Merge branch 'main' into py_test_tts

# Conflicts:
#	main/manager-api/src/main/resources/db/changelog/db.changelog-master.yaml
This commit is contained in:
hrz
2025-07-17 14:32:50 +08:00
61 changed files with 3783 additions and 354 deletions
+169 -37
View File
@@ -1,15 +1,18 @@
import os
import wave
import copy
import uuid
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
from typing import Optional, Tuple, List
from typing import Optional, Tuple, List, Dict, Any
from core.handle.receiveAudioHandle import startToChat
from core.handle.reportHandle import enqueue_asr_report
from core.utils.util import remove_punctuation_and_length
@@ -24,10 +27,7 @@ class ASRProviderBase(ABC):
pass
# 打开音频通道
# 这里默认是非流式的处理方式
# 流式处理方式请在子类中重写
async def open_audio_channels(self, conn):
# tts 消化线程
conn.asr_priority_thread = threading.Thread(
target=self.asr_text_priority_thread, args=(conn,), daemon=True
)
@@ -52,41 +52,171 @@ class ASRProviderBase(ABC):
continue
# 接收音频
# 这里默认是非流式的处理方式
# 流式处理方式请在子类中重写
async def receive_audio(self, conn, audio, audio_have_voice):
if conn.client_listen_mode == "auto" or conn.client_listen_mode == "realtime":
have_voice = audio_have_voice
else:
have_voice = conn.client_have_voice
# 如果本次没有声音,本段也没声音,就把声音丢弃了
conn.asr_audio.append(audio)
if have_voice == False and conn.client_have_voice == False:
if not have_voice and not conn.client_have_voice:
conn.asr_audio = conn.asr_audio[-10:]
return
# 如果本段有声音,且已经停止了
if conn.client_voice_stop:
asr_audio_task = copy.deepcopy(conn.asr_audio)
asr_audio_task = conn.asr_audio.copy()
conn.asr_audio.clear()
# 音频太短了,无法识别
conn.reset_vad_states()
if len(asr_audio_task) > 15:
await self.handle_voice_stop(conn, asr_audio_task)
# 处理语音停止
async def handle_voice_stop(self, conn, asr_audio_task):
raw_text, _ = await self.speech_to_text(
asr_audio_task, conn.session_id, conn.audio_format
) # 确保ASR模块返回原始文本
conn.logger.bind(tag=TAG).info(f"识别文本: {raw_text}")
text_len, _ = remove_punctuation_and_length(raw_text)
self.stop_ws_connection()
if text_len > 0:
# 使用自定义模块进行上报
await startToChat(conn, raw_text)
enqueue_asr_report(conn, raw_text, asr_audio_task)
async def handle_voice_stop(self, conn, asr_audio_task: List[bytes]):
"""并行处理ASR和声纹识别"""
try:
total_start_time = time.monotonic()
# 准备音频数据
if conn.audio_format == "pcm":
pcm_data = asr_audio_task
else:
pcm_data = self.decode_opus(asr_audio_task)
combined_pcm_data = b"".join(pcm_data)
# 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)
# 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
def run_voiceprint():
if not wav_data:
return None
try:
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
try:
# 使用连接的声纹识别提供者
result = loop.run_until_complete(
conn.voiceprint_provider.identify_speaker(wav_data, conn.session_id)
)
return result
finally:
loop.close()
except Exception as e:
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
parallel_start_time = time.monotonic()
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)
# 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:
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)
# 使用自定义模块进行上报
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}")
import traceback
logger.bind(tag=TAG).debug(f"异常详情: {traceback.format_exc()}")
def _build_enhanced_text(self, text: str, speaker_name: Optional[str]) -> str:
"""构建包含说话人信息的文本"""
if speaker_name and speaker_name.strip():
return json.dumps({
"speaker": speaker_name,
"content": text
}, ensure_ascii=False)
else:
return text
def _pcm_to_wav(self, pcm_data: bytes) -> bytes:
"""将PCM数据转换为WAV格式"""
if len(pcm_data) == 0:
logger.bind(tag=TAG).warning("PCM数据为空,无法转换WAV")
return b""
# 确保数据长度是偶数(16位音频)
if len(pcm_data) % 2 != 0:
pcm_data = pcm_data[:-1]
# 创建WAV文件头
wav_buffer = io.BytesIO()
