr_audio_task = conn.asr_audio.copy()
@@ -87,32 +99,13 @@ class ASRProviderBase(ABC):
# 预先准备WAV数据
wav_data = None
# 使用连接的声纹识别提供者
if conn.voiceprint_provider and combined_pcm_data:
wav_data = self._pcm_to_wav(combined_pcm_data)
# 检查是否处于唤醒模式
wakeup_mode = getattr(conn, 'wakeup_mode', False)
# 定义ASR任务
def run_asr():
start_time = time.monotonic()
try:
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
try:
result = loop.run_until_complete(
self.speech_to_text(asr_audio_task, conn.session_id, conn.audio_format)
)
end_time = time.monotonic()
logger.bind(tag=TAG).info(f"ASR耗时: {end_time - start_time:.3f}s")
return result
finally:
loop.close()
except Exception as e:
end_time = time.monotonic()
logger.bind(tag=TAG).error(f"ASR失败: {e}")
return ("", None)
# 定义声纹识别任务
# 声纹识别任务(公共逻辑)
def run_voiceprint():
if not wav_data:
return None
@@ -131,48 +124,83 @@ class ASRProviderBase(ABC):
logger.bind(tag=TAG).error(f"声纹识别失败: {e}")
return None
# 使用线程池执行器并行运行
with concurrent.futures.ThreadPoolExecutor(max_workers=2) as thread_executor:
asr_future = thread_executor.submit(run_asr)
if wakeup_mode and conn.voiceprint_provider and wav_data:
conn.wakeup_mode = False
# 设置处理锁,防止后续音频片段重复处理
conn.wakeup_processing_lock = time.monotonic() + 3 # 3秒锁定期
if conn.voiceprint_provider and wav_data:
# 唤醒模式:只执行声纹识别
with concurrent.futures.ThreadPoolExecutor(max_workers=1) as thread_executor:
voiceprint_future = thread_executor.submit(run_voiceprint)
# 等待两个线程都完成
asr_result = asr_future.result(timeout=15)
voiceprint_result = voiceprint_future.result(timeout=15)
results = {"asr": asr_result, "voiceprint": voiceprint_result}
else:
asr_result = asr_future.result(timeout=15)
results = {"asr": asr_result, "voiceprint": None}
# 处理结果
raw_text, file_path = results.get("asr", ("", None))
speaker_name = results.get("voiceprint", None)
# 记录识别结果
if raw_text:
logger.bind(tag=TAG).info(f"识别文本: {raw_text}")
if speaker_name:
speaker_name = voiceprint_result
logger.bind(tag=TAG).info(f"识别说话人: {speaker_name}")
# 性能监控
total_time = time.monotonic() - total_start_time
logger.bind(tag=TAG).info(f"总处理耗时: {total_time:.3f}s")
# 检查文本长度
text_len, _ = remove_punctuation_and_length(raw_text)
self.stop_ws_connection()
if text_len > 0:
# 构建包含说话人信息的JSON字符串
enhanced_text = self._build_enhanced_text(raw_text, speaker_name)
fixed_text = "嘿,你好啊"
enhanced_text = self._build_enhanced_text(fixed_text, speaker_name)
# 使用自定义模块进行上报
# 性能监控
total_time = time.monotonic() - total_start_time
logger.bind(tag=TAG).info(f"唤醒模式总处理耗时: {total_time:.3f}s")
await startToChat(conn, enhanced_text)
enqueue_asr_report(conn, enhanced_text, asr_audio_task)
else:
# 正常模式:执行声纹识别和ASR
def run_asr():
start_time = time.monotonic()
try:
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
try:
result = loop.run_until_complete(
self.speech_to_text(asr_audio_task, conn.session_id, conn.audio_format)
)
end_time = time.monotonic()
logger.bind(tag=TAG).info(f"ASR耗时: {end_time - start_time:.3f}s")
return result
finally:
loop.close()
except Exception as e:
end_time = time.monotonic()
logger.bind(tag=TAG).error(f"ASR失败: {e}")
return ("", None)
# 使用线程池执行器并行运行
with concurrent.futures.ThreadPoolExecutor(max_workers=2) as thread_executor:
asr_future = thread_executor.submit(run_asr)
