Merge branch 'main' into py_tts_huoshan

This commit is contained in:
CGD
2025-12-26 14:46:51 +08:00
committed by GitHub
134 changed files with 5839 additions and 1176 deletions
+9 -1
View File
@@ -10,7 +10,15 @@ class BaseHandler:
def _add_cors_headers(self, response):
"""添加CORS头信息"""
response.headers["Access-Control-Allow-Headers"] = (
"client-id, content-type, device-id"
"client-id, content-type, device-id, authorization"
)
response.headers["Access-Control-Allow-Credentials"] = "true"
response.headers["Access-Control-Allow-Origin"] = "*"
async def handle_options(self, request):
"""处理OPTIONS请求,添加CORS头信息"""
response = web.Response(body=b"", content_type="text/plain")
self._add_cors_headers(response)
# 添加允许的方法
response.headers["Access-Control-Allow-Methods"] = "GET, POST, OPTIONS"
return response
+231 -15
View File
@@ -3,15 +3,46 @@ import time
import base64
import hashlib
import hmac
import os
import re
import glob
from typing import Dict, List, Tuple
from aiohttp import web
from core.auth import AuthManager
from core.utils.util import get_local_ip
from core.utils.util import get_local_ip, get_vision_url
from core.api.base_handler import BaseHandler
TAG = __name__
def _safe_basename(filename: str) -> str:
# Prevent directory traversal
return os.path.basename(filename)
def _parse_version(ver: str) -> Tuple[int, ...]:
# conservative parser: split by non-digit, keep numeric parts
parts = re.findall(r"\d+", ver)
return tuple(int(p) for p in parts) if parts else (0,)
def _is_higher_version(a: str, b: str) -> bool:
"""Return True if version string a > b (semver-like numeric compare)."""
ta = _parse_version(a)
tb = _parse_version(b)
# compare tuple lexicographically, but allow different lengths
maxlen = max(len(ta), len(tb))
for i in range(maxlen):
ai = ta[i] if i < len(ta) else 0
bi = tb[i] if i < len(tb) else 0
if ai > bi:
return True
if ai < bi:
return False
return False
class OTAHandler(BaseHandler):
def __init__(self, config: dict):
super().__init__(config)
@@ -23,6 +54,54 @@ class OTAHandler(BaseHandler):
expire_seconds = auth_config.get("expire_seconds")
self.auth = AuthManager(secret_key=secret_key, expire_seconds=expire_seconds)
# firmware storage
self.bin_dir = os.path.join(os.getcwd(), "data", "bin")
# cache structure: { 'updated_at': timestamp, 'ttl': seconds, 'files_by_model': { model: [(version, filename), ...] } }
self._bin_cache: Dict = {
"updated_at": 0,
"ttl": config.get("firmware_cache_ttl", 30),
"files_by_model": {},
}
def _refresh_bin_cache_if_needed(self):
now = int(time.time())
ttl = int(self._bin_cache.get("ttl", 30))
if now - int(
self._bin_cache.get("updated_at", 0)
) < ttl and self._bin_cache.get("files_by_model"):
return
files_by_model: Dict[str, List[Tuple[str, str]]] = {}
try:
if not os.path.isdir(self.bin_dir):
os.makedirs(self.bin_dir, exist_ok=True)
# match files like model_1.2.3.bin (allow dots, dashes, underscores in model and version)
pattern = os.path.join(self.bin_dir, "*.bin")
for path in glob.glob(pattern):
fname = os.path.basename(path)
# filename format: {model}_{version}.bin
m = re.match(r"^(.+?)_([0-9][A-Za-z0-9\.\-_]*)\.bin$", fname)
if not m:
# skip files not conforming to naming rule
continue
model = m.group(1)
version = m.group(2)
files_by_model.setdefault(model, []).append((version, fname))
# sort versions for each model descending
for model, items in files_by_model.items():
items.sort(key=lambda it: _parse_version(it[0]), reverse=True)
self._bin_cache["files_by_model"] = files_by_model
self._bin_cache["updated_at"] = now
self.logger.bind(tag=TAG).info(
f"Firmware cache refreshed: {len(files_by_model)} models"
)
except Exception as e:
self.logger.bind(tag=TAG).error(f"刷新固件缓存失败: {e}")
# keep previous cache if any
def generate_password_signature(self, content: str, secret_key: str) -> str:
"""生成MQTT密码签名
@@ -62,7 +141,14 @@ class OTAHandler(BaseHandler):
return f"ws://{local_ip}:{port}/xiaozhi/v1/"
async def handle_post(self, request):
"""处理 OTA POST 请求"""
"""处理 OTA POST 请求
This handler will:
- read device id/client id (as before)
- attempt to determine device model and current firmware version (prefer headers, fallback to body)
- check data/bin for newer firmware for that model
- if found a newer firmware, set firmware.url to the download endpoint
"""
try:
data = await request.text()
self.logger.bind(tag=TAG).debug(f"OTA请求方法: {request.method}")
@@ -81,33 +167,76 @@ class OTAHandler(BaseHandler):
else:
raise Exception("OTA请求ClientID为空")
data_json = json.loads(data)
data_json = {}
try:
data_json = json.loads(data) if data else {}
except Exception:
data_json = {}
server_config = self.config["server"]
port = int(server_config.get("port", 8000))
# Distinguish ports:
# - websocket_port is used to construct websocket URL (server["port"])
# - http_port is used to construct OTA download URLs (server["http_port"])
websocket_port = int(server_config.get("port", 8000))
http_port = int(server_config.get("http_port", 8003))
local_ip = get_local_ip()
# Determine device model (prefer headers)
device_model = ""
# header candidates
for h in ("device-model", "device_model", "model"):
if h in request.headers:
device_model = request.headers.get(h, "").strip()
break
# body fallback
if not device_model:
try:
if "board" in data_json and isinstance(data_json["board"], dict):
device_model = data_json["board"].get("type", "")
elif "model" in data_json:
device_model = data_json.get("model", "")
except Exception:
device_model = ""
if not device_model:
device_model = "default"
# Determine device current version (prefer headers)
device_version = ""
for h in (
"device-version",
"device_version",
"firmware-version",
"app-version",
"application-version",
):
if h in request.headers:
device_version = request.headers.get(h, "").strip()
break
if not device_version:
try:
device_version = data_json.get("application", {}).get("version", "")
except Exception:
device_version = ""
if not device_version:
device_version = "0.0.0"
return_json = {
"server_time": {
"timestamp": int(round(time.time() * 1000)),
"timezone_offset": server_config.get("timezone_offset", 8) * 60,
},
"firmware": {
"version": data_json["application"].get("version", "1.0.0"),
"version": device_version,
"url": "",
},
}
# existing mqtt/websocket logic (unchanged)
mqtt_gateway_endpoint = server_config.get("mqtt_gateway")
if mqtt_gateway_endpoint: # 如果配置了非空字符串
# 尝试从请求数据中获取设备型号
device_model = "default"
# 尝试从请求数据中获取设备型号(已解析 above
try:
if "device" in data_json and isinstance(data_json["device"], dict):
device_model = data_json["device"].get("model", "default")
elif "model" in data_json:
device_model = data_json["model"]
group_id = f"GID_{device_model}".replace(":", "_").replace(" ", "_")
except Exception as e:
self.logger.bind(tag=TAG).error(f"获取设备型号失败: {e}")
@@ -159,20 +288,61 @@ class OTAHandler(BaseHandler):
token = self.auth.generate_token(client_id, device_id)
else:
token = self.auth.generate_token(client_id, device_id)
# NOTE: use websocket_port here
return_json["websocket"] = {
"url": self._get_websocket_url(local_ip, port),
"url": self._get_websocket_url(local_ip, websocket_port),
"token": token,
}
self.logger.bind(tag=TAG).info(
f"未配置MQTT网关,为设备 {device_id} 下发WebSocket配置"
)
self.logger.bind(tag=TAG).info(f"{return_json}")
# Now check firmware files for updates
try:
self._refresh_bin_cache_if_needed()
files_by_model = self._bin_cache.get("files_by_model", {})
candidates = files_by_model.get(device_model, [])
self.logger.bind(tag=TAG).info(
f"查找型号 {device_model} 的固件,找到 {len(candidates)} 个候选"
)
chosen_url = ""
chosen_version = device_version
# candidates are sorted descending by version
for ver, fname in candidates:
if _is_higher_version(ver, device_version):
# build download url (only allow download via our download endpoint)
chosen_version = ver
# Use get_vision_url to get the base URL and replace the path
vision_url = get_vision_url(self.config)
# Replace the path from "/mcp/vision/explain" to "/xiaozhi/ota/download/{fname}"
chosen_url = vision_url.replace(
"/mcp/vision/explain", f"/xiaozhi/ota/download/{fname}"
)
break
if chosen_url:
return_json["firmware"]["version"] = chosen_version
return_json["firmware"]["url"] = chosen_url
self.logger.bind(tag=TAG).info(
f"为设备 {device_id} 下发固件 {chosen_version} [如果地址前缀有误,请检查配置文件中的server.vision_explain]-> {chosen_url} "
)
else:
self.logger.bind(tag=TAG).info(
f"设备 {device_id} 固件已是最新: {device_version}"
)
except Exception as e:
self.logger.bind(tag=TAG).error(f"检查固件版本时出错: {e}")
response = web.Response(
text=json.dumps(return_json, separators=(",", ":")),
content_type="application/json",
)
except Exception as e:
self.logger.bind(tag=TAG).error(f"OTA POST处理异常: {e}")
return_json = {"success": False, "message": "request error."}
response = web.Response(
text=json.dumps(return_json, separators=(",", ":")),
@@ -187,8 +357,9 @@ class OTAHandler(BaseHandler):
try:
server_config = self.config["server"]
local_ip = get_local_ip()
port = int(server_config.get("port", 8000))
websocket_url = self._get_websocket_url(local_ip, port)
# use websocket port for websocket URL
websocket_port = int(server_config.get("port", 8000))
websocket_url = self._get_websocket_url(local_ip, websocket_port)
message = f"OTA接口运行正常,向设备发送的websocket地址是:{websocket_url}"
response = web.Response(text=message, content_type="text/plain")
except Exception as e:
@@ -197,3 +368,48 @@ class OTAHandler(BaseHandler):
finally:
self._add_cors_headers(response)
return response
async def handle_download(self, request):
"""
下载固件接口
URL: /xiaozhi/ota/download/{filename}
- 只允许下载 data/bin 目录下的 .bin 文件
- filename 必须是 basename 且匹配安全的模式
"""
try:
fname = request.match_info.get("filename", "")
if not fname:
raise web.HTTPBadRequest(text="filename required")
# sanitize
fname = _safe_basename(fname)
# pattern: allow letters, numbers, dot, underscore, dash
if not re.match(r"^[A-Za-z0-9\.\-_]+\.bin$", fname):
raise web.HTTPBadRequest(text="invalid filename")
file_path = os.path.join(self.bin_dir, fname)
# ensure realpath is under bin_dir
file_real = os.path.realpath(file_path)
bin_dir_real = os.path.realpath(self.bin_dir)
if (
not file_real.startswith(bin_dir_real + os.sep)
and file_real != bin_dir_real
):
raise web.HTTPForbidden(text="forbidden")
if not os.path.isfile(file_real):
raise web.HTTPNotFound(text="file not found")
# use FileResponse to stream file
resp = web.FileResponse(path=file_real)
except web.HTTPError as e:
resp = e
except Exception as e:
self.logger.bind(tag=TAG).error(f"固件下载异常: {e}")
resp = web.Response(text="download error", status=500)
finally:
try:
self._add_cors_headers(resp)
except Exception:
pass
return resp
+4 -12
View File
@@ -2,6 +2,7 @@ import json
import copy
from aiohttp import web
from config.logger import setup_logging
from core.api.base_handler import BaseHandler
from core.utils.util import get_vision_url, is_valid_image_file
from core.utils.vllm import create_instance
from config.config_loader import get_private_config_from_api
@@ -16,10 +17,9 @@ TAG = __name__
MAX_FILE_SIZE = 5 * 1024 * 1024
class VisionHandler:
class VisionHandler(BaseHandler):
def __init__(self, config: dict):
self.config = config
self.logger = setup_logging()
super().__init__(config)
# 初始化认证工具
self.auth = AuthToken(config["server"]["auth_key"])
@@ -96,7 +96,7 @@ class VisionHandler:
current_config = copy.deepcopy(self.config)
read_config_from_api = current_config.get("read_config_from_api", False)
if read_config_from_api:
current_config = get_private_config_from_api(
current_config = await get_private_config_from_api(
current_config,
device_id,
client_id,
@@ -172,11 +172,3 @@ class VisionHandler:
finally:
self._add_cors_headers(response)
return response
def _add_cors_headers(self, response):
"""添加CORS头信息"""
response.headers["Access-Control-Allow-Headers"] = (
"client-id, content-type, device-id"
)
response.headers["Access-Control-Allow-Credentials"] = "true"
response.headers["Access-Control-Allow-Origin"] = "*"
+141 -49
View File
@@ -68,8 +68,12 @@ class ConnectionHandler:
self.logger = setup_logging()