try:
with wave.open(wav_buffer, 'wb') as wav_file:
wav_file.setnchannels(1) # 单声道
wav_file.setsampwidth(2) # 16位
wav_file.setframerate(16000) # 16kHz采样率
wav_file.writeframes(pcm_data)
wav_buffer.seek(0)
wav_data = wav_buffer.read()
return wav_data
except Exception as e:
logger.bind(tag=TAG).error(f"WAV转换失败: {e}")
return b""
def stop_ws_connection(self):
pass
@@ -113,27 +243,29 @@ class ASRProviderBase(ABC):
pass
@staticmethod
def decode_opus(opus_data: List[bytes]) -> bytes:
def decode_opus(opus_data: List[bytes]) -> List[bytes]:
"""将Opus音频数据解码为PCM数据"""
try:
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
decoder = opuslib_next.Decoder(16000, 1)
pcm_data = []
buffer_size = 960 # 每次处理960个采样点
for opus_packet in opus_data:
buffer_size = 960 # 每次处理960个采样点 (60ms at 16kHz)
for i, opus_packet in enumerate(opus_data):
try:
# 使用较小的缓冲区大小进行处理
if not opus_packet or len(opus_packet) == 0:
continue
pcm_frame = decoder.decode(opus_packet, buffer_size)
if pcm_frame:
if pcm_frame and len(pcm_frame) > 0:
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).warning(f"Opus解码错误,跳过当前数据包: {e}")
continue
logger.bind(tag=TAG).warning(f"Opus解码错误,跳过数据包 {i}: {e}")
except Exception as e:
logger.bind(tag=TAG).error(f"音频处理错误: {e}", exc_info=True)
continue
logger.bind(tag=TAG).error(f"音频处理错误,数据包 {i}: {e}")
return pcm_data
except Exception as e:
logger.bind(tag=TAG).error(f"音频解码过程发生错误: {e}", exc_info=True)
logger.bind(tag=TAG).error(f"音频解码过程发生错误: {e}")
return []
@@ -56,6 +56,16 @@ class ASRProvider(ASRProviderBase):
async def receive_audio(self, conn, audio, audio_have_voice):
conn.asr_audio.append(audio)
conn.asr_audio = conn.asr_audio[-10:]
# 存储音频数据
if not hasattr(conn, 'asr_audio_for_voiceprint'):
conn.asr_audio_for_voiceprint = []
conn.asr_audio_for_voiceprint.append(audio)
# 当没有音频数据时处理完整语音片段
if not audio and len(conn.asr_audio_for_voiceprint) > 0:
await self.handle_voice_stop(conn, conn.asr_audio_for_voiceprint)
conn.asr_audio_for_voiceprint = []
# 如果本次有声音,且之前没有建立连接
if audio_have_voice and self.asr_ws is None and not self.is_processing:
@@ -148,6 +158,8 @@ class ASRProvider(ASRProviderBase):
async def _forward_asr_results(self, conn):
try:
while self.asr_ws and not conn.stop_event.is_set():
# 获取当前连接的音频数据
audio_data = getattr(conn, 'asr_audio_for_voiceprint', [])
try:
response = await self.asr_ws.recv()
result = self.parse_response(response)
@@ -171,7 +183,8 @@ class ASRProvider(ASRProviderBase):
logger.bind(tag=TAG).error(f"识别文本:空")
self.text = ""
conn.reset_vad_states()
await self.handle_voice_stop(conn, None)
if len(audio_data) > 15: # 确保有足够音频数据
await self.handle_voice_stop(conn, audio_data)
break
for utterance in utterances:
@@ -181,7 +194,8 @@ class ASRProvider(ASRProviderBase):
f"识别到文本: {self.text}"
)
conn.reset_vad_states()
await self.handle_voice_stop(conn, None)
if len(audio_data) > 15: # 确保有足够音频数据
await self.handle_voice_stop(conn, audio_data)
break
elif "error" in payload:
error_msg = payload.get("error", "未知错误")
@@ -208,6 +222,13 @@ class ASRProvider(ASRProviderBase):
await self.asr_ws.close()
self.asr_ws = None
self.is_processing = False
if conn:
if hasattr(conn, 'asr_audio_for_voiceprint'):
conn.asr_audio_for_voiceprint = []
if hasattr(conn, 'asr_audio'):
conn.asr_audio = []
if hasattr(conn, 'has_valid_voice'):
conn.has_valid_voice = False
def stop_ws_connection(self):
if self.asr_ws:
@@ -349,3 +370,12 @@ class ASRProvider(ASRProviderBase):
pass
self.forward_task = None
self.is_processing = False
# 清理所有连接的音频缓冲区
if hasattr(self, '_connections'):
for conn in self._connections.values():
if hasattr(conn, 'asr_audio_for_voiceprint'):
conn.asr_audio_for_voiceprint = []
if hasattr(conn, 'asr_audio'):
conn.asr_audio = []
if hasattr(conn, 'has_valid_voice'):