if conn.voiceprint_provider and wav_data:
voiceprint_future = thread_executor.submit(run_voiceprint)
asr_result = asr_future.result(timeout=15)
voiceprint_result = voiceprint_future.result(timeout=15)
results = {"asr": asr_result, "voiceprint": voiceprint_result}
else:
asr_result = asr_future.result(timeout=15)
results = {"asr": asr_result, "voiceprint": None}
# 处理结果
raw_text, _ = results.get("asr", ("", None))
speaker_name = results.get("voiceprint", None)
if raw_text:
logger.bind(tag=TAG).info(f"识别文本: {raw_text}")
if speaker_name:
logger.bind(tag=TAG).info(f"识别说话人: {speaker_name}")
# 性能监控
total_time = time.monotonic() - total_start_time
logger.bind(tag=TAG).info(f"总处理耗时: {total_time:.3f}s")
# 检查文本长度
text_len, _ = remove_punctuation_and_length(raw_text)
self.stop_ws_connection()
if text_len > 0:
enhanced_text = self._build_enhanced_text(raw_text, speaker_name)
await startToChat(conn, enhanced_text)
enqueue_asr_report(conn, enhanced_text, asr_audio_task)
except Exception as e:
logger.bind(tag=TAG).error(f"处理语音停止失败: {e}")
@@ -1,9 +1,7 @@
import os
import time
import queue
import aiohttp
import asyncio
import requests
import traceback
from config.logger import setup_logging
from core.utils.tts import MarkdownCleaner
@@ -111,10 +109,6 @@ class TTSProvider(TTSProviderBase):
finally:
return None
###################################################################################
# linkerai单流式TTS重写父类的方法--结束
###################################################################################
async def text_to_speak(self, text, is_last):
"""流式处理TTS音频,每句只推送一次音频列表"""
await self._tts_request(text, is_last)
@@ -201,71 +195,3 @@ class TTSProvider(TTSProviderBase):
except Exception as e:
logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
self.tts_audio_queue.put((SentenceType.LAST, [], None))
def to_tts(self, text: str) -> list:
"""非流式TTS处理,用于测试及保存音频文件的场景
Args:
text: 要转换的文本
Returns:
list: 返回opus编码后的音频数据列表
"""
start_time = time.time()
text = MarkdownCleaner.clean_markdown(text)
params = {
"tts_text": text,
"spk_id": self.voice,
"frame_duration": 60,
"stream": False,
"target_sr": 16000,
"audio_format": self.audio_format,
"instruct_text": "请生成一段自然流畅的语音",
}
headers = {
"Authorization": f"Bearer {self.access_token}",
"Content-Type": "application/json",
}
try:
with requests.get(
self.api_url, params=params, headers=headers, timeout=5
) as response:
if response.status_code != 200:
logger.bind(tag=TAG).error(
f"TTS请求失败: {response.status_code}, {response.text}"
)
return []
logger.info(f"TTS请求成功: {text}, 耗时: {time.time() - start_time}")
# 使用opus编码器处理PCM数据
opus_datas = []
pcm_data = response.content
# 计算每帧的字节数
frame_bytes = int(
self.opus_encoder.sample_rate
* self.opus_encoder.channels
* self.opus_encoder.frame_size_ms
/ 1000
* 2
)
# 分帧处理PCM数据
for i in range(0, len(pcm_data), frame_bytes):
frame = pcm_data[i : i + frame_bytes]
if len(frame) < frame_bytes:
# 最后一帧可能不足,用0填充
frame = frame + b"\x00" * (frame_bytes - len(frame))
self.opus_encoder.encode_pcm_to_opus_stream(
frame,
end_of_stream=(i + frame_bytes >= len(pcm_data)),
callback=lambda opus: opus_datas.append(opus)
)
return opus_datas
except Exception as e:
logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
return []