self.server = server # 保存server实例的引用
self.need_bind = False
self.bind_code = None
self.need_bind = False # 是否需要绑定设备
self.bind_completed_event = asyncio.Event()
self.bind_code = None # 绑定设备的验证码
self.last_bind_prompt_time = 0 # 上次播放绑定提示的时间戳(秒)
self.bind_prompt_interval = 60 # 绑定提示播放间隔(秒)
self.read_config_from_api = self.config.get("read_config_from_api", False)
self.websocket = None
@@ -88,7 +92,7 @@ class ConnectionHandler:
self.client_listen_mode = "auto"
# 线程任务相关
self.loop = asyncio.get_event_loop()
self.loop = None # 在 handle_connection 中获取运行中的事件循环
self.stop_event = threading.Event()
self.executor = ThreadPoolExecutor(max_workers=5)
@@ -116,6 +120,7 @@ class ConnectionHandler:
self.client_audio_buffer = bytearray()
self.client_have_voice = False
self.client_voice_window = deque(maxlen=5)
self.first_activity_time = 0.0 # 记录首次活动的时间(毫秒)
self.last_activity_time = 0.0 # 统一的活动时间戳(毫秒)
self.client_voice_stop = False
self.last_is_voice = False
@@ -158,10 +163,13 @@ class ConnectionHandler:
self.conn_from_mqtt_gateway = False
# 初始化提示词管理器
self.prompt_manager = PromptManager(config, self.logger)
self.prompt_manager = PromptManager(self.config, self.logger)
async def handle_connection(self, ws):
try:
# 获取运行中的事件循环(必须在异步上下文中)
self.loop = asyncio.get_running_loop()
# 获取并验证headers
self.headers = dict(ws.request.headers)
real_ip = self.headers.get("x-real-ip") or self.headers.get(
@@ -187,6 +195,7 @@ class ConnectionHandler:
self.logger.bind(tag=TAG).info("连接来自:MQTT网关")
# 初始化活动时间戳
self.first_activity_time = time.time() * 1000
self.last_activity_time = time.time() * 1000
# 启动超时检查任务
@@ -195,10 +204,8 @@ class ConnectionHandler:
self.welcome_msg = self.config["xiaozhi"]
self.welcome_msg["session_id"] = self.session_id
# 获取差异化配置
self._initialize_private_config()
# 异步初始化
self.executor.submit(self._initialize_components)
# 在后台初始化配置和组件(完全不阻塞主循环)
asyncio.create_task(self._background_initialize())
try:
async for message in self.websocket:
@@ -237,7 +244,9 @@ class ConnectionHandler:
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
loop.run_until_complete(
self.memory.save_memory(self.dialogue.dialogue)
self.memory.save_memory(
self.dialogue.dialogue, self.session_id
)
)
except Exception as e:
self.logger.bind(tag=TAG).error(f"保存记忆失败: {e}")
@@ -260,8 +269,37 @@ class ConnectionHandler:
f"保存记忆后关闭连接失败: {close_error}"
)
async def _discard_message_with_bind_prompt(self):
"""丢弃消息并检查是否需要播放绑定提示"""
current_time = time.time()
# 检查是否需要播放绑定提示
if current_time - self.last_bind_prompt_time >= self.bind_prompt_interval:
self.last_bind_prompt_time = current_time
# 复用现有的绑定提示逻辑
from core.handle.receiveAudioHandle import check_bind_device
asyncio.create_task(check_bind_device(self))
async def _route_message(self, message):
"""消息路由"""
# 检查是否已经获取到真实的绑定状态
if not self.bind_completed_event.is_set():
# 还没有获取到真实状态,等待直到获取到真实状态或超时
try:
await asyncio.wait_for(self.bind_completed_event.wait(), timeout=1)
except asyncio.TimeoutError:
# 超时仍未获取到真实状态,丢弃消息
await self._discard_message_with_bind_prompt()
return
# 已经获取到真实状态,检查是否需要绑定
if self.need_bind:
# 需要绑定,丢弃消息
await self._discard_message_with_bind_prompt()
return
# 不需要绑定,继续处理消息
if isinstance(message, str):
await handleTextMessage(self, message)
elif isinstance(message, bytes):
@@ -391,6 +429,15 @@ class ConnectionHandler:
def _initialize_components(self):
try:
if self.tts is None:
self.tts = self._initialize_tts()
# 打开语音合成通道
asyncio.run_coroutine_threadsafe(
self.tts.open_audio_channels(self), self.loop
)
if self.need_bind:
self.bind_completed_event.set()
return
self.selected_module_str = build_module_string(
self.config.get("selected_module", {})
)
@@ -414,17 +461,10 @@ class ConnectionHandler:
# 初始化声纹识别
self._initialize_voiceprint()
# 打开语音识别通道
asyncio.run_coroutine_threadsafe(
self.asr.open_audio_channels(self), self.loop
)
if self.tts is None:
self.tts = self._initialize_tts()
# 打开语音合成通道
asyncio.run_coroutine_threadsafe(
self.tts.open_audio_channels(self), self.loop
)
"""加载记忆"""
self._initialize_memory()
@@ -439,6 +479,7 @@ class ConnectionHandler:
self.logger.bind(tag=TAG).error(f"实例化组件失败: {e}")
def _init_prompt_enhancement(self):
# 更新上下文信息
self.prompt_manager.update_context_info(self, self.client_ip)
enhanced_prompt = self.prompt_manager.build_enhanced_prompt(
@@ -474,7 +515,11 @@ class ConnectionHandler:
def _initialize_asr(self):
"""初始化ASR"""
if self._asr.interface_type == InterfaceType.LOCAL:
if (
self._asr is not None
and hasattr(self._asr, "interface_type")
and self._asr.interface_type == InterfaceType.LOCAL
):
# 如果公共ASR是本地服务,则直接返回
# 因为本地一个实例ASR,可以被多个连接共享
asr = self._asr
@@ -501,22 +546,35 @@ class ConnectionHandler:
except Exception as e:
self.logger.bind(tag=TAG).warning(f"声纹识别初始化失败: {str(e)}")
def _initialize_private_config(self):
"""如果是从配置文件获取,则进行二次实例化"""
async def _background_initialize(self):
"""在后台初始化配置和组件(完全不阻塞主循环)"""
try:
# 异步获取差异化配置
await self._initialize_private_config_async()
# 在线程池中初始化组件
self.executor.submit(self._initialize_components)
except Exception as e:
self.logger.bind(tag=TAG).error(f"后台初始化失败: {e}")
async def _initialize_private_config_async(self):
"""从接口异步获取差异化配置(异步版本,不阻塞主循环)"""
if not self.read_config_from_api:
self.need_bind = False
self.bind_completed_event.set()
return
"""从接口获取差异化的配置进行二次实例化,非全量重新实例化"""
try:
begin_time = time.time()
private_config = get_private_config_from_api(
private_config = await get_private_config_from_api(
self.config,
self.headers.get("device-id"),
self.headers.get("client-id", self.headers.get("device-id")),
)
private_config["delete_audio"] = bool(self.config.get("delete_audio", True))
self.logger.bind(tag=TAG).info(
f"{time.time() - begin_time} 秒,获取差异化配置成功: {json.dumps(filter_sensitive_info(private_config), ensure_ascii=False)}"
f"{time.time() - begin_time} 秒,异步获取差异化配置成功: {json.dumps(filter_sensitive_info(private_config), ensure_ascii=False)}"
)
self.need_bind = False
self.bind_completed_event.set()
except DeviceNotFoundException as e:
self.need_bind = True
private_config = {}
@@ -526,7 +584,7 @@ class ConnectionHandler:
private_config = {}
except Exception as e:
self.need_bind = True
self.logger.bind(tag=TAG).error(f"获取差异化配置失败: {e}")
self.logger.bind(tag=TAG).error(f"异步获取差异化配置失败: {e}")
private_config = {}
init_llm, init_tts, init_memory, init_intent = (
@@ -599,8 +657,14 @@ class ConnectionHandler:
self.chat_history_conf = int(private_config["chat_history_conf"])
if private_config.get("mcp_endpoint", None) is not None:
self.config["mcp_endpoint"] = private_config["mcp_endpoint"]
if private_config.get("context_providers", None) is not None:
self.config["context_providers"] = private_config["context_providers"]
# 使用 run_in_executor 在线程池中执行 initialize_modules,避免阻塞主循环
try:
modules = initialize_modules(
modules = await self.loop.run_in_executor(
None, # 使用默认线程池
initialize_modules,
self.logger,
private_config,
init_vad,
@@ -744,18 +808,26 @@ class ConnectionHandler:
force_final_answer = False # 标记是否强制最终回答
if depth >= MAX_DEPTH:
self.logger.bind(tag=TAG).debug(f"已达到最大工具调用深度 {MAX_DEPTH},将强制基于现有信息回答")
self.logger.bind(tag=TAG).debug(
f"已达到最大工具调用深度 {MAX_DEPTH},将强制基于现有信息回答"
)
force_final_answer = True
# 添加系统指令,要求 LLM 基于现有信息回答
self.dialogue.put(Message(
role="user",
content="[系统提示] 已达到最大工具调用次数限制,请你基于目前已经获取的所有信息,直接给出最终答案。不要再尝试调用任何工具。"
))
self.dialogue.put(
Message(
role="user",
content="[系统提示] 已达到最大工具调用次数限制,请你基于目前已经获取的所有信息,直接给出最终答案。不要再尝试调用任何工具。",
)
)
# Define intent functions
functions = None
# 达到最大深度时,禁用工具调用,强制 LLM 直接回答
if self.intent_type == "function_call" and hasattr(self, "func_handler") and not force_final_answer:
if (
self.intent_type == "function_call"
and hasattr(self, "func_handler")
and not force_final_answer
):
functions = self.func_handler.get_functions()
response_message = []
@@ -844,11 +916,16 @@ class ConnectionHandler:
if a is not None:
try:
content_arguments_json = json.loads(a)
tool_calls_list.append({
"id": str(uuid.uuid4().hex),
"name": content_arguments_json["name"],
"arguments": json.dumps(content_arguments_json["arguments"], ensure_ascii=False)
})
tool_calls_list.append(
{
"id": str(uuid.uuid4().hex),
"name": content_arguments_json["name"],
"arguments": json.dumps(
content_arguments_json["arguments"],
ensure_ascii=False,
),
}
)
except Exception as e:
bHasError = True
response_message.append(a)
@@ -880,7 +957,9 @@ class ConnectionHandler:
)
future = asyncio.run_coroutine_threadsafe(
self.func_handler.handle_llm_function_call(self, tool_call_data),
self.func_handler.handle_llm_function_call(
self, tool_call_data
),
self.loop,
)
futures_with_data.append((future, tool_call_data))
@@ -888,7 +967,7 @@ class ConnectionHandler:
# 等待协程结束(实际等待时长为最慢的那个)
tool_results = []
for future, tool_call_data in futures_with_data:
result = future.result()
result = future.result()
tool_results.append((result, tool_call_data))
# 统一处理所有工具调用结果
@@ -922,7 +1001,11 @@ class ConnectionHandler:
need_llm_tools = []
for result, tool_call_data in tool_results:
if result.action in [Action.RESPONSE, Action.NOTFOUND, Action.ERROR]: # 直接回复前端
if result.action in [
Action.RESPONSE,
Action.NOTFOUND,
Action.ERROR,
]: # 直接回复前端
text = result.response if result.response else result.result
self.tts.tts_one_sentence(self, ContentType.TEXT, content_detail=text)
self.dialogue.put(Message(role="assistant", content=text))
@@ -957,7 +1040,11 @@ class ConnectionHandler:
self.dialogue.put(
Message(
role="tool",
tool_call_id=str(uuid.uuid4()) if tool_call_data["id"] is None else tool_call_data["id"],
tool_call_id=(
str(uuid.uuid4())
if tool_call_data["id"] is None
else tool_call_data["id"]
),
content=text,
)
)
@@ -990,8 +1077,8 @@ class ConnectionHandler:
def _process_report(self, type, text, audio_data, report_time):
"""处理上报任务"""
try:
# 执行上报(传入二进制数据
report(self, type, text, audio_data, report_time)
# 执行异步上报(在事件循环中运行
asyncio.run(report(self, type, text, audio_data, report_time))
except Exception as e:
self.logger.bind(tag=TAG).error(f"上报处理异常: {e}")
finally:
@@ -1082,7 +1169,6 @@ class ConnectionHandler:
f"关闭线程池时出错: {executor_error}"
)
self.executor = None
self.logger.bind(tag=TAG).info("连接资源已释放")
except Exception as e:
self.logger.bind(tag=TAG).error(f"关闭连接时出错: {e}")
@@ -1112,6 +1198,11 @@ class ConnectionHandler:
except queue.Empty:
break
# 重置音频流控器(取消后台任务并清空队列)
if hasattr(self, "audio_rate_controller") and self.audio_rate_controller:
self.audio_rate_controller.reset()
self.logger.bind(tag=TAG).debug("已重置音频流控器")
self.logger.bind(tag=TAG).debug(
f"清理结束: TTS队列大小={self.tts.tts_text_queue.qsize()}, 音频队列大小={self.tts.tts_audio_queue.qsize()}"
)
@@ -1137,13 +1228,14 @@ class ConnectionHandler:
"""检查连接超时"""
try:
while not self.stop_event.is_set():
last_activity_time = self.last_activity_time
if self.need_bind:
last_activity_time = self.first_activity_time
# 检查是否超时(只有在时间戳已初始化的情况下)
if self.last_activity_time > 0.0:
if last_activity_time > 0.0:
current_time = time.time() * 1000
if (
current_time - self.last_activity_time
> self.timeout_seconds * 1000
):
if current_time - last_activity_time > self.timeout_seconds * 1000:
if not self.stop_event.is_set():
self.logger.bind(tag=TAG).info("连接超时,准备关闭")
# 设置停止事件,防止重复处理
@@ -1171,7 +1263,7 @@ class ConnectionHandler:
tools_call: 新的工具调用
"""
for tool_call in tools_call:
tool_index = getattr(tool_call, 'index', None)
tool_index = getattr(tool_call, "index", None)
if tool_index is None:
if tool_call.function.name:
# 有 function_name,说明是新的工具调用
@@ -1189,4 +1281,4 @@ class ConnectionHandler:
if tool_call.function.name:
tool_calls_list[tool_index]["name"] = tool_call.function.name
if tool_call.function.arguments:
tool_calls_list[tool_index]["arguments"] += tool_call.function.arguments
tool_calls_list[tool_index]["arguments"] += tool_call.function.arguments
@@ -101,7 +101,7 @@ async def checkWakeupWords(conn, text):
}
# 获取音频数据
opus_packets = audio_to_data(response.get("file_path"))
opus_packets = await audio_to_data(response.get("file_path"), use_cache=False)
# 播放唤醒词回复
conn.client_abort = False
@@ -123,7 +123,7 @@ async def max_out_size(conn):
text = "不好意思,我现在有点事情要忙,明天这个时候我们再聊,约好了哦!明天不见不散,拜拜!"
await send_stt_message(conn, text)
file_path = "config/assets/max_output_size.wav"
opus_packets = audio_to_data(file_path)
opus_packets = await audio_to_data(file_path)
conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
conn.close_after_chat = True
@@ -142,7 +142,7 @@ async def check_bind_device(conn):
# 播放提示音
music_path = "config/assets/bind_code.wav"
opus_packets = audio_to_data(music_path)
opus_packets = await audio_to_data(music_path)
conn.tts.tts_audio_queue.put((SentenceType.FIRST, opus_packets, text))
# 逐个播放数字
@@ -150,7 +150,7 @@ async def check_bind_device(conn):
try:
digit = conn.bind_code[i]
num_path = f"config/assets/bind_code/{digit}.wav"
num_packets = audio_to_data(num_path)
num_packets = await audio_to_data(num_path)
conn.tts.tts_audio_queue.put((SentenceType.MIDDLE, num_packets, None))
except Exception as e:
conn.logger.bind(tag=TAG).error(f"播放数字音频失败: {e}")
@@ -162,5 +162,5 @@ async def check_bind_device(conn):
text = f"没有找到该设备的版本信息,请正确配置 OTA地址,然后重新编译固件。"
await send_stt_message(conn, text)
music_path = "config/assets/bind_not_found.wav"
opus_packets = audio_to_data(music_path)
opus_packets = await audio_to_data(music_path)
conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
+41 -34
View File
@@ -10,7 +10,6 @@ TTS上报功能已集成到ConnectionHandler类中。
"""
import time
import opuslib_next
from config.manage_api_client import report as manage_report
@@ -18,7 +17,7 @@ from config.manage_api_client import report as manage_report
TAG = __name__
def report(conn, type, text, opus_data, report_time):
async def report(conn, type, text, opus_data, report_time):
"""执行聊天记录上报操作
Args:
@@ -33,8 +32,8 @@ def report(conn, type, text, opus_data, report_time):
audio_data = opus_to_wav(conn, opus_data)
else:
audio_data = None
# 执行上报
manage_report(
# 执行异步上报
await manage_report(
mac_address=conn.device_id,
session_id=conn.session_id,
chat_type=type,
@@ -56,41 +55,49 @@ def opus_to_wav(conn, opus_data):
Returns:
bytes: WAV格式的音频数据
"""
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
decoder = None
try:
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
conn.logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
conn.logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
if not pcm_data:
raise ValueError("没有有效的PCM数据")
if not pcm_data:
raise ValueError("没有有效的PCM数据")
# 创建WAV文件头
pcm_data_bytes = b"".join(pcm_data)
num_samples = len(pcm_data_bytes) // 2 # 16-bit samples
# 创建WAV文件头
pcm_data_bytes = b"".join(pcm_data)
num_samples = len(pcm_data_bytes) // 2 # 16-bit samples
# WAV文件头
wav_header = bytearray()
wav_header.extend(b"RIFF") # ChunkID
wav_header.extend((36 + len(pcm_data_bytes)).to_bytes(4, "little")) # ChunkSize
wav_header.extend(b"WAVE") # Format
wav_header.extend(b"fmt ") # Subchunk1ID
wav_header.extend((16).to_bytes(4, "little")) # Subchunk1Size
wav_header.extend((1).to_bytes(2, "little")) # AudioFormat (PCM)
wav_header.extend((1).to_bytes(2, "little")) # NumChannels
wav_header.extend((16000).to_bytes(4, "little")) # SampleRate
wav_header.extend((32000).to_bytes(4, "little")) # ByteRate
wav_header.extend((2).to_bytes(2, "little")) # BlockAlign
wav_header.extend((16).to_bytes(2, "little")) # BitsPerSample
wav_header.extend(b"data") # Subchunk2ID
wav_header.extend(len(pcm_data_bytes).to_bytes(4, "little")) # Subchunk2Size
# WAV文件头
wav_header = bytearray()
wav_header.extend(b"RIFF") # ChunkID
wav_header.extend((36 + len(pcm_data_bytes)).to_bytes(4, "little")) # ChunkSize
wav_header.extend(b"WAVE") # Format
wav_header.extend(b"fmt ") # Subchunk1ID
wav_header.extend((16).to_bytes(4, "little")) # Subchunk1Size
wav_header.extend((1).to_bytes(2, "little")) # AudioFormat (PCM)
wav_header.extend((1).to_bytes(2, "little")) # NumChannels
wav_header.extend((16000).to_bytes(4, "little")) # SampleRate
wav_header.extend((32000).to_bytes(4, "little")) # ByteRate
wav_header.extend((2).to_bytes(2, "little")) # BlockAlign
wav_header.extend((16).to_bytes(2, "little")) # BitsPerSample
wav_header.extend(b"data") # Subchunk2ID
wav_header.extend(len(pcm_data_bytes).to_bytes(4, "little")) # Subchunk2Size
# 返回完整的WAV数据
return bytes(wav_header) + pcm_data_bytes
# 返回完整的WAV数据
return bytes(wav_header) + pcm_data_bytes
finally:
if decoder is not None:
try:
del decoder
except Exception as e:
conn.logger.bind(tag=TAG).debug(f"释放decoder资源时出错: {e}")
def enqueue_tts_report(conn, text, opus_data):
+159 -126
View File
@@ -4,8 +4,13 @@ import asyncio
from core.utils import textUtils
from core.utils.util import audio_to_data
from core.providers.tts.dto.dto import SentenceType
from core.utils.audioRateController import AudioRateController
TAG = __name__
# 音频帧时长(毫秒)
AUDIO_FRAME_DURATION = 60
# 预缓冲包数量,直接发送以减少延迟
PRE_BUFFER_COUNT = 5
async def sendAudioMessage(conn, sentenceType, audios, text):
@@ -15,7 +20,19 @@ async def sendAudioMessage(conn, sentenceType, audios, text):
await send_tts_message(conn, "start", None)
if sentenceType == SentenceType.FIRST:
await send_tts_message(conn, "sentence_start", text)
# 同一句子的后续消息加入流控队列,其他情况立即发送
if (
hasattr(conn, "audio_rate_controller")
and conn.audio_rate_controller
and getattr(conn, "audio_flow_control", {}).get("sentence_id")
== conn.sentence_id
):
conn.audio_rate_controller.add_message(
lambda: send_tts_message(conn, "sentence_start", text)
)
else:
# 新句子或流控器未初始化,立即发送
await send_tts_message(conn, "sentence_start", text)
await sendAudio(conn, audios)
# 发送句子开始消息
@@ -30,29 +47,27 @@ async def sendAudioMessage(conn, sentenceType, audios, text):
await conn.close()
def calculate_timestamp_and_sequence(conn, start_time, packet_index, frame_duration=60):
async def _wait_for_audio_completion(conn):
"""
计算音频数据包的时间戳和序列号
等待音频队列清空并等待预缓冲包播放完成
Args:
conn: 连接对象
start_time: 起始时间(性能计数器值)
packet_index: 数据包索引
frame_duration: 帧时长(毫秒),匹配 Opus 编码
Returns:
tuple: (timestamp, sequence)
"""
# 计算时间戳(使用播放位置计算)
timestamp = int((start_time + packet_index * frame_duration / 1000) * 1000) % (
2**32
)
if hasattr(conn, "audio_rate_controller") and conn.audio_rate_controller:
rate_controller = conn.audio_rate_controller
conn.logger.bind(tag=TAG).debug(
f"等待音频发送完成,队列中还有 {len(rate_controller.queue)} 个包"
)
await rate_controller.queue_empty_event.wait()
# 计算序列号
if hasattr(conn, "audio_flow_control"):
sequence = conn.audio_flow_control["sequence"]
else:
sequence = packet_index # 如果没有流控状态,直接使用索引
# 等待预缓冲包播放完成
# 前N个包直接发送,增加2个网络抖动包,需要额外等待它们在客户端播放完成
frame_duration_ms = rate_controller.frame_duration
pre_buffer_playback_time = (PRE_BUFFER_COUNT + 2) * frame_duration_ms / 1000.0
await asyncio.sleep(pre_buffer_playback_time)
return timestamp, sequence
conn.logger.bind(tag=TAG).debug("音频发送完成")
async def _send_to_mqtt_gateway(conn, opus_packet, timestamp, sequence):
@@ -77,135 +92,151 @@ async def _send_to_mqtt_gateway(conn, opus_packet, timestamp, sequence):
await conn.websocket.send(complete_packet)
# 播放音频
async def sendAudio(conn, audios, frame_duration=60):
async def sendAudio(conn, audios, frame_duration=AUDIO_FRAME_DURATION):
"""
发送单个opus包,支持流控
发送音频包,使用 AudioRateController 进行精确的流量控制
Args:
conn: 连接对象
opus_packet: 单个opus数据包
pre_buffer: 快速发送音频
frame_duration: 帧时长(毫秒),匹配 Opus 编码
audios: 单个opus包(bytes) 或 opus包列表
frame_duration: 帧时长(毫秒),默认使用全局常量AUDIO_FRAME_DURATION
"""
if audios is None or len(audios) == 0:
return
# 获取发送延迟配置
send_delay = conn.config.get("tts_audio_send_delay", -1) / 1000.0
is_single_packet = isinstance(audios, bytes)
if isinstance(audios, bytes):
# 重置流控状态,第一次读取和会话发生转变时
if not hasattr(conn, "audio_flow_control") or conn.audio_flow_control.get("sentence_id") != conn.sentence_id:
conn.audio_flow_control = {
"last_send_time": 0,
"packet_count": 0,
"start_time": time.perf_counter(),
"sequence": 0, # 添加序列号
"sentence_id": conn.sentence_id,
}
# 初始化或获取 RateController
rate_controller, flow_control = _get_or_create_rate_controller(
conn, frame_duration, is_single_packet
)
# 统一转换为列表处理
audio_list = [audios] if is_single_packet else audios
# 发送音频包
await _send_audio_with_rate_control(
conn, audio_list, rate_controller, flow_control, send_delay
)
def _get_or_create_rate_controller(conn, frame_duration, is_single_packet):
"""
获取或创建 RateController 和 flow_control
Args:
conn: 连接对象
frame_duration: 帧时长
is_single_packet: 是否单包模式(True: TTS流式单包, False: 批量包)
Returns:
(rate_controller, flow_control)
"""
# 判断是否需要重置:单包模式且 sentence_id 变化,或者控制器不存在
need_reset = (
is_single_packet
and getattr(conn, "audio_flow_control", {}).get("sentence_id")
!= conn.sentence_id
) or not hasattr(conn, "audio_rate_controller")
if need_reset:
# 创建或获取 rate_controller
if not hasattr(conn, "audio_rate_controller"):
conn.audio_rate_controller = AudioRateController(frame_duration)
else:
conn.audio_rate_controller.reset()
# 初始化 flow_control
conn.audio_flow_control = {
"packet_count": 0,
"sequence": 0,
"sentence_id": conn.sentence_id,
}
# 启动后台发送循环
_start_background_sender(
conn, conn.audio_rate_controller, conn.audio_flow_control
)
return conn.audio_rate_controller, conn.audio_flow_control
def _start_background_sender(conn, rate_controller, flow_control):
"""
启动后台发送循环任务
Args:
conn: 连接对象
rate_controller: 速率控制器
flow_control: 流控状态
"""
async def send_callback(packet):
# 检查是否应该中止
if conn.client_abort:
raise asyncio.CancelledError("客户端已中止")
conn.last_activity_time = time.time() * 1000
await _do_send_audio(conn, packet, flow_control)
conn.client_is_speaking = True
# 使用 start_sending 启动后台循环
rate_controller.start_sending(send_callback)
async def _send_audio_with_rate_control(
conn, audio_list, rate_controller, flow_control, send_delay
):
"""
使用 rate_controller 发送音频包
Args:
conn: 连接对象
audio_list: 音频包列表
rate_controller: 速率控制器
flow_control: 流控状态
send_delay: 固定延迟(秒),-1表示使用动态流控
"""
for packet in audio_list:
if conn.client_abort:
return
conn.last_activity_time = time.time() * 1000
# 预缓冲:前5个包直接发送,不做延迟
pre_buffer_count = 5
flow_control = conn.audio_flow_control
current_time = time.perf_counter()
if flow_control["packet_count"] < pre_buffer_count:
# 预缓冲阶段,直接发送不延迟
pass
# 预缓冲:前N个包直接发送
if flow_control["packet_count"] < PRE_BUFFER_COUNT:
await _do_send_audio(conn, packet, flow_control)
conn.client_is_speaking = True
elif send_delay > 0:
# 使用固定延迟
# 固定延迟模式
await asyncio.sleep(send_delay)
await _do_send_audio(conn, packet, flow_control)
conn.client_is_speaking = True
else:
effective_packet = flow_control["packet_count"] - pre_buffer_count
expected_time = flow_control["start_time"] + (
effective_packet * frame_duration / 1000
)
delay = expected_time - current_time
if delay > 0:
await asyncio.sleep(delay)
else:
# 纠正误差
flow_control["start_time"] += abs(delay)
# 动态流控模式:仅添加到队列,由后台循环负责发送
rate_controller.add_audio(packet)
if conn.conn_from_mqtt_gateway:
# 计算时间戳和序列号
timestamp, sequence = calculate_timestamp_and_sequence(
conn,
flow_control["start_time"],
flow_control["packet_count"],
frame_duration,
)
# 调用通用函数发送带头部的数据包
await _send_to_mqtt_gateway(conn, audios, timestamp, sequence)
else:
# 直接发送opus数据包,不添加头部
await conn.websocket.send(audios)
conn.client_is_speaking = True
# 更新流控状态
flow_control["packet_count"] += 1
flow_control["sequence"] += 1
flow_control["last_send_time"] = time.perf_counter()
async def _do_send_audio(conn, opus_packet, flow_control):
"""
执行实际的音频发送
"""
packet_index = flow_control.get("packet_count", 0)
sequence = flow_control.get("sequence", 0)
if conn.conn_from_mqtt_gateway:
# 计算时间戳(基于播放位置)
start_time = time.time()
timestamp = int(start_time * 1000) % (2**32)
await _send_to_mqtt_gateway(conn, opus_packet, timestamp, sequence)
else:
# 文件型音频走普通播放
start_time = time.perf_counter()
play_position = 0
# 直接发送opus数据包
await conn.websocket.send(opus_packet)
# 执行预缓冲
pre_buffer_frames = min(5, len(audios))
for i in range(pre_buffer_frames):
if conn.conn_from_mqtt_gateway:
# 计算时间戳和序列号
timestamp, sequence = calculate_timestamp_and_sequence(
conn, start_time, i, frame_duration
)
# 调用通用函数发送带头部的数据包
await _send_to_mqtt_gateway(conn, audios[i], timestamp, sequence)
else:
# 直接发送预缓冲包,不添加头部
await conn.websocket.send(audios[i])
conn.client_is_speaking = True
remaining_audios = audios[pre_buffer_frames:]
# 播放剩余音频帧
for i, opus_packet in enumerate(remaining_audios):
if conn.client_abort:
break
# 重置没有声音的状态
conn.last_activity_time = time.time() * 1000
if send_delay > 0:
# 固定延迟模式
await asyncio.sleep(send_delay)
else:
# 计算预期发送时间
expected_time = start_time + (play_position / 1000)
current_time = time.perf_counter()
delay = expected_time - current_time
if delay > 0:
await asyncio.sleep(delay)
if conn.conn_from_mqtt_gateway:
# 计算时间戳和序列号(使用当前的数据包索引确保连续性)
packet_index = pre_buffer_frames + i
timestamp, sequence = calculate_timestamp_and_sequence(
conn, start_time, packet_index, frame_duration
)
# 调用通用函数发送带头部的数据包
await _send_to_mqtt_gateway(conn, opus_packet, timestamp, sequence)
else:
# 直接发送opus数据包,不添加头部
await conn.websocket.send(opus_packet)
conn.client_is_speaking = True
play_position += frame_duration
# 更新流控状态
flow_control["packet_count"] = packet_index + 1
flow_control["sequence"] = sequence + 1
async def send_tts_message(conn, state, text=None):
@@ -224,8 +255,10 @@ async def send_tts_message(conn, state, text=None):
stop_tts_notify_voice = conn.config.get(
"stop_tts_notify_voice", "config/assets/tts_notify.mp3"
)
audios = audio_to_data(stop_tts_notify_voice, is_opus=True)
audios = await audio_to_data(stop_tts_notify_voice, is_opus=True)
await sendAudio(conn, audios)
# 等待所有音频包发送完成
await _wait_for_audio_completion(conn)
# 清除服务端讲话状态
conn.clearSpeakStatus()
@@ -1,12 +1,14 @@
import time
import asyncio
from typing import Dict, Any
from core.handle.receiveAudioHandle import handleAudioMessage, startToChat
from core.handle.receiveAudioHandle import startToChat
from core.handle.reportHandle import enqueue_asr_report
from core.handle.sendAudioHandle import send_stt_message, send_tts_message
from core.handle.textMessageHandler import TextMessageHandler
from core.handle.textMessageType import TextMessageType
from core.utils.util import remove_punctuation_and_length
from core.providers.asr.dto.dto import InterfaceType
TAG = __name__
@@ -29,8 +31,18 @@ class ListenTextMessageHandler(TextMessageHandler):
elif msg_json["state"] == "stop":
conn.client_have_voice = True
conn.client_voice_stop = True
if len(conn.asr_audio) > 0:
await handleAudioMessage(conn, b"")
if conn.asr.interface_type == InterfaceType.STREAM:
# 流式模式下,发送结束请求
asyncio.create_task(conn.asr._send_stop_request())
else:
# 非流式模式:直接触发ASR识别
if len(conn.asr_audio) > 0:
asr_audio_task = conn.asr_audio.copy()
conn.asr_audio.clear()
conn.reset_vad_states()
if len(asr_audio_task) > 0:
await conn.asr.handle_voice_stop(conn, asr_audio_task)
elif msg_json["state"] == "detect":
conn.client_have_voice = False
conn.asr_audio.clear()
@@ -57,6 +69,7 @@ class ListenTextMessageHandler(TextMessageHandler):
enqueue_asr_report(conn, "嘿,你好呀", [])
await startToChat(conn, "嘿,你好呀")
else:
conn.just_woken_up = True
# 上报纯文字数据(复用ASR上报功能,但不提供音频数据)
enqueue_asr_report(conn, original_text, [])
# 否则需要LLM对文字内容进行答复
@@ -0,0 +1,45 @@
import json
import time
from typing import Dict, Any
from core.handle.textMessageHandler import TextMessageHandler
from core.handle.textMessageType import TextMessageType
TAG = __name__
class PingMessageHandler(TextMessageHandler):
"""Ping消息处理器,用于保持WebSocket连接"""
@property
def message_type(self) -> TextMessageType:
return TextMessageType.PING
async def handle(self, conn, msg_json: Dict[str, Any]) -> None:
"""
处理PING消息,发送PONG响应
消息格式:{"type": "ping"}
Args:
conn: WebSocket连接对象
msg_json: PING消息的JSON数据
"""
# 检查是否启用了WebSocket心跳功能
enable_websocket_ping = conn.config.get("enable_websocket_ping", False)
if not enable_websocket_ping:
conn.logger.debug(f"WebSocket心跳功能未启用,忽略PING消息")
return
try:
conn.logger.debug(f"收到PING消息,发送PONG响应")
conn.last_activity_time = time.time() * 1000
# 构造PONG响应消息
pong_message = {
"type": "pong",
"timestamp": time.strftime("%Y-%m-%d %H:%M:%S", time.localtime()),
}
# 发送PONG响应
await conn.websocket.send(json.dumps(pong_message))
except Exception as e:
conn.logger.error(f"处理PING消息时发生错误: {e}")
@@ -7,6 +7,7 @@ from core.handle.textHandler.listenMessageHandler import ListenTextMessageHandle
from core.handle.textHandler.mcpMessageHandler import McpTextMessageHandler
from core.handle.textMessageHandler import TextMessageHandler
from core.handle.textHandler.serverMessageHandler import ServerTextMessageHandler
from core.handle.textHandler.pingMessageHandler import PingMessageHandler
TAG = __name__
@@ -27,6 +28,7 @@ class TextMessageHandlerRegistry:
IotTextMessageHandler(),
McpTextMessageHandler(),
ServerTextMessageHandler(),
PingMessageHandler(),
]
for handler in handlers:
@@ -9,3 +9,4 @@ class TextMessageType(Enum):
IOT = "iot"
MCP = "mcp"
SERVER = "server"
PING = "ping"
+49 -27
View File
@@ -33,38 +33,60 @@ class SimpleHttpServer:
return f"ws://{local_ip}:{port}/xiaozhi/v1/"
async def start(self):
server_config = self.config["server"]
read_config_from_api = self.config.get("read_config_from_api", False)
host = server_config.get("ip", "0.0.0.0")
port = int(server_config.get("http_port", 8003))
try:
server_config = self.config["server"]
read_config_from_api = self.config.get("read_config_from_api", False)
host = server_config.get("ip", "0.0.0.0")
port = int(server_config.get("http_port", 8003))
if port:
app = web.Application()
if port:
app = web.Application()
if not read_config_from_api:
# 如果没有开启智控台,只是单模块运行,就需要再添加简单OTA接口,用于下发websocket接口
if not read_config_from_api:
# 如果没有开启智控台,只是单模块运行,就需要再添加简单OTA接口,用于下发websocket接口
app.add_routes(
[
web.get("/xiaozhi/ota/", self.ota_handler.handle_get),
web.post("/xiaozhi/ota/", self.ota_handler.handle_post),
web.options(
"/xiaozhi/ota/", self.ota_handler.handle_options
),
# 下载接口,仅提供 data/bin/*.bin 下载
web.get(
"/xiaozhi/ota/download/{filename}",
self.ota_handler.handle_download,
),
web.options(
"/xiaozhi/ota/download/{filename}",
self.ota_handler.handle_options,
),
]
)
# 添加路由
app.add_routes(
[
web.get("/xiaozhi/ota/", self.ota_handler.handle_get),
web.post("/xiaozhi/ota/", self.ota_handler.handle_post),
web.options("/xiaozhi/ota/", self.ota_handler.handle_post),
web.get("/mcp/vision/explain", self.vision_handler.handle_get),
web.post(
"/mcp/vision/explain", self.vision_handler.handle_post
),
web.options(
"/mcp/vision/explain", self.vision_handler.handle_options
),
]
)
# 添加路由
app.add_routes(
[
web.get("/mcp/vision/explain", self.vision_handler.handle_get),
web.post("/mcp/vision/explain", self.vision_handler.handle_post),
web.options("/mcp/vision/explain", self.vision_handler.handle_post),
]
)
# 运行服务
runner = web.AppRunner(app)
await runner.setup()
site = web.TCPSite(runner, host, port)
await site.start()
# 运行服务
runner = web.AppRunner(app)
await runner.setup()
site = web.TCPSite(runner, host, port)
await site.start()
# 保持服务运行
while True:
await asyncio.sleep(3600) # 每隔 1 小时检查一次
# 保持服务运行
while True:
await asyncio.sleep(3600) # 每隔 1 小时检查一次
except Exception as e:
self.logger.bind(tag=TAG).error(f"HTTP服务器启动失败: {e}")
import traceback
self.logger.bind(tag=TAG).error(f"错误堆栈: {traceback.format_exc()}")
raise
@@ -8,8 +8,6 @@ import asyncio
import requests
import websockets
import opuslib_next
import random
from typing import Optional, Tuple, List
from urllib import parse
from datetime import datetime
from config.logger import setup_logging
@@ -139,13 +137,13 @@ class ASRProvider(ASRProviderBase):
conn.asr_audio.append(audio)
conn.asr_audio = conn.asr_audio[-10:]
# 只在有声音且没有连接时建立连接
if audio_have_voice and not self.is_processing:
# 只在有声音且没有连接时建立连接(排除正在停止的情况)
if audio_have_voice and not self.is_processing and not self.asr_ws:
try:
await self._start_recognition(conn)
except Exception as e:
logger.bind(tag=TAG).error(f"开始识别失败: {str(e)}")
await self._cleanup(conn)
await self._cleanup()
return
if self.asr_ws and self.is_processing and self.server_ready:
@@ -185,10 +183,8 @@ class ASRProvider(ASRProviderBase):
"header": {
"namespace": "SpeechTranscriber",
"name": "StartTranscription",
"status": 20000000,
"message_id": uuid.uuid4().hex,
"task_id": self.task_id,
"status_text": "Gateway:SUCCESS:Success.",
"appkey": self.appkey
},
"payload": {
@@ -207,18 +203,21 @@ class ASRProvider(ASRProviderBase):
async def _forward_results(self, conn):
"""转发识别结果"""
try:
while self.asr_ws and not conn.stop_event.is_set():
while not conn.stop_event.is_set():
try:
response = await asyncio.wait_for(self.asr_ws.recv(), timeout=1.0)
result = json.loads(response)
header = result.get("header", {})
payload = result.get("payload", {})
message_name = header.get("name", "")
status = header.get("status", 0)
if status != 20000000:
if status in [40000004, 40010004]: # 连接超时或客户端断开
if status == 40010004:
logger.bind(tag=TAG).warning(f"请在服务端响应完成后再关闭链接,状态码: {status}")
break
if status in [40000004, 40010003]: # 连接超时或客户端断开
logger.bind(tag=TAG).warning(f"连接问题,状态码: {status}")
break
elif status in [40270002, 40270003]: # 音频问题
@@ -227,12 +226,12 @@ class ASRProvider(ASRProviderBase):
else:
logger.bind(tag=TAG).error(f"识别错误,状态码: {status}, 消息: {header.get('status_text', '')}")
continue
# 收到TranscriptionStarted表示服务器准备好接收音频数据
if message_name == "TranscriptionStarted":
self.server_ready = True
logger.bind(tag=TAG).debug("服务器已准备,开始发送缓存音频...")
# 发送缓存音频
if conn.asr_audio:
for cached_audio in conn.asr_audio[-10:]:
@@ -243,89 +242,89 @@ class ASRProvider(ASRProviderBase):
logger.bind(tag=TAG).warning(f"发送缓存音频失败: {e}")
break
continue
if message_name == "TranscriptionResultChanged":
# 中间结果
text = payload.get("result", "")
if text:
self.text = text
elif message_name == "SentenceEnd":
# 最终结果
# 句子结束(每个句子都会触发)
text = payload.get("result", "")
if text:
self.text = text
conn.reset_vad_states()
# 传递缓存的音频数据
audio_data = getattr(conn, 'asr_audio_for_voiceprint', [])
await self.handle_voice_stop(conn, audio_data)
# 清空缓存
conn.asr_audio_for_voiceprint = []
break
elif message_name == "TranscriptionCompleted":
# 识别完成
self.is_processing = False
break
logger.bind(tag=TAG).info(f"识别到文本: {text}")
# 手动模式下累积识别结果
if conn.client_listen_mode == "manual":
if self.text:
self.text += text
else:
self.text = text
# 手动模式下,只有在收到stop信号后才触发处理(仅处理一次)
if conn.client_voice_stop:
audio_data = getattr(conn, 'asr_audio_for_voiceprint', [])
if len(audio_data) > 0:
logger.bind(tag=TAG).debug("收到最终识别结果,触发处理")
await self.handle_voice_stop(conn, audio_data)
# 清理音频缓存
conn.asr_audio.clear()
conn.reset_vad_states()
break
else:
# 自动模式下直接覆盖
self.text = text
conn.reset_vad_states()
audio_data = getattr(conn, 'asr_audio_for_voiceprint', [])
await self.handle_voice_stop(conn, audio_data)
break
except asyncio.TimeoutError:
continue
except websockets.exceptions.ConnectionClosed:
logger.bind(tag=TAG).error("接收结果超时")
break
except websockets.ConnectionClosed:
logger.bind(tag=TAG).info("ASR服务连接已关闭")
self.is_processing = False
break
except Exception as e:
logger.bind(tag=TAG).error(f"处理结果失败: {str(e)}")
break
except Exception as e:
logger.bind(tag=TAG).error(f"结果转发失败: {str(e)}")
finally:
await self._cleanup(conn)
# 清理连接的音频缓存
await self._cleanup()
if conn:
if hasattr(conn, 'asr_audio_for_voiceprint'):
conn.asr_audio_for_voiceprint = []
if hasattr(conn, 'asr_audio'):
conn.asr_audio = []
async def _cleanup(self, conn):
"""清理资源"""
logger.bind(tag=TAG).debug(f"开始ASR会话清理 | 当前状态: processing={self.is_processing}, server_ready={self.server_ready}")
# 清理连接的音频缓存
if conn and hasattr(conn, 'asr_audio_for_voiceprint'):
conn.asr_audio_for_voiceprint = []
# 判断是否需要发送终止请求
should_stop = self.is_processing or self.server_ready
# 发送停止识别请求
if self.asr_ws and should_stop:
async def _send_stop_request(self):
"""发送停止识别请求(不关闭连接)"""
if self.asr_ws:
try:
# 先停止音频发送
self.is_processing = False
stop_msg = {
"header": {
"namespace": "SpeechTranscriber",
"name": "StopTranscription",
"status": 20000000,
"message_id": uuid.uuid4().hex,
"task_id": self.task_id,
"status_text": "Client:Stop",
"appkey": self.appkey
}
}
logger.bind(tag=TAG).debug("正在发送ASR终止请求")
logger.bind(tag=TAG).debug("停止识别请求已发送")
await self.asr_ws.send(json.dumps(stop_msg, ensure_ascii=False))
await asyncio.sleep(0.1)
logger.bind(tag=TAG).debug("ASR终止请求已发送")
except Exception as e:
logger.bind(tag=TAG).error(f"ASR终止请求发送失败: {e}")
# 状态重置(在终止请求发送后)
logger.bind(tag=TAG).error(f"发送停止识别请求失败: {e}")
async def _cleanup(self):
"""清理资源(关闭连接)"""
logger.bind(tag=TAG).debug(f"开始ASR会话清理 | 当前状态: processing={self.is_processing}, server_ready={self.server_ready}")
# 状态重置
self.is_processing = False
self.server_ready = False
logger.bind(tag=TAG).debug("ASR状态已重置")
# 清理任务
if self.forward_task and not self.forward_task.done():
self.forward_task.cancel()
try:
await asyncio.wait_for(self.forward_task, timeout=1.0)
except Exception as e:
logger.bind(tag=TAG).debug(f"forward_task取消异常: {e}")
finally:
self.forward_task = None
# 关闭连接
if self.asr_ws:
try:
@@ -336,7 +335,10 @@ class ASRProvider(ASRProviderBase):
logger.bind(tag=TAG).error(f"关闭WebSocket连接失败: {e}")
finally:
self.asr_ws = None
# 清理任务引用
self.forward_task = None
logger.bind(tag=TAG).debug("ASR会话清理完成")
async def speech_to_text(self, opus_data, session_id, audio_format):
@@ -347,4 +349,11 @@ class ASRProvider(ASRProviderBase):
async def close(self):
"""关闭资源"""
await self._cleanup()
await self._cleanup(None)
if hasattr(self, 'decoder') and self.decoder is not None:
try:
del self.decoder
self.decoder = None
logger.bind(tag=TAG).debug("Aliyun decoder resources released")
except Exception as e:
logger.bind(tag=TAG).debug(f"释放Aliyun decoder资源时出错: {e}")
+57 -83
View File
@@ -9,7 +9,6 @@ import asyncio
import traceback
import threading
import opuslib_next
import concurrent.futures
from abc import ABC, abstractmethod
from config.logger import setup_logging
from typing import Optional, Tuple, List
@@ -53,121 +52,89 @@ class ASRProviderBase(ABC):
# 接收音频
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
if conn.client_listen_mode == "manual":
# 手动模式:缓存音频用于ASR识别
conn.asr_audio.append(audio)
else:
have_voice = conn.client_have_voice
conn.asr_audio.append(audio)
if not have_voice and not conn.client_have_voice:
conn.asr_audio = conn.asr_audio[-10:]
return
# 自动/实时模式:使用VAD检测
have_voice = audio_have_voice
if conn.client_voice_stop:
asr_audio_task = conn.asr_audio.copy()
conn.asr_audio.clear()
conn.reset_vad_states()
conn.asr_audio.append(audio)
if not have_voice and not conn.client_have_voice:
conn.asr_audio = conn.asr_audio[-10:]
return
if len(asr_audio_task) > 15:
await self.handle_voice_stop(conn, asr_audio_task)
# 自动模式下通过VAD检测到语音停止时触发识别
if conn.client_voice_stop:
asr_audio_task = conn.asr_audio.copy()
conn.asr_audio.clear()
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: 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
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).debug(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
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, _ = results.get("asr", ("", None))
speaker_name = results.get("voiceprint", None)
# 记录识别结果
asr_task = self.speech_to_text(asr_audio_task, conn.session_id, conn.audio_format)
if conn.voiceprint_provider and wav_data:
voiceprint_task = conn.voiceprint_provider.identify_speaker(wav_data, conn.session_id)
# 并发等待两个结果
asr_result, voiceprint_result = await asyncio.gather(
asr_task, voiceprint_task, return_exceptions=True
)
else:
asr_result = await asr_task
voiceprint_result = None
# 记录识别结果 - 检查是否为异常
if isinstance(asr_result, Exception):
logger.bind(tag=TAG).error(f"ASR识别失败: {asr_result}")
raw_text = ""
else:
raw_text, _ = asr_result
if isinstance(voiceprint_result, Exception):
logger.bind(tag=TAG).error(f"声纹识别失败: {voiceprint_result}")
speaker_name = ""
else:
speaker_name = voiceprint_result
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).debug(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)
@@ -241,6 +208,7 @@ class ASRProviderBase(ABC):
@staticmethod
def decode_opus(opus_data: List[bytes]) -> List[bytes]:
"""将Opus音频数据解码为PCM数据"""
decoder = None
try:
decoder = opuslib_next.Decoder(16000, 1)
pcm_data = []
@@ -265,3 +233,9 @@ class ASRProviderBase(ABC):
except Exception as e:
logger.bind(tag=TAG).error(f"音频解码过程发生错误: {e}")
return []
finally:
if decoder is not None:
try:
del decoder
except Exception as e:
logger.bind(tag=TAG).debug(f"释放decoder资源时出错: {e}")
@@ -18,8 +18,6 @@ class ASRProvider(ASRProviderBase):
self.interface_type = InterfaceType.STREAM
self.config = config
self.text = ""
self.max_retries = 3
self.retry_delay = 2
self.decoder = opuslib_next.Decoder(16000, 1)
self.asr_ws = None
self.forward_task = None
@@ -49,6 +47,8 @@ class ASRProvider(ASRProviderBase):
self.channel = config.get("channel", 1)
self.auth_method = config.get("auth_method", "token")
self.secret = config.get("secret", "access_secret")
end_window_size = config.get("end_window_size")
self.end_window_size = int(end_window_size) if end_window_size else 200
async def open_audio_channels(self, conn):
await super().open_audio_channels(conn)
@@ -56,14 +56,13 @@ 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:
if conn.client_listen_mode != "manual" and 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 = []
@@ -179,6 +178,7 @@ class ASRProvider(ASRProviderBase):
payload.get("audio_info", {}).get("duration", 0) > 2000
and not utterances
and not payload["result"].get("text")
and conn.client_listen_mode != "manual"
):
logger.bind(tag=TAG).error(f"识别文本:空")
self.text = ""
@@ -187,15 +187,44 @@ class ASRProvider(ASRProviderBase):
await self.handle_voice_stop(conn, audio_data)
break
# 专门处理没有文本的识别结果(手动模式下可能已经识别完成但是没松按键)
elif not payload["result"].get("text") and not utterances:
if conn.client_listen_mode == "manual" and conn.client_voice_stop and len(audio_data) > 0:
logger.bind(tag=TAG).debug("消息结束收到停止信号,触发处理")
await self.handle_voice_stop(conn, audio_data)
# 清理音频缓存
conn.asr_audio.clear()
conn.reset_vad_states()
break
for utterance in utterances:
if utterance.get("definite", False):
self.text = utterance["text"]
current_text = utterance["text"]
logger.bind(tag=TAG).info(
f"识别到文本: {self.text}"
f"识别到文本: {current_text}"
)
conn.reset_vad_states()
if len(audio_data) > 15: # 确保有足够音频数据
await self.handle_voice_stop(conn, audio_data)
# 手动模式下累积识别结果
if conn.client_listen_mode == "manual":
if self.text:
self.text += current_text
else:
self.text = current_text
# 在接收消息中途时收到停止信号
if conn.client_voice_stop and len(audio_data) > 0:
logger.bind(tag=TAG).debug("消息中途收到停止信号,触发处理")
await self.handle_voice_stop(conn, audio_data)
# 清理音频缓存
conn.asr_audio.clear()
conn.reset_vad_states()
break
else:
# 自动模式下直接覆盖
self.text = current_text
conn.reset_vad_states()
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")
@@ -227,8 +256,6 @@ class ASRProvider(ASRProviderBase):
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:
@@ -236,6 +263,20 @@ class ASRProvider(ASRProviderBase):
self.asr_ws = None
self.is_processing = False
async def _send_stop_request(self):
"""发送最后一个音频帧以通知服务器结束"""
if self.asr_ws:
try:
# 发送结束标记的音频帧(gzip压缩的空数据)
empty_payload = gzip.compress(b"")
last_audio_request = bytearray(self.generate_last_audio_default_header())
last_audio_request.extend(len(empty_payload).to_bytes(4, "big"))
last_audio_request.extend(empty_payload)
await self.asr_ws.send(last_audio_request)
logger.bind(tag=TAG).debug("已发送结束音频帧")
except Exception as e:
logger.bind(tag=TAG).debug(f"发送结束音频帧时出错: {e}")
def construct_request(self, reqid):
req = {
"app": {
@@ -252,7 +293,7 @@ class ASRProvider(ASRProviderBase):
"sequence": 1,
"boosting_table_name": self.boosting_table_name,
"correct_table_name": self.correct_table_name,
"end_window_size": 200,
"end_window_size": self.end_window_size,
},
"audio": {
"format": self.format,
@@ -370,6 +411,16 @@ class ASRProvider(ASRProviderBase):
pass
self.forward_task = None
self.is_processing = False
# 显式释放decoder资源
if hasattr(self, 'decoder') and self.decoder is not None:
try:
del self.decoder
self.decoder = None
logger.bind(tag=TAG).debug("Doubao decoder resources released")
except Exception as e:
logger.bind(tag=TAG).debug(f"释放Doubao decoder资源时出错: {e}")
# 清理所有连接的音频缓冲区
if hasattr(self, '_connections'):
for conn in self._connections.values():
@@ -377,5 +428,3 @@ class ASRProvider(ASRProviderBase):
conn.asr_audio_for_voiceprint = []
if hasattr(conn, 'asr_audio'):
conn.asr_audio = []
if hasattr(conn, 'has_valid_voice'):
conn.has_valid_voice = False
@@ -1,14 +1,16 @@
import time
import os
import sys
import io
import sys
import time
import shutil
import psutil
import asyncio
from config.logger import setup_logging
from typing import Optional, Tuple, List
from core.providers.asr.base import ASRProviderBase
from funasr import AutoModel
from funasr.utils.postprocess_utils import rich_transcription_postprocess
import shutil
from core.providers.asr.base import ASRProviderBase
from core.providers.asr.dto.dto import InterfaceType
TAG = __name__
@@ -90,16 +92,17 @@ class ASRProvider(ASRProviderBase):
else:
file_path = self.save_audio_to_file(pcm_data, session_id)
# 语音识别
# 语音识别 - 使用线程池避免阻塞事件循环
start_time = time.time()
result = self.model.generate(
result = await asyncio.to_thread(
self.model.generate,
input=combined_pcm_data,
cache={},
language="auto",
use_itn=True,
batch_size_s=60,
)
text = rich_transcription_postprocess(result[0]["text"])
text = await asyncio.to_thread(rich_transcription_postprocess, result[0]["text"])
logger.bind(tag=TAG).debug(
f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}"
)
@@ -1,8 +1,5 @@
import os
import json
import asyncio
import tempfile
import difflib
from typing import Optional, Tuple, List
import dashscope
from config.logger import setup_logging
@@ -16,7 +13,8 @@ logger = setup_logging()
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
super().__init__()
self.interface_type = InterfaceType.STREAM
# 音频文件上传类型,流式文本识别输出
self.interface_type = InterfaceType.NON_STREAM
"""Qwen3-ASR-Flash ASR初始化"""
# 配置参数
@@ -130,27 +128,11 @@ class ASRProvider(ASRProviderBase):
# 处理流式响应
full_text = ""
last_text = "" # 用于存储上一个文本片段
for chunk in response:
try:
text = chunk["output"]["choices"][0]["message"].content[0]["text"]
# 标准化文本片段(去除首尾空格)
normalized_text = text.strip()
# 只有当新文本片段与上一个不同时才处理
if normalized_text != last_text:
# 提取新增的文本部分
# 通过比较当前文本和上一个文本,找到新增的部分
if normalized_text.startswith(last_text):
# 如果当前文本以最后一个文本开头,则新增部分是两者的差集
new_part = normalized_text[len(last_text):]
else:
# 如果不以最后一个文本开头,说明识别结果发生了较大变化,直接使用当前文本
new_part = normalized_text
# 将新增部分添加到完整文本中
full_text += new_part
last_text = normalized_text
# 这里可以实时处理文本片段,例如通过回调函数
# 更新为最新的完整文本
full_text = text.strip()
except:
pass
@@ -5,6 +5,7 @@ import hashlib
import asyncio
import websockets
import opuslib_next
import gc
from time import mktime
from datetime import datetime
from urllib.parse import urlencode
@@ -34,9 +35,6 @@ class ASRProvider(ASRProviderBase):
self.forward_task = None
self.is_processing = False
self.server_ready = False
self.last_frame_sent = False # 标记是否已发送最终帧
self.best_text = "" # 保存最佳识别结果
self.has_final_result = False # 标记是否收到最终识别结果
# 讯飞配置
self.app_id = config.get("app_id")
@@ -51,7 +49,6 @@ class ASRProvider(ASRProviderBase):
"domain": config.get("domain", "slm"),
"language": config.get("language", "zh_cn"),
"accent": config.get("accent", "mandarin"),
"dwa": config.get("dwa", "wpgs"),
"result": {"encoding": "utf8", "compress": "raw", "format": "plain"},
}
@@ -115,7 +112,7 @@ class ASRProvider(ASRProviderBase):
await self._start_recognition(conn)
except Exception as e:
logger.bind(tag=TAG).error(f"建立ASR连接失败: {str(e)}")
await self._cleanup(conn)
await self._cleanup()
return
# 发送当前音频数据
@@ -125,7 +122,7 @@ class ASRProvider(ASRProviderBase):
await self._send_audio_frame(pcm_frame, STATUS_CONTINUE_FRAME)
except Exception as e:
logger.bind(tag=TAG).warning(f"发送音频数据时发生错误: {e}")
await self._cleanup(conn)
await self._cleanup()
async def _start_recognition(self, conn):
"""开始识别会话"""
@@ -135,6 +132,10 @@ class ASRProvider(ASRProviderBase):
ws_url = self.create_url()
logger.bind(tag=TAG).info(f"正在连接ASR服务: {ws_url[:50]}...")
# 如果为手动模式,设置超时时长为一分钟
if conn.client_listen_mode == "manual":
self.iat_params["eos"] = 60000
self.asr_ws = await websockets.connect(
ws_url,
max_size=1000000000,
@@ -145,8 +146,6 @@ class ASRProvider(ASRProviderBase):
logger.bind(tag=TAG).info("ASR WebSocket连接已建立")
self.server_ready = False
self.last_frame_sent = False
self.best_text = ""
self.forward_task = asyncio.create_task(self._forward_results(conn))
# 发送首帧音频
@@ -195,23 +194,12 @@ class ASRProvider(ASRProviderBase):
await self.asr_ws.send(json.dumps(frame_data, ensure_ascii=False))
# 标记是否发送了最终帧
if status == STATUS_LAST_FRAME:
self.last_frame_sent = True
logger.bind(tag=TAG).info("标记最终帧已发送")
async def _forward_results(self, conn):
"""转发识别结果"""
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", [])
while not conn.stop_event.is_set():
try:
# 如果已发送最终帧,增加超时时间等待完整结果
timeout = 3.0 if self.last_frame_sent else 30.0
response = await asyncio.wait_for(
self.asr_ws.recv(), timeout=timeout
)
response = await asyncio.wait_for(self.asr_ws.recv(), timeout=60)
result = json.loads(response)
logger.bind(tag=TAG).debug(f"收到ASR结果: {result}")
@@ -235,144 +223,27 @@ class ASRProvider(ASRProviderBase):
# 解码base64文本
decoded_text = base64.b64decode(text_data).decode("utf-8")
text_json = json.loads(decoded_text)
# 提取文本内容
text_ws = text_json.get("ws", [])
result_text = ""
for i in text_ws:
for j in i.get("cw", []):
w = j.get("w", "")
result_text += w
self.text += w
# 更新识别文本 - 实时更新策略
# 只检查是否为空字符串,不再过滤任何标点符号
# 这样可以确保所有识别到的内容,包括标点符号都能被实时更新
if result_text and result_text.strip():
# 实时更新:正常情况下都更新,提高响应速度
should_update = True
# 保存最佳文本
# 1. 如果是识别完成状态或最终帧后收到的结果,优先保存
# 2. 否则保存最长的有意义文本
# 取消对标点符号的过滤,只检查是否为空
# 这样可以保留所有识别到的内容,包括各种标点符号
is_valid_text = len(result_text.strip()) > 0
if (
self.last_frame_sent or status == 2
) and is_valid_text:
self.best_text = result_text
self.has_final_result = True # 标记已收到最终结果
logger.bind(tag=TAG).debug(
f"保存最终识别结果: {self.best_text}"
)
elif (
len(result_text) > len(self.best_text)
and is_valid_text
and not self.has_final_result
):
self.best_text = result_text
logger.bind(tag=TAG).debug(
f"保存中间最佳文本: {self.best_text}"
)
# 如果已发送最终帧,只过滤空文本
if self.last_frame_sent:
# 只拒绝完全空的结果
if not result_text.strip():
should_update = False
logger.bind(tag=TAG).warning(
f"最终帧后拒绝空文本"
)
if should_update:
# 处理流式识别结果,避免简单替换导致内容丢失
# 1. 如果是中间状态(非最终帧后),可能需要替换为更完整的识别
# 2. 如果是最终帧后收到的结果,可能是对前面文本的补充
if self.last_frame_sent:
# 最终帧后收到的结果可能是标点符号等补充内容
# 检查是否需要合并文本而不是替换
# 如果当前文本是纯标点而前面已有内容,应该追加而不是替换
if len(
self.text
) > 0 and result_text.strip() in [
"",
".",
"?",
"",
"!",
"",
",",
"",
";",
"",
]:
# 对于标点符号,追加到现有文本后
self.text = (
self.text.rstrip().rstrip("。.")
+ result_text
)
else:
# 其他情况保持替换逻辑
self.text = result_text
else:
# 中间状态替换为新的识别结果
self.text = result_text
logger.bind(tag=TAG).info(
f"实时更新识别文本: {self.text} (最终帧已发送: {self.last_frame_sent})"
)
# 识别完成,但如果还没发送最终帧,继续等待
if status == 2:
logger.bind(tag=TAG).info(
f"识别完成状态已到达,当前识别文本: {self.text}"
)
# 如果还没发送最终帧,继续等待
if not self.last_frame_sent:
logger.bind(tag=TAG).info(
"识别完成但最终帧未发送,继续等待..."
)
continue
# 已发送最终帧且收到完成状态,使用最佳策略选择最终结果
# 优先使用识别完成状态下的最新结果,而不是仅仅基于长度
if self.best_text:
# 如果当前文本是在最终帧发送后或识别完成状态下收到的,优先使用
if (
self.last_frame_sent or status == 2
) and self.text.strip():
logger.bind(tag=TAG).info(
f"使用完成状态下的最新识别结果: {self.text}"
)
elif len(self.best_text) > len(self.text):
logger.bind(tag=TAG).info(
f"使用更长的最佳文本作为最终结果: {self.text} -> {self.best_text}"
)
self.text = self.best_text
logger.bind(tag=TAG).info(f"获取到最终完整文本: {self.text}")
if conn.client_listen_mode == "manual":
audio_data = getattr(conn, 'asr_audio_for_voiceprint', [])
if len(audio_data) > 0:
logger.bind(tag=TAG).debug("收到最终识别结果,触发处理")
await self.handle_voice_stop(conn, audio_data)
# 清理音频缓存
conn.asr_audio.clear()
conn.reset_vad_states()
if len(audio_data) > 15: # 确保有足够音频数据
# 准备处理结果
pass
break
except asyncio.TimeoutError:
if self.last_frame_sent:
# 超时时也使用最佳文本
if self.best_text and len(self.best_text) > len(self.text):
logger.bind(tag=TAG).info(
f"超时,使用最佳文本: {self.text} -> {self.best_text}"
)
self.text = self.best_text
logger.bind(tag=TAG).info(
f"最终帧后超时,使用结果: {self.text}"
)
break
# 如果还没发送最终帧,继续等待
continue
logger.bind(tag=TAG).error("接收结果超时")
break
except websockets.ConnectionClosed:
logger.bind(tag=TAG).info("ASR服务连接已关闭")
self.is_processing = False
@@ -389,17 +260,15 @@ class ASRProvider(ASRProviderBase):
if hasattr(e, "__cause__") and e.__cause__:
logger.bind(tag=TAG).error(f"错误原因: {str(e.__cause__)}")
finally:
if self.asr_ws:
await self.asr_ws.close()
self.asr_ws = None
self.is_processing = False
# 清理连接资源
await self._cleanup()
# 清理连接的音频缓存
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
async def handle_voice_stop(self, conn, asr_audio_task: List[bytes]):
"""处理语音停止,发送最后一帧并处理识别结果"""
@@ -407,22 +276,13 @@ class ASRProvider(ASRProviderBase):
# 先发送最后一帧表示音频结束
if self.asr_ws and self.is_processing:
try:
# 取最后一个有效的音频帧作为最后一帧数据
last_frame = b""
if asr_audio_task:
last_audio = asr_audio_task[-1]
last_frame = self.decoder.decode(last_audio, 960)
await self._send_audio_frame(last_frame, STATUS_LAST_FRAME)
logger.bind(tag=TAG).info("已发送最后一帧")
await self._send_audio_frame(b"", STATUS_LAST_FRAME)
logger.bind(tag=TAG).debug(f"已发送停止请求")
# 发送最终帧后,给_forward_results适当时间处理最终结果
await asyncio.sleep(0.25)
logger.bind(tag=TAG).info(f"准备处理最终识别结果: {self.text}")
except Exception as e:
logger.bind(tag=TAG).error(f"发送最后一帧失败: {e}")
logger.bind(tag=TAG).error(f"发送停止请求失败: {e}")
# 调用父类的handle_voice_stop方法处理识别结果
await super().handle_voice_stop(conn, asr_audio_task)
except Exception as e:
logger.bind(tag=TAG).error(f"处理语音停止失败: {e}")
@@ -436,40 +296,27 @@ class ASRProvider(ASRProviderBase):
self.asr_ws = None
self.is_processing = False
async def _cleanup(self, conn):
"""清理资源"""
logger.bind(tag=TAG).info(
async def _send_stop_request(self):
"""发送停止识别请求(不关闭连接)"""
if self.asr_ws:
try:
# 先停止音频发送
self.is_processing = False
await self._send_audio_frame(b"", STATUS_LAST_FRAME)
logger.bind(tag=TAG).debug("已发送停止请求")
except Exception as e:
logger.bind(tag=TAG).error(f"发送停止请求失败: {e}")
async def _cleanup(self):
"""清理资源(关闭连接)"""
logger.bind(tag=TAG).debug(
f"开始ASR会话清理 | 当前状态: processing={self.is_processing}, server_ready={self.server_ready}"
)
# 发送最后一帧
if self.asr_ws and self.is_processing:
try:
await self._send_audio_frame(b"", STATUS_LAST_FRAME)
await asyncio.sleep(0.1)
logger.bind(tag=TAG).info("已发送最后一帧")
except Exception as e:
logger.bind(tag=TAG).error(f"发送最后一帧失败: {e}")
# 状态重置
self.is_processing = False
self.server_ready = False
self.last_frame_sent = False
self.best_text = ""
self.has_final_result = False
logger.bind(tag=TAG).info("ASR状态已重置")
# 清理任务
if self.forward_task and not self.forward_task.done():
self.forward_task.cancel()
try:
await asyncio.wait_for(self.forward_task, timeout=1.0)
except asyncio.CancelledError:
pass
except Exception as e:
logger.bind(tag=TAG).debug(f"forward_task取消异常: {e}")
finally:
self.forward_task = None
logger.bind(tag=TAG).debug("ASR状态已重置")
# 关闭连接
if self.asr_ws:
@@ -482,16 +329,10 @@ class ASRProvider(ASRProviderBase):
finally:
self.asr_ws = None
# 清理连接的音频缓存
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
# 清理任务引用
self.forward_task = None
logger.bind(tag=TAG).info("ASR会话清理完成")
logger.bind(tag=TAG).debug("ASR会话清理完成")
async def speech_to_text(self, opus_data, session_id, audio_format):
"""获取识别结果"""
@@ -512,6 +353,16 @@ class ASRProvider(ASRProviderBase):
pass
self.forward_task = None
self.is_processing = False
# 显式释放decoder资源
if hasattr(self, 'decoder') and self.decoder is not None:
try:
del self.decoder
self.decoder = None
logger.bind(tag=TAG).debug("Xunfei decoder resources released")
except Exception as e:
logger.bind(tag=TAG).debug(f"释放Xunfei decoder资源时出错: {e}")
# 清理所有连接的音频缓冲区
if hasattr(self, "_connections"):
for conn in self._connections.values():
@@ -519,5 +370,3 @@ class ASRProvider(ASRProviderBase):
conn.asr_audio_for_voiceprint = []
if hasattr(conn, "asr_audio"):
conn.asr_audio = []
if hasattr(conn, "has_valid_voice"):
conn.has_valid_voice = False
@@ -24,7 +24,6 @@ class LLMProvider(LLMProviderBase):
"max_tokens": int,
"temperature": lambda x: round(float(x), 1),
"top_p": lambda x: round(float(x), 1),
"top_k": int,
"frequency_penalty": lambda x: round(float(x), 1),
}
@@ -40,7 +39,7 @@ class LLMProvider(LLMProviderBase):
setattr(self, param, None)
logger.debug(
f"意图识别参数初始化: {self.temperature}, {self.max_tokens}, {self.top_p}, {self.top_k}, {self.frequency_penalty}"
f"意图识别参数初始化: {self.temperature}, {self.max_tokens}, {self.top_p}, {self.frequency_penalty}"
)
model_key_msg = check_model_key("LLM", self.api_key)
@@ -71,7 +70,6 @@ class LLMProvider(LLMProviderBase):
"max_tokens": kwargs.get("max_tokens", self.max_tokens),
"temperature": kwargs.get("temperature", self.temperature),
"top_p": kwargs.get("top_p", self.top_p),
"top_k": kwargs.get("top_k", self.top_k),
"frequency_penalty": kwargs.get("frequency_penalty", self.frequency_penalty),
}
@@ -116,7 +114,6 @@ class LLMProvider(LLMProviderBase):
"max_tokens": kwargs.get("max_tokens", self.max_tokens),
"temperature": kwargs.get("temperature", self.temperature),
"top_p": kwargs.get("top_p", self.top_p),
"top_k": kwargs.get("top_k", self.top_k),
"frequency_penalty": kwargs.get("frequency_penalty", self.frequency_penalty),
}
@@ -14,7 +14,7 @@ class MemoryProviderBase(ABC):
self.llm = llm
@abstractmethod
async def save_memory(self, msgs):
async def save_memory(self, msgs, session_id=None):
"""Save a new memory for specific role and return memory ID"""
print("this is base func", msgs)
@@ -28,7 +28,7 @@ class MemoryProvider(MemoryProviderBase):
logger.bind(tag=TAG).error(f"详细错误: {traceback.format_exc()}")
self.use_mem0 = False
async def save_memory(self, msgs):
async def save_memory(self, msgs, session_id=None):
if not self.use_mem0:
return None
if len(msgs) < 2:
@@ -41,9 +41,7 @@ class MemoryProvider(MemoryProviderBase):
for message in msgs
if message.role != "system"
]
result = self.client.add(
messages, user_id=self.role_id
)
result = self.client.add(messages, user_id=self.role_id)
logger.bind(tag=TAG).debug(f"Save memory result: {result}")
except Exception as e:
logger.bind(tag=TAG).error(f"保存记忆失败: {str(e)}")
@@ -4,7 +4,8 @@ import json
import os
import yaml
from config.config_loader import get_project_dir
from config.manage_api_client import save_mem_local_short
from config.manage_api_client import generate_and_save_chat_summary
import asyncio
from core.utils.util import check_model_key
@@ -74,18 +75,6 @@ short_term_memory_prompt = """
```
"""
short_term_memory_prompt_only_content = """
你是一个经验丰富的记忆总结者,擅长将对话内容进行总结摘要,遵循以下规则:
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、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
"""
def extract_json_data(json_code):
start = json_code.find("```json")
@@ -143,7 +132,7 @@ class MemoryProvider(MemoryProviderBase):
with open(self.memory_path, "w", encoding="utf-8") as f:
yaml.dump(all_memory, f, allow_unicode=True)
async def save_memory(self, msgs):
async def save_memory(self, msgs, session_id=None):
# 打印使用的模型信息
model_info = getattr(self.llm, "model_name", str(self.llm.__class__.__name__))
logger.bind(tag=TAG).debug(f"使用记忆保存模型: {model_info}")
@@ -187,14 +176,12 @@ class MemoryProvider(MemoryProviderBase):
except Exception as e:
print("Error:", e)
else:
result = self.llm.response_no_stream(
short_term_memory_prompt_only_content,
msgStr,
max_tokens=2000,
temperature=0.2,
)
save_mem_local_short(self.role_id, result)
logger.bind(tag=TAG).info(f"Save memory successful - Role: {self.role_id}")
# 当save_to_file为False时,调用Java端的聊天记录总结接口
summary_id = session_id if session_id else self.role_id
await generate_and_save_chat_summary(summary_id)
logger.bind(tag=TAG).info(
f"Save memory successful - Role: {self.role_id}, Session: {session_id}"
)
return self.short_memory
@@ -11,7 +11,7 @@ class MemoryProvider(MemoryProviderBase):
def __init__(self, config, summary_memory=None):
super().__init__(config)
async def save_memory(self, msgs):
async def save_memory(self, msgs, session_id=None):
logger.bind(tag=TAG).debug("nomem mode: No memory saving is performed.")
return None
@@ -3,12 +3,8 @@
import asyncio
import os
import json
from datetime import timedelta
from typing import Dict, Any, List
from mcp import Implementation
from mcp.client.session import SamplingFnT, ElicitationFnT, ListRootsFnT, LoggingFnT, MessageHandlerFnT
from mcp.shared.session import ProgressFnT
from mcp.types import LoggingMessageNotificationParams
from config.config_loader import get_project_dir
@@ -33,6 +29,7 @@ class ServerMCPManager:
)
self.clients: Dict[str, ServerMCPClient] = {}
self.tools = []
self._init_lock = asyncio.Lock()
def load_config(self) -> Dict[str, Any]:
"""加载MCP服务配置"""
@@ -49,29 +46,50 @@ class ServerMCPManager:
)
return {}
async def _init_server(self, name: str, srv_config: Dict[str, Any]):
"""初始化单个MCP服务"""
client = None
try:
# 初始化服务端MCP客户端
logger.bind(tag=TAG).info(f"初始化服务端MCP客户端: {name}")
client = ServerMCPClient(srv_config)
# 设置超时时间10秒
await asyncio.wait_for(client.initialize(logging_callback=self.logging_callback), timeout=10)
# 使用锁保护共享状态的修改
async with self._init_lock:
self.clients[name] = client
client_tools = client.get_available_tools()
self.tools.extend(client_tools)
except asyncio.TimeoutError:
logger.bind(tag=TAG).error(
f"Failed to initialize MCP server {name}: Timeout"
)
if client:
await client.cleanup()
except Exception as e:
logger.bind(tag=TAG).error(
f"Failed to initialize MCP server {name}: {e}"
)
if client:
await client.cleanup()
async def initialize_servers(self) -> None:
"""初始化所有MCP服务"""
config = self.load_config()
tasks = []
for name, srv_config in config.items():
if not srv_config.get("command") and not srv_config.get("url"):
logger.bind(tag=TAG).warning(
f"Skipping server {name}: neither command nor url specified"
)
continue
try:
# 初始化服务端MCP客户端
logger.bind(tag=TAG).info(f"初始化服务端MCP客户端: {name}")
client = ServerMCPClient(srv_config)
await client.initialize(logging_callback=self.logging_callback)
self.clients[name] = client
client_tools = client.get_available_tools()
self.tools.extend(client_tools)
except Exception as e:
logger.bind(tag=TAG).error(
f"Failed to initialize MCP server {name}: {e}"
)
tasks.append(self._init_server(name, srv_config))
if tasks:
await asyncio.gather(*tasks)
# 输出当前支持的服务端MCP工具列表
if hasattr(self.conn, "func_handler") and self.conn.func_handler:
@@ -36,7 +36,18 @@ class VADProvider(VADProviderBase):
# 至少要多少帧才算有语音
self.frame_window_threshold = 3
def __del__(self):
if hasattr(self, 'decoder') and self.decoder is not None:
try:
del self.decoder
except Exception:
pass
def is_vad(self, conn, opus_packet):
# 手动模式:直接返回True,不进行实时VAD检测,所有音频都缓存
if conn.client_listen_mode == "manual":
return True
try:
pcm_frame = self.decoder.decode(opus_packet, 960)
conn.client_audio_buffer.extend(pcm_frame) # 将新数据加入缓冲区
@@ -0,0 +1,160 @@
import time
import asyncio
from collections import deque
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class AudioRateController:
"""
音频速率控制器 - 按照60ms帧时长精确控制音频发送
解决高并发下的时间累积误差问题
"""
def __init__(self, frame_duration=60):
"""
Args:
frame_duration: 单个音频帧时长(毫秒),默认60ms
"""
self.frame_duration = frame_duration
self.queue = deque()
self.play_position = 0 # 虚拟播放位置(毫秒)
self.start_timestamp = None # 开始时间戳(只读,不修改)
self.pending_send_task = None
self.logger = logger
self.queue_empty_event = asyncio.Event() # 队列清空事件
self.queue_empty_event.set() # 初始为空状态
self.queue_has_data_event = asyncio.Event() # 队列数据事件
def reset(self):
"""重置控制器状态"""
if self.pending_send_task and not self.pending_send_task.done():
self.pending_send_task.cancel()
# 取消任务后,任务会在下次事件循环时清理,无需阻塞等待
self.queue.clear()
self.play_position = 0
self.start_timestamp = None # 由首个音频包设置
# 相关事件处理
self.queue_empty_event.set()
self.queue_has_data_event.clear()
def add_audio(self, opus_packet):
"""添加音频包到队列"""
self.queue.append(("audio", opus_packet))
# 相关事件处理
self.queue_empty_event.clear()
self.queue_has_data_event.set()
def add_message(self, message_callback):
"""
添加消息到队列(立即发送,不占用播放时间)
Args:
message_callback: 消息发送回调函数 async def()
"""
self.queue.append(("message", message_callback))
# 相关事件处理
self.queue_empty_event.clear()
self.queue_has_data_event.set()
def _get_elapsed_ms(self):
"""获取已经过的时间(毫秒)"""
if self.start_timestamp is None:
return 0
return (time.monotonic() - self.start_timestamp) * 1000
async def check_queue(self, send_audio_callback):
"""
检查队列并按时发送音频/消息
Args:
send_audio_callback: 发送音频的回调函数 async def(opus_packet)
"""
while self.queue:
item = self.queue[0]
item_type = item[0]
if item_type == "message":
# 消息类型:立即发送,不占用播放时间
_, message_callback = item
self.queue.popleft()
try:
await message_callback()
except Exception as e:
self.logger.bind(tag=TAG).error(f"发送消息失败: {e}")
raise
elif item_type == "audio":
if self.start_timestamp is None:
self.start_timestamp = time.monotonic()
_, opus_packet = item
# 循环等待直到时间到达
while True:
# 计算时间差
elapsed_ms = self._get_elapsed_ms()
output_ms = self.play_position
if elapsed_ms < output_ms:
# 还不到发送时间,计算等待时长
wait_ms = output_ms - elapsed_ms
# 等待后继续检查(允许被中断)
try:
await asyncio.sleep(wait_ms / 1000)
except asyncio.CancelledError:
self.logger.bind(tag=TAG).debug("音频发送任务被取消")
raise
# 等待结束后重新检查时间(循环回到 while True)
else:
# 时间已到,跳出等待循环
break
# 时间已到,从队列移除并发送
self.queue.popleft()
self.play_position += self.frame_duration
try:
await send_audio_callback(opus_packet)
except Exception as e:
self.logger.bind(tag=TAG).error(f"发送音频失败: {e}")
raise
# 队列处理完后清除事件
self.queue_empty_event.set()
self.queue_has_data_event.clear()
def start_sending(self, send_audio_callback):
"""
启动异步发送任务
Args:
send_audio_callback: 发送音频的回调函数
Returns:
asyncio.Task: 发送任务
"""
async def _send_loop():
try:
while True:
# 等待队列数据事件,不轮询等待占用CPU
await self.queue_has_data_event.wait()
await self.check_queue(send_audio_callback)
except asyncio.CancelledError:
self.logger.bind(tag=TAG).debug("音频发送循环已停止")
except Exception as e:
self.logger.bind(tag=TAG).error(f"音频发送循环异常: {e}")
self.pending_send_task = asyncio.create_task(_send_loop())
return self.pending_send_task
def stop_sending(self):
"""停止发送任务"""
if self.pending_send_task and not self.pending_send_task.done():
self.pending_send_task.cancel()
self.logger.bind(tag=TAG).debug("已取消音频发送任务")
+4
View File
@@ -19,6 +19,7 @@ class CacheType(Enum):
CONFIG = "config"
DEVICE_PROMPT = "device_prompt"
VOICEPRINT_HEALTH = "voiceprint_health" # 声纹识别健康检查
AUDIO_DATA = "audio_data" # 音频数据缓存
@dataclass
@@ -58,5 +59,8 @@ class CacheConfig:
CacheType.VOICEPRINT_HEALTH: cls(
strategy=CacheStrategy.TTL, ttl=600, max_size=100 # 10分钟过期
),
CacheType.AUDIO_DATA: cls(
strategy=CacheStrategy.TTL, ttl=600, max_size=100 # 10分钟过期
),
}
return configs.get(cache_type, cls())
@@ -0,0 +1,64 @@
import httpx
from typing import Dict, Any, List
from config.logger import setup_logging
TAG = __name__
class ContextDataProvider:
"""数据上下文填充,负责从配置的API获取数据"""
def __init__(self, config: Dict[str, Any], logger=None):
self.config = config
self.logger = logger or setup_logging()
self.context_data = ""
def fetch_all(self, device_id: str) -> str:
"""获取所有配置的上下文数据"""
context_providers = self.config.get("context_providers", [])
if not context_providers:
return ""
formatted_lines = []
for provider in context_providers:
url = provider.get("url")
headers = provider.get("headers", {})
if not url:
continue
try:
headers = headers.copy() if isinstance(headers, dict) else {}
# 将 device_id 添加到请求头
headers["device-id"] = device_id
# 发送请求
response = httpx.get(url, headers=headers, timeout=3)
if response.status_code == 200:
result = response.json()
if isinstance(result, dict):
if result.get("code") == 0:
data = result.get("data")
# 格式化数据
if isinstance(data, dict):
for k, v in data.items():
formatted_lines.append(f"- **{k}** {v}")
elif isinstance(data, list):
for item in data:
formatted_lines.append(f"- {item}")
else:
formatted_lines.append(f"- {data}")
else:
self.logger.bind(tag=TAG).warning(f"API {url} 返回错误码: {result.get('msg')}")
else:
self.logger.bind(tag=TAG).warning(f"API {url} 返回的不是JSON字典")
else:
self.logger.bind(tag=TAG).warning(f"API {url} 请求失败: {response.status_code}")
except Exception as e:
self.logger.bind(tag=TAG).error(f"获取上下文数据 {url} 失败: {e}")
# 将所有格式化后的行拼接成一个字符串
self.context_data = "\n".join(formatted_lines)
if self.context_data:
self.logger.bind(tag=TAG).debug(f"已注入动态上下文数据:\n{self.context_data}")
return self.context_data
@@ -0,0 +1,122 @@
"""
全局GC管理模块
定期执行垃圾回收,避免频繁触发GC导致的GIL锁问题
"""
import gc
import asyncio
import threading
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class GlobalGCManager:
"""全局垃圾回收管理器"""
def __init__(self, interval_seconds=300):
"""
初始化GC管理器
Args:
interval_seconds: GC执行间隔(秒),默认300秒(5分钟)
"""
self.interval_seconds = interval_seconds
self._task = None
self._stop_event = asyncio.Event()
self._lock = threading.Lock()
async def start(self):
"""启动定时GC任务"""
if self._task is not None:
logger.bind(tag=TAG).warning("GC管理器已经在运行")
return
logger.bind(tag=TAG).info(f"启动全局GC管理器,间隔{self.interval_seconds}")
self._stop_event.clear()
self._task = asyncio.create_task(self._gc_loop())
async def stop(self):
"""停止定时GC任务"""
if self._task is None:
return
logger.bind(tag=TAG).info("停止全局GC管理器")
self._stop_event.set()
if self._task and not self._task.done():
self._task.cancel()
try:
await self._task
except asyncio.CancelledError:
pass
self._task = None
async def _gc_loop(self):
"""GC循环任务"""
try:
while not self._stop_event.is_set():
# 等待指定间隔
try:
await asyncio.wait_for(
self._stop_event.wait(), timeout=self.interval_seconds
)
# 如果stop_event被设置,退出循环
break
except asyncio.TimeoutError:
# 超时表示到了执行GC的时间
pass
# 执行GC
await self._run_gc()
except asyncio.CancelledError:
logger.bind(tag=TAG).info("GC循环任务被取消")
raise
except Exception as e:
logger.bind(tag=TAG).error(f"GC循环任务异常: {e}")
finally:
logger.bind(tag=TAG).info("GC循环任务已退出")
async def _run_gc(self):
"""执行垃圾回收"""
try:
# 在线程池中执行GC,避免阻塞事件循环
loop = asyncio.get_running_loop()
def do_gc():
with self._lock:
before = len(gc.get_objects())
collected = gc.collect()
after = len(gc.get_objects())
return before, collected, after
before, collected, after = await loop.run_in_executor(None, do_gc)
logger.bind(tag=TAG).debug(
f"全局GC执行完成 - 回收对象: {collected}, "
f"对象数量: {before} -> {after}"
)
except Exception as e:
logger.bind(tag=TAG).error(f"执行GC时出错: {e}")
# 全局单例
_gc_manager_instance = None
def get_gc_manager(interval_seconds=300):
"""
获取全局GC管理器实例(单例模式)
Args:
interval_seconds: GC执行间隔(秒),默认300秒(5分钟)
Returns:
GlobalGCManager实例
"""
global _gc_manager_instance
if _gc_manager_instance is None:
_gc_manager_instance = GlobalGCManager(interval_seconds)
return _gc_manager_instance
@@ -102,6 +102,9 @@ class OpusEncoderUtils:
def _encode(self, frame: np.ndarray) -> Optional[bytes]:
"""编码一帧音频数据"""
try:
# 编码器已释放,跳过编码
if not hasattr(self, 'encoder') or self.encoder is None:
return None
# 将numpy数组转换为bytes
frame_bytes = frame.tobytes()
# opuslib要求输入字节数必须是channels*2的倍数
@@ -128,5 +131,9 @@ class OpusEncoderUtils:
def close(self):
"""关闭编码器并释放资源"""
# opuslib没有明确的关闭方法,Python的垃圾回收会处理
pass
if hasattr(self, 'encoder') and self.encoder:
try:
del self.encoder
self.encoder = None
except Exception as e:
logging.error(f"Error releasing Opus encoder: {e}")
@@ -4,7 +4,6 @@
"""
import os
import cnlunar
from typing import Dict, Any
from config.logger import setup_logging
from jinja2 import Template
@@ -60,6 +59,11 @@ class PromptManager:
self.cache_manager = cache_manager
self.CacheType = CacheType
# 初始化上下文源
from core.utils.context_provider import ContextDataProvider
self.context_provider = ContextDataProvider(config, self.logger)
self.context_data = {}
self._load_base_template()
@@ -180,10 +184,33 @@ class PromptManager:
def update_context_info(self, conn, client_ip: str):
"""同步更新上下文信息"""
try:
# 获取位置信息(使用全局缓存)
local_address = self._get_location_info(client_ip)
# 获取天气信息(使用全局缓存)
self._get_weather_info(conn, local_address)
local_address = ""
if (
client_ip
and self.base_prompt_template
and (
"local_address" in self.base_prompt_template
or "weather_info" in self.base_prompt_template
)
):
# 获取位置信息(使用全局缓存)
local_address = self._get_location_info(client_ip)
if (
self.base_prompt_template
and "weather_info" in self.base_prompt_template
and local_address
):
# 获取天气信息(使用全局缓存)
self._get_weather_info(conn, local_address)
# 获取配置的上下文数据
if hasattr(conn, "device_id") and conn.device_id:
if self.base_prompt_template and "dynamic_context" in self.base_prompt_template:
self.context_data = self.context_provider.fetch_all(conn.device_id)
else:
self.context_data = ""
self.logger.bind(tag=TAG).debug(f"上下文信息更新完成")
except Exception as e:
@@ -230,6 +257,7 @@ class PromptManager:
emojiList=EMOJI_List,
device_id=device_id,
client_ip=client_ip,
dynamic_context=self.context_data,
*args,
**kwargs,
)
+85 -51
View File
@@ -4,6 +4,7 @@ import json
import copy
import wave
import socket
import asyncio
import requests
import subprocess
import numpy as np
@@ -268,56 +269,82 @@ def audio_to_data_stream(
pcm_to_data_stream(raw_data, is_opus, callback)
def audio_to_data(audio_file_path: str, is_opus: bool = True) -> list[bytes]:
async def audio_to_data(
audio_file_path: str, is_opus: bool = True, use_cache: bool = True
) -> list[bytes]:
"""
将音频文件转换为Opus/PCM编码的帧列表
Args:
audio_file_path: 音频文件路径
is_opus: 是否进行Opus编码
use_cache: 是否使用缓存
"""
# 获取文件后缀名
file_type = os.path.splitext(audio_file_path)[1]
if file_type:
file_type = file_type.lstrip(".")
# 读取音频文件,-nostdin 参数:不要从标准输入读取数据,否则FFmpeg会阻塞
audio = AudioSegment.from_file(
audio_file_path, format=file_type, parameters=["-nostdin"]
)
from core.utils.cache.manager import cache_manager
from core.utils.cache.config import CacheType
# 转换为单声道/16kHz采样率/16位小端编码(确保与编码器匹配)
audio = audio.set_channels(1).set_frame_rate(16000).set_sample_width(2)
# 生成缓存键,包含文件路径和编码类型
cache_key = f"{audio_file_path}:{is_opus}"
# 获取原始PCM数据(16位小端)
raw_data = audio.raw_data
# 尝试从缓存获取结果
if use_cache:
cached_result = cache_manager.get(CacheType.AUDIO_DATA, cache_key)
if cached_result is not None:
return cached_result
# 初始化Opus编码器
encoder = opuslib_next.Encoder(16000, 1, opuslib_next.APPLICATION_AUDIO)
def _sync_audio_to_data():
# 获取文件后缀名
file_type = os.path.splitext(audio_file_path)[1]
if file_type:
file_type = file_type.lstrip(".")
# 读取音频文件,-nostdin 参数:不要从标准输入读取数据,否则FFmpeg会阻塞
audio = AudioSegment.from_file(
audio_file_path, format=file_type, parameters=["-nostdin"]
)
# 编码参数
frame_duration = 60 # 60ms per frame
frame_size = int(16000 * frame_duration / 1000) # 960 samples/frame
# 转换为单声道/16kHz采样率/16位小端编码(确保与编码器匹配)
audio = audio.set_channels(1).set_frame_rate(16000).set_sample_width(2)
datas = []
# 按帧处理所有音频数据(包括最后一帧可能补零)
for i in range(0, len(raw_data), frame_size * 2): # 16bit=2bytes/sample
# 获取当前帧的二进制数据
chunk = raw_data[i : i + frame_size * 2]
# 获取原始PCM数据(16位小端)
raw_data = audio.raw_data
# 如果最后一帧不足,补零
if len(chunk) < frame_size * 2:
chunk += b"\x00" * (frame_size * 2 - len(chunk))
# 初始化Opus编码器
encoder = opuslib_next.Encoder(16000, 1, opuslib_next.APPLICATION_AUDIO)
if is_opus:
# 转换为numpy数组处理
np_frame = np.frombuffer(chunk, dtype=np.int16)
# 编码Opus数据
frame_data = encoder.encode(np_frame.tobytes(), frame_size)
else:
frame_data = chunk if isinstance(chunk, bytes) else bytes(chunk)
# 编码参数
frame_duration = 60 # 60ms per frame
frame_size = int(16000 * frame_duration / 1000) # 960 samples/frame
datas.append(frame_data)
datas = []
# 按帧处理所有音频数据(包括最后一帧可能补零)
for i in range(0, len(raw_data), frame_size * 2): # 16bit=2bytes/sample
# 获取当前帧的二进制数据
chunk = raw_data[i : i + frame_size * 2]
return datas
# 如果最后一帧不足,补零
if len(chunk) < frame_size * 2:
chunk += b"\x00" * (frame_size * 2 - len(chunk))
if is_opus:
# 转换为numpy数组处理
np_frame = np.frombuffer(chunk, dtype=np.int16)
# 编码Opus数据
frame_data = encoder.encode(np_frame.tobytes(), frame_size)
else:
frame_data = chunk if isinstance(chunk, bytes) else bytes(chunk)
datas.append(frame_data)
return datas
loop = asyncio.get_running_loop()
# 在单独的线程中执行同步的音频处理操作
result = await loop.run_in_executor(None, _sync_audio_to_data)
# 将结果存入缓存,使用配置中定义的TTL(10分钟)
if use_cache:
cache_manager.set(CacheType.AUDIO_DATA, cache_key, result)
return result
def audio_bytes_to_data_stream(
@@ -372,26 +399,33 @@ def opus_datas_to_wav_bytes(opus_datas, sample_rate=16000, channels=1):
将opus帧列表解码为wav字节流
"""
decoder = opuslib_next.Decoder(sample_rate, channels)
pcm_datas = []
try:
pcm_datas = []
frame_duration = 60 # ms
frame_size = int(sample_rate * frame_duration / 1000) # 960
frame_duration = 60 # ms
frame_size = int(sample_rate * frame_duration / 1000) # 960
for opus_frame in opus_datas:
# 解码为PCM(返回bytes,2字节/采样点)
pcm = decoder.decode(opus_frame, frame_size)
pcm_datas.append(pcm)
for opus_frame in opus_datas:
# 解码为PCM(返回bytes,2字节/采样点)
pcm = decoder.decode(opus_frame, frame_size)
pcm_datas.append(pcm)
pcm_bytes = b"".join(pcm_datas)
pcm_bytes = b"".join(pcm_datas)
# 写入wav字节流
wav_buffer = BytesIO()
with wave.open(wav_buffer, "wb") as wf:
wf.setnchannels(channels)
wf.setsampwidth(2) # 16bit
wf.setframerate(sample_rate)
wf.writeframes(pcm_bytes)
return wav_buffer.getvalue()
# 写入wav字节流
wav_buffer = BytesIO()
with wave.open(wav_buffer, "wb") as wf:
wf.setnchannels(channels)
wf.setsampwidth(2) # 16bit
wf.setframerate(sample_rate)
wf.writeframes(pcm_bytes)
return wav_buffer.getvalue()
finally:
if decoder is not None:
try:
del decoder
except Exception:
pass
def check_vad_update(before_config, new_config):
+31 -4
View File
@@ -1,10 +1,37 @@
import asyncio
import json
import logging
import websockets
from config.logger import setup_logging
class SuppressInvalidHandshakeFilter(logging.Filter):
"""过滤掉无效握手错误日志(如HTTPS访问WS端口)"""
def filter(self, record):
msg = record.getMessage()
suppress_keywords = [
"opening handshake failed",
"did not receive a valid HTTP request",
"connection closed while reading HTTP request",
"line without CRLF",
]
return not any(keyword in msg for keyword in suppress_keywords)
def _setup_websockets_logger():
"""配置 websockets 相关的所有 logger,过滤无效握手错误"""
filter_instance = SuppressInvalidHandshakeFilter()
for logger_name in ["websockets", "websockets.server", "websockets.client"]:
logger = logging.getLogger(logger_name)
logger.addFilter(filter_instance)
_setup_websockets_logger()
from core.connection import ConnectionHandler
from config.config_loader import get_config_from_api
from config.config_loader import get_config_from_api_async
from core.auth import AuthManager, AuthenticationError
from core.utils.modules_initialize import initialize_modules
from core.utils.util import check_vad_update, check_asr_update
@@ -133,8 +160,8 @@ class WebSocketServer:
"""
try:
async with self.config_lock:
# 重新获取配置
new_config = get_config_from_api(self.config)
# 重新获取配置(使用异步版本)
new_config = await get_config_from_api_async(self.config)
if new_config is None:
self.logger.bind(tag=TAG).error("获取新配置失败")
return False