conn.has_valid_voice = False
@@ -16,10 +16,11 @@ class IntentProvider(IntentProviderBase):
super().__init__(config)
self.llm = None
self.promot = ""
# 添加缓存管理
self.intent_cache = {} # 缓存意图识别结果
self.cache_expiry = 600 # 缓存有效期10分钟
self.cache_max_size = 100 # 最多缓存100个意图
# 导入全局缓存管理
from core.utils.cache.manager import cache_manager, CacheType
self.cache_manager = cache_manager
self.CacheType = CacheType
self.history_count = 4 # 默认使用最近4条对话记录
def get_intent_system_prompt(self, functions_list: str) -> str:
@@ -102,27 +103,6 @@ class IntentProvider(IntentProviderBase):
)
return prompt
def clean_cache(self):
"""清理过期缓存"""
now = time.time()
# 找出过期键
expired_keys = [
k
for k, v in self.intent_cache.items()
if now - v["timestamp"] > self.cache_expiry
]
for key in expired_keys:
del self.intent_cache[key]
# 如果缓存太大,移除最旧的条目
if len(self.intent_cache) > self.cache_max_size:
# 按时间戳排序并保留最新的条目
sorted_items = sorted(
self.intent_cache.items(), key=lambda x: x[1]["timestamp"]
)
for key, _ in sorted_items[: len(sorted_items) - self.cache_max_size]:
del self.intent_cache[key]
def replyResult(self, text: str, original_text: str):
llm_result = self.llm.response_no_stream(
system_prompt=text,
@@ -145,21 +125,16 @@ class IntentProvider(IntentProviderBase):
logger.bind(tag=TAG).debug(f"使用意图识别模型: {model_info}")
# 计算缓存键
cache_key = hashlib.md5(text.encode()).hexdigest()
cache_key = hashlib.md5((conn.device_id + text).encode()).hexdigest()
# 检查缓存
if cache_key in self.intent_cache:
cache_entry = self.intent_cache[cache_key]
# 检查缓存是否过期
if time.time() - cache_entry["timestamp"] <= self.cache_expiry:
cache_time = time.time() - total_start_time
logger.bind(tag=TAG).debug(
f"使用缓存的意图: {cache_key} -> {cache_entry['intent']}, 耗时: {cache_time:.4f}"
)
return cache_entry["intent"]
# 清理缓存
self.clean_cache()
cached_intent = self.cache_manager.get(self.CacheType.INTENT, cache_key)
if cached_intent is not None:
cache_time = time.time() - total_start_time
logger.bind(tag=TAG).debug(
f"使用缓存的意图: {cache_key} -> {cached_intent}, 耗时: {cache_time:.4f}"
)
return cached_intent
if self.promot == "":
functions = conn.func_handler.get_functions()
@@ -259,10 +234,7 @@ class IntentProvider(IntentProviderBase):
conn.dialogue.dialogue = clean_history
# 添加到缓存
self.intent_cache[cache_key] = {
"intent": intent,
"timestamp": time.time(),
}
self.cache_manager.set(self.CacheType.INTENT, cache_key, intent)
# 后处理时间
postprocess_time = time.time() - postprocess_start_time
@@ -272,10 +244,7 @@ class IntentProvider(IntentProviderBase):
return intent
else:
# 添加到缓存
self.intent_cache[cache_key] = {
"intent": intent,
"timestamp": time.time(),
}
self.cache_manager.set(self.CacheType.INTENT, cache_key, intent)
# 后处理时间
postprocess_time = time.time() - postprocess_start_time
@@ -79,10 +79,11 @@ short_term_memory_prompt_only_content = """
1、总结user的重要信息,以便在未来的对话中提供更个性化的服务
2、不要重复总结,不要遗忘之前记忆,除非原来的记忆超过了1800字内,否则不要遗忘、不要压缩用户的历史记忆
3、用户操控的设备音量、播放音乐、天气、退出、不想对话等和用户本身无关的内容,这些信息不需要加入到总结中
4、不要把设备操控的成果结果和失败结果加入到总结中,也不要把用户的一些废话加入到总结中
5、不要为了总结而总结,如果用户的聊天没有意义,请返回原来的历史记录也是可以的
6、只需要返回总结摘要,严格控制在1800字内
7、不要包含代码、xml,不需要解释、注释和说明,保存记忆时仅从对话提取信息,不要混入示例内容
4、聊天内容中的今天的日期时间、今天的天气情况与用户事件无关的数据,这些信息如果当成记忆存储会影响后序对话,这些信息不需要加入到总结中
5、不要把设备操控的成果结果和失败结果加入到总结中,也不要把用户的一些废话加入到总结中
6、不要为了总结而总结,如果用户的聊天没有意义,请返回原来的历史记录也是可以的
7、只需要返回总结摘要,严格控制在1800字内
8、不要包含代码、xml,不需要解释、注释和说明,保存记忆时仅从对话提取信息,不要混入示例内容
"""
@@ -51,13 +51,13 @@ class ServerPluginExecutor(ToolExecutor):
tools = {}
# 获取必要的函数
necessary_functions = ["handle_exit_intent", "get_time", "get_lunar"]
necessary_functions = ["handle_exit_intent", "get_lunar"]
# 获取配置中的函数
config_functions = self.config["Intent"][
self.config["selected_module"]["Intent"]
].get("functions", [])
# 转换为列表
if not isinstance(config_functions, list):
try: