merge main

This commit is contained in:
lizhongxiang
2025-03-10 10:51:25 +08:00
319 changed files with 20283 additions and 89644 deletions
@@ -0,0 +1,19 @@
from abc import ABC, abstractmethod
from typing import Optional, Tuple, List
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class ASRProviderBase(ABC):
@abstractmethod
def save_audio_to_file(self, opus_data: List[bytes], session_id: str) -> str:
"""解码Opus数据并保存为WAV文件"""
pass
@abstractmethod
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
"""将语音数据转换为文本"""
pass
@@ -0,0 +1,286 @@
import time
import io
import wave
import os
from typing import Optional, Tuple, List
import uuid
import websockets
import json
import gzip
import opuslib_next
from core.providers.asr.base import ASRProviderBase
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
CLIENT_FULL_REQUEST = 0b0001
CLIENT_AUDIO_ONLY_REQUEST = 0b0010
NO_SEQUENCE = 0b0000
NEG_SEQUENCE = 0b0010
SERVER_FULL_RESPONSE = 0b1001
SERVER_ACK = 0b1011
SERVER_ERROR_RESPONSE = 0b1111
NO_SERIALIZATION = 0b0000
JSON = 0b0001
THRIFT = 0b0011
CUSTOM_TYPE = 0b1111
NO_COMPRESSION = 0b0000
GZIP = 0b0001
CUSTOM_COMPRESSION = 0b1111
def parse_response(res):
"""
protocol_version(4 bits), header_size(4 bits),
message_type(4 bits), message_type_specific_flags(4 bits)
serialization_method(4 bits) message_compression(4 bits)
reserved 8bits) 保留字段
header_extensions 扩展头(大小等于 8 * 4 * (header_size - 1) )
payload 类似与http 请求体
"""
protocol_version = res[0] >> 4
header_size = res[0] & 0x0f
message_type = res[1] >> 4
message_type_specific_flags = res[1] & 0x0f
serialization_method = res[2] >> 4
message_compression = res[2] & 0x0f
reserved = res[3]
header_extensions = res[4:header_size * 4]
payload = res[header_size * 4:]
result = {}
payload_msg = None
payload_size = 0
if message_type == SERVER_FULL_RESPONSE:
payload_size = int.from_bytes(payload[:4], "big", signed=True)
payload_msg = payload[4:]
elif message_type == SERVER_ACK:
seq = int.from_bytes(payload[:4], "big", signed=True)
result['seq'] = seq
if len(payload) >= 8:
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
payload_msg = payload[8:]
elif message_type == SERVER_ERROR_RESPONSE:
code = int.from_bytes(payload[:4], "big", signed=False)
result['code'] = code
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
payload_msg = payload[8:]
if payload_msg is None:
return result
if message_compression == GZIP:
payload_msg = gzip.decompress(payload_msg)
if serialization_method == JSON:
payload_msg = json.loads(str(payload_msg, "utf-8"))
elif serialization_method != NO_SERIALIZATION:
payload_msg = str(payload_msg, "utf-8")
result['payload_msg'] = payload_msg
result['payload_size'] = payload_size
return result
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
self.appid = config.get("appid")
self.cluster = config.get("cluster")
self.access_token = config.get("access_token")
self.output_dir = config.get("output_dir")
self.host = "openspeech.bytedance.com"
self.ws_url = f"wss://{self.host}/api/v2/asr"
self.success_code = 1000
self.seg_duration = 15000
# 确保输出目录存在
os.makedirs(self.output_dir, exist_ok=True)
def save_audio_to_file(self, opus_data: List[bytes], session_id: str) -> str:
"""将Opus音频数据解码并保存为WAV文件"""
file_name = f"asr_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
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:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
@staticmethod
def _generate_header(message_type=CLIENT_FULL_REQUEST, message_type_specific_flags=NO_SEQUENCE) -> bytearray:
"""Generate protocol header."""
header = bytearray()
header_size = 1
header.append((0b0001 << 4) | header_size) # Protocol version
header.append((message_type << 4) | message_type_specific_flags)
header.append((0b0001 << 4) | 0b0001) # JSON serialization & GZIP compression
header.append(0x00) # reserved
return header
def _construct_request(self, reqid) -> dict:
"""Construct the request payload."""
return {
"app": {
"appid": f"{self.appid}",
"cluster": self.cluster,
"token": self.access_token,
},
"user": {
"uid": str(uuid.uuid4()),
},
"request": {
"reqid": reqid,
"show_utterances": False,
"sequence": 1
},
"audio": {
"format": "wav",
"rate": 16000,
"language": "zh-CN",
"bits": 16,
"channel": 1,
"codec": "raw",
},
}
async def _send_request(self, audio_data: List[bytes], segment_size: int) -> Optional[str]:
"""Send request to Volcano ASR service."""
try:
auth_header = {'Authorization': 'Bearer; {}'.format(self.access_token)}
async with websockets.connect(self.ws_url, additional_headers=auth_header) as websocket:
# Prepare request data
request_params = self._construct_request(str(uuid.uuid4()))
print(request_params)
payload_bytes = str.encode(json.dumps(request_params))
payload_bytes = gzip.compress(payload_bytes)
full_client_request = self._generate_header()
full_client_request.extend((len(payload_bytes)).to_bytes(4, 'big')) # payload size(4 bytes)
full_client_request.extend(payload_bytes) # payload
# Send header and metadata
# full_client_request
await websocket.send(full_client_request)
res = await websocket.recv()
result = parse_response(res)
if 'payload_msg' in result and result['payload_msg']['code'] != self.success_code:
logger.bind(tag=TAG).error(f"ASR error: {result}")
return None
for seq, (chunk, last) in enumerate(self.slice_data(audio_data, segment_size), 1):
if last:
audio_only_request = self._generate_header(
message_type=CLIENT_AUDIO_ONLY_REQUEST,
message_type_specific_flags=NEG_SEQUENCE
)
else:
audio_only_request = self._generate_header(
message_type=CLIENT_AUDIO_ONLY_REQUEST
)
payload_bytes = gzip.compress(chunk)
audio_only_request.extend((len(payload_bytes)).to_bytes(4, 'big')) # payload size(4 bytes)
audio_only_request.extend(payload_bytes) # payload
# Send audio data
await websocket.send(audio_only_request)
# Receive response
response = await websocket.recv()
result = parse_response(response)
if 'payload_msg' in result and result['payload_msg']['code'] == self.success_code:
if len(result['payload_msg']['result']) > 0:
return result['payload_msg']['result'][0]["text"]
return None
else:
logger.bind(tag=TAG).error(f"ASR error: {result}")
return None
except Exception as e:
logger.bind(tag=TAG).error(f"ASR request failed: {e}", exc_info=True)
return None
@staticmethod
def decode_opus(opus_data: List[bytes], session_id: str) -> List[bytes]:
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:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
@staticmethod
def read_wav_info(data: io.BytesIO = None) -> (int, int, int, int, int):
with io.BytesIO(data) as _f:
wave_fp = wave.open(_f, 'rb')
nchannels, sampwidth, framerate, nframes = wave_fp.getparams()[:4]
wave_bytes = wave_fp.readframes(nframes)
return nchannels, sampwidth, framerate, nframes, len(wave_bytes)
@staticmethod
def slice_data(data: bytes, chunk_size: int) -> (list, bool):
"""
slice data
:param data: wav data
:param chunk_size: the segment size in one request
:return: segment data, last flag
"""
data_len = len(data)
offset = 0
while offset + chunk_size < data_len:
yield data[offset: offset + chunk_size], False
offset += chunk_size
else:
yield data[offset: data_len], True
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
"""将语音数据转换为文本"""
try:
# 合并所有opus数据包
pcm_data = self.decode_opus(opus_data, session_id)
combined_pcm_data = b''.join(pcm_data)
wav_buffer = io.BytesIO()
with wave.open(wav_buffer, "wb") as wav_file:
wav_file.setnchannels(1) # 设置声道数
wav_file.setsampwidth(2) # 设置采样宽度
wav_file.setframerate(16000) # 设置采样率
wav_file.writeframes(combined_pcm_data) # 写入 PCM 数据
# 获取封装后的 WAV 数据
wav_data = wav_buffer.getvalue()
nchannels, sampwidth, framerate, nframes, wav_len = self.read_wav_info(wav_data)
size_per_sec = nchannels * sampwidth * framerate
segment_size = int(size_per_sec * self.seg_duration / 1000)
# 语音识别
start_time = time.time()
text = await self._send_request(wav_data, segment_size)
if text:
logger.bind(tag=TAG).debug(f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}")
return text, None
return "", None
except Exception as e:
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
return "", None
@@ -0,0 +1,110 @@
import time
import wave
import os
import sys
import io
from config.logger import setup_logging
from typing import Optional, Tuple, List
import uuid
import opuslib_next
from core.providers.asr.base import ASRProviderBase
from funasr import AutoModel
from funasr.utils.postprocess_utils import rich_transcription_postprocess
TAG = __name__
logger = setup_logging()
# 捕获标准输出
class CaptureOutput:
def __enter__(self):
self._output = io.StringIO()
self._original_stdout = sys.stdout
sys.stdout = self._output
def __exit__(self, exc_type, exc_value, traceback):
sys.stdout = self._original_stdout
self.output = self._output.getvalue()
self._output.close()
# 将捕获到的内容通过 logger 输出
if self.output:
logger.bind(tag=TAG).info(self.output.strip())
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
self.model_dir = config.get("model_dir")
self.output_dir = config.get("output_dir") # 修正配置键名
self.delete_audio_file = delete_audio_file
# 确保输出目录存在
os.makedirs(self.output_dir, exist_ok=True)
with CaptureOutput():
self.model = AutoModel(
model=self.model_dir,
vad_kwargs={"max_single_segment_time": 30000},
disable_update=True,
hub="hf"
# device="cuda:0", # 启用GPU加速
)
def save_audio_to_file(self, opus_data: List[bytes], session_id: str) -> str:
"""将Opus音频数据解码并保存为WAV文件"""
file_name = f"asr_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
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:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
"""语音转文本主处理逻辑"""
file_path = None
try:
# 保存音频文件
start_time = time.time()
file_path = self.save_audio_to_file(opus_data, session_id)
logger.bind(tag=TAG).debug(f"音频文件保存耗时: {time.time() - start_time:.3f}s | 路径: {file_path}")
# 语音识别
start_time = time.time()
result = self.model.generate(
input=file_path,
cache={},
language="auto",
use_itn=True,
batch_size_s=60,
)
text = 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}")
return text, file_path
except Exception as e:
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
return "", None
finally:
# 文件清理逻辑
if self.delete_audio_file and file_path and os.path.exists(file_path):
try:
os.remove(file_path)
logger.bind(tag=TAG).debug(f"已删除临时音频文件: {file_path}")
except Exception as e:
logger.bind(tag=TAG).error(f"文件删除失败: {file_path} | 错误: {e}")
@@ -0,0 +1,33 @@
from abc import ABC, abstractmethod
from typing import List, Dict
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class IntentProviderBase(ABC):
def __init__(self, config):
self.config = config
self.intent_options = config.get("intent_options", {
"continue_chat": "继续聊天",
"end_chat": "结束聊天",
"play_music": "播放音乐"
})
def set_llm(self, llm):
self.llm = llm
logger.bind(tag=TAG).debug("Set LLM for intent provider")
@abstractmethod
async def detect_intent(self, dialogue_history: List[Dict]) -> str:
"""
检测用户最后一句话的意图
Args:
dialogue_history: 对话历史记录列表,每条记录包含role和content
Returns:
返回识别出的意图,格式为:
- "继续聊天"
- "结束聊天"
- "播放音乐 歌名""随机播放音乐"
"""
pass
@@ -0,0 +1,61 @@
from typing import List, Dict
from ..base import IntentProviderBase
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class IntentProvider(IntentProviderBase):
def __init__(self, config):
super().__init__(config)
self.llm = None
self.promot = self.get_intent_system_prompt()
def get_intent_system_prompt(self) -> str:
"""
根据配置的意图选项动态生成系统提示词
Returns:
格式化后的系统提示词
"""
intent_list = []
for key, value in self.intent_options.items():
if key == "play_music":
intent_list.append(f"{value} [歌名]")
else:
intent_list.append(value)
prompt = (
"你是一个意图识别助手。你需要根据和用户的对话记录,重点分析用户的最后一句话,判断用户意图属于以下哪一类:\n"
f"{', '.join(intent_list)}\n"
"如果是唱歌、听歌、播放音乐,请指定歌名,格式为'播放音乐 [识别出的歌名]'\n"
"如果听不出具体歌名,可以返回'随机播放音乐'\n"
"只需要返回意图结果的json,不要解释。"
"返回格式如下:\n"
"{intent: '用户意图'}"
)
return prompt
async def detect_intent(self, dialogue_history: List[Dict]) -> str:
if not self.llm:
raise ValueError("LLM provider not set")
# 构建用户最后一句话的提示
msgStr = ""
for msg in dialogue_history:
if msg.role == "user":
msgStr += f"User: {msg.content}\n"
elif msg.role== "assistant":
msgStr += f"Assistant: {msg.content}\n"
user_prompt = f"请分析用户的意图:\n{msgStr}"
# 使用LLM进行意图识别
intent = self.llm.response_no_stream(
system_prompt=self.promot,
user_prompt=user_prompt
)
logger.bind(tag=TAG).info(f"Detected intent: {intent}")
return intent.strip()
@@ -0,0 +1,18 @@
from ..base import IntentProviderBase
from typing import List, Dict
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class IntentProvider(IntentProviderBase):
async def detect_intent(self, dialogue_history: List[Dict]) -> str:
"""
默认的意图识别实现,始终返回继续聊天
Args:
dialogue_history: 对话历史记录列表
Returns:
固定返回"继续聊天"
"""
logger.bind(tag=TAG).debug("Using NoIntentProvider, always returning continue chat")
return self.intent_options["continue_chat"]
@@ -0,0 +1,38 @@
from abc import ABC, abstractmethod
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class LLMProviderBase(ABC):
@abstractmethod
def response(self, session_id, dialogue):
"""LLM response generator"""
pass
def response_no_stream(self, system_prompt, user_prompt):
try:
# 构造对话格式
dialogue = [
{"role": "system", "content": system_prompt},
{"role": "user", "content": user_prompt}
]
result = ""
for part in self.response("", dialogue):
result += part
return result
except Exception as e:
logger.bind(tag=TAG).error(f"Error in Ollama response generation: {e}")
return "【LLM服务响应异常】"
def response_with_functions(self, session_id, dialogue, functions=None):
"""
Default implementation for function calling (streaming)
This should be overridden by providers that support function calls
Returns: generator that yields either text tokens or a special function call token
"""
# For providers that don't support functions, just return regular response
for token in self.response(session_id, dialogue):
yield {"type": "content", "content": token}
@@ -0,0 +1,37 @@
from config.logger import setup_logging
import requests
import json
import re
from core.providers.llm.base import LLMProviderBase
import os
# official coze sdk for Python [cozepy](https://github.com/coze-dev/coze-py)
from cozepy import COZE_CN_BASE_URL
from cozepy import Coze, TokenAuth, Message, ChatStatus, MessageContentType, ChatEventType # noqa
TAG = __name__
logger = setup_logging()
class LLMProvider(LLMProviderBase):
def __init__(self, config):
self.personal_access_token = config.get("personal_access_token")
self.bot_id = config.get("bot_id")
self.user_id = config.get("user_id")
def response(self, session_id, dialogue):
coze_api_token = self.personal_access_token
coze_api_base = COZE_CN_BASE_URL
last_msg = next(m for m in reversed(dialogue) if m["role"] == "user")
coze = Coze(auth=TokenAuth(token=coze_api_token), base_url=coze_api_base)
for event in coze.chat.stream(
bot_id=self.bot_id,
user_id=self.user_id,
additional_messages=[
Message.build_user_question_text(last_msg["content"]),
],
):
if event.event == ChatEventType.CONVERSATION_MESSAGE_DELTA:
print(event.message.content, end="", flush=True)
yield event.message.content
@@ -0,0 +1,39 @@
import json
from config.logger import setup_logging
import requests
from core.providers.llm.base import LLMProviderBase
TAG = __name__
logger = setup_logging()
class LLMProvider(LLMProviderBase):
def __init__(self, config):
self.api_key = config["api_key"]
self.base_url = config.get("base_url", "https://api.dify.ai/v1").rstrip('/')
def response(self, session_id, dialogue):
try:
# 取最后一条用户消息
last_msg = next(m for m in reversed(dialogue) if m["role"] == "user")
# 发起流式请求
with requests.post(
f"{self.base_url}/chat-messages",
headers={"Authorization": f"Bearer {self.api_key}"},
json={
"query": last_msg["content"],
"response_mode": "streaming",
"user": session_id,
"inputs": {}
},
stream=True
) as r:
for line in r.iter_lines():
if line.startswith(b'data: '):
event = json.loads(line[6:])
if event.get('answer'):
yield event['answer']
except Exception as e:
logger.bind(tag=TAG).error(f"Error in response generation: {e}")
yield "【服务响应异常】"
@@ -0,0 +1,65 @@
import json
from config.logger import setup_logging
import requests
from core.providers.llm.base import LLMProviderBase
TAG = __name__
logger = setup_logging()
class LLMProvider(LLMProviderBase):
def __init__(self, config):
self.api_key = config["api_key"]
self.base_url = config.get("base_url")
self.detail = config.get("detail", False)
self.variables = config.get("variables", {})
def response(self, session_id, dialogue):
try:
# 取最后一条用户消息
last_msg = next(m for m in reversed(dialogue) if m["role"] == "user")
# 发起流式请求
with requests.post(
f"{self.base_url}/chat/completions",
headers={"Authorization": f"Bearer {self.api_key}"},
json={
"stream": True,
"chatId": session_id,
"detail": self.detail,
"variables": self.variables,
"messages": [
{
"role": "user",
"content": last_msg["content"]
}
]
},
stream=True
) as r:
for line in r.iter_lines():
if line:
try:
if line.startswith(b'data: '):
if line[6:].decode('utf-8') == '[DONE]':
break
data = json.loads(line[6:])
if 'choices' in data and len(data['choices']) > 0:
delta = data['choices'][0].get('delta', {})
if delta and 'content' in delta and delta['content'] is not None:
content = delta['content']
if '<think>' in content:
continue
if '</think>' in content:
continue
yield content
except json.JSONDecodeError as e:
continue
except Exception as e:
continue
except Exception as e:
logger.bind(tag=TAG).error(f"Error in response generation: {e}")
yield "【服务响应异常】"
@@ -0,0 +1,80 @@
import google.generativeai as genai
from core.utils.util import check_model_key
from core.providers.llm.base import LLMProviderBase
class LLMProvider(LLMProviderBase):
def __init__(self, config):
"""初始化Gemini LLM Provider"""
self.model_name = config.get("model_name", "gemini-1.5-pro")
self.api_key = config.get("api_key")
have_key = check_model_key("LLM", self.api_key)
if not have_key:
return
try:
# 初始化Gemini客户端
genai.configure(api_key=self.api_key)
self.model = genai.GenerativeModel(self.model_name)
# 设置生成参数
self.generation_config = {
"temperature": 0.7,
"top_p": 0.9,
"top_k": 40,
"max_output_tokens": 2048,
}
self.chat = None
except Exception as e:
logger.bind(tag=TAG).error(f"Gemini初始化失败: {e}")
self.model = None
def response(self, session_id, dialogue):
"""生成Gemini对话响应"""
if not self.model:
yield "【Gemini服务未正确初始化】"
return
try:
# 处理对话历史
chat_history = []
for msg in dialogue[:-1]: # 历史对话
role = "model" if msg["role"] == "assistant" else "user"
content = msg["content"].strip()
if content:
chat_history.append({
"role": role,
"parts": [content]
})
# 获取当前消息
current_msg = dialogue[-1]["content"]
# 创建新的聊天会话
chat = self.model.start_chat(history=chat_history)
# 发送消息并获取流式响应
response = chat.send_message(
current_msg,
stream=True,
generation_config=self.generation_config
)
# 处理流式响应
for chunk in response:
if hasattr(chunk, 'text') and chunk.text:
yield chunk.text
except Exception as e:
error_msg = str(e)
logger.bind(tag=TAG).error(f"Gemini响应生成错误: {error_msg}")
# 针对不同错误返回友好提示
if "Rate limit" in error_msg:
yield "【Gemini服务请求太频繁,请稍后再试】"
elif "Invalid API key" in error_msg:
yield "【Gemini API key无效】"
else:
yield f"【Gemini服务响应异常: {error_msg}"
@@ -0,0 +1,62 @@
import requests
from requests.exceptions import RequestException
from config.logger import setup_logging
from core.providers.llm.base import LLMProviderBase
TAG = __name__
logger = setup_logging()
class LLMProvider(LLMProviderBase):
def __init__(self, config):
self.agent_id = config.get("agent_id") # 对应 agent_id
self.api_key = config.get("api_key")
self.base_url = config.get("base_url", config.get("url")) # 默认使用 base_url
self.api_url = f"{self.base_url}/api/conversation/process" # 拼接完整的 API URL
def response(self, session_id, dialogue):
print(dialogue)
try:
# home assistant语音助手自带意图,无需使用xiaozhi ai自带的,只需要把用户说的话传递给home assistant即可
# 提取最后一个 role 为 'user' 的 content
input_text = None
if isinstance(dialogue, list): # 确保 dialogue 是一个列表
# 逆序遍历,找到最后一个 role 为 'user' 的消息
for message in reversed(dialogue):
if message.get("role") == "user": # 找到 role 为 'user' 的消息
input_text = message.get("content", "")
break # 找到后立即退出循环
# 构造请求数据
payload = {
"text": input_text,
"agent_id": self.agent_id,
"conversation_id": session_id # 使用 session_id 作为 conversation_id
}
# 设置请求头
headers = {
"Authorization": f"Bearer {self.api_key}",
"Content-Type": "application/json"
}
# 发起 POST 请求
response = requests.post(self.api_url, json=payload, headers=headers)
# 检查请求是否成功
response.raise_for_status()
# 解析返回数据
data = response.json()
speech = data.get("response", {}).get("speech", {}).get("plain", {}).get("speech", "")
# 返回生成的内容
if speech:
yield speech
else:
logger.bind(tag=TAG).warning("API 返回数据中没有 speech 内容")
except RequestException as e:
logger.bind(tag=TAG).error(f"HTTP 请求错误: {e}")
except Exception as e:
logger.bind(tag=TAG).error(f"生成响应时出错: {e}")
@@ -0,0 +1,81 @@
from config.logger import setup_logging
from openai import OpenAI
import json
from core.providers.llm.base import LLMProviderBase
TAG = __name__
logger = setup_logging()
class LLMProvider(LLMProviderBase):
def __init__(self, config):
self.model_name = config.get("model_name")
self.base_url = config.get("base_url", "http://localhost:11434")
# Initialize OpenAI client with Ollama base URL
#如果没有v1,增加v1
if not self.base_url.endswith("/v1"):
self.base_url = f"{self.base_url}/v1"
self.client = OpenAI(
base_url=self.base_url,
api_key="ollama" # Ollama doesn't need an API key but OpenAI client requires one
)
def response(self, session_id, dialogue):
try:
responses = self.client.chat.completions.create(
model=self.model_name,
messages=dialogue,
stream=True
)
for chunk in responses:
try:
delta = chunk.choices[0].delta if getattr(chunk, 'choices', None) else None
content = delta.content if hasattr(delta, 'content') else ''
if content:
yield content
except Exception as e:
logger.bind(tag=TAG).error(f"Error processing chunk: {e}")
except Exception as e:
logger.bind(tag=TAG).error(f"Error in Ollama response generation: {e}")
yield "【Ollama服务响应异常】"
def response_with_functions(self, session_id, dialogue, functions=None):
try:
stream = self.client.chat.completions.create(
model=self.model_name,
messages=dialogue,
stream=True,
tools=functions,
)
current_function_call = None
current_content = ""
for chunk in stream:
delta = chunk.choices[0].delta
if delta.content:
current_content += delta.content
yield {"type": "content", "content": delta.content}
if delta.tool_calls:
tool_call = delta.tool_calls[0]
# Handle the function call data using proper attribute access
if not current_function_call:
current_function_call = {
"function": {
"name": tool_call.function.name,
"arguments": tool_call.function.arguments
}
}
if current_function_call:
logger.bind(tag=TAG).debug(f"ollama Function call detected: {current_function_call}")
yield {"type": "function_call", "function_call": current_function_call}
except Exception as e:
logger.bind(tag=TAG).error(f"Error in Ollama function call: {e}")
yield {"type": "content", "content": f"【Ollama服务响应异常: {str(e)}"}
@@ -0,0 +1,83 @@
import openai
from core.utils.util import check_model_key
from core.providers.llm.base import LLMProviderBase
class LLMProvider(LLMProviderBase):
def __init__(self, config):
self.model_name = config.get("model_name")
self.api_key = config.get("api_key")
if 'base_url' in config:
self.base_url = config.get("base_url")
else:
self.base_url = config.get("url")
check_model_key("LLM", self.api_key)
self.client = openai.OpenAI(api_key=self.api_key, base_url=self.base_url)
def response(self, session_id, dialogue):
try:
responses = self.client.chat.completions.create(
model=self.model_name,
messages=dialogue,
stream=True
)
is_active = True
for chunk in responses:
try:
# 检查是否存在有效的choice且content不为空
delta = chunk.choices[0].delta if getattr(chunk, 'choices', None) else None
content = delta.content if hasattr(delta, 'content') else ''
except IndexError:
content = ''
if content:
# 处理标签跨多个chunk的情况
if '<think>' in content:
is_active = False
content = content.split('<think>')[0]
if '</think>' in content:
is_active = True
content = content.split('</think>')[-1]
if is_active:
yield content
except Exception as e:
logger.bind(tag=TAG).error(f"Error in response generation: {e}")
def response_with_functions(self, session_id, dialogue, functions=None):
try:
stream = self.client.chat.completions.create(
model=self.model_name,
messages=dialogue,
stream=True,
tools=functions,
)
current_function_call = None
current_content = ""
for chunk in stream:
delta = chunk.choices[0].delta
if delta.content:
current_content += delta.content
yield {"type": "content", "content": delta.content}
if delta.tool_calls:
tool_call = delta.tool_calls[0]
# Handle the function call data using proper attribute access
if not current_function_call:
current_function_call = {
"function": {
"name": tool_call.function.name,
"arguments": tool_call.function.arguments
}
}
if current_function_call:
logger.bind(tag=TAG).debug(f"openai Function call detected: {current_function_call}")
yield {"type": "function_call", "function_call": current_function_call}
except Exception as e:
self.logger.bind(tag=TAG).error(f"Error in function call streaming: {e}")
yield {"type": "content", "content": f"【OpenAI服务响应异常: {e}"}
@@ -0,0 +1,25 @@
from abc import ABC, abstractmethod
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class MemoryProviderBase(ABC):
def __init__(self, config):
self.config = config
self.role_id = None
self.llm = None
@abstractmethod
async def save_memory(self, msgs):
"""Save a new memory for specific role and return memory ID"""
print("this is base func", msgs)
@abstractmethod
async def query_memory(self, query: str) -> str:
"""Query memories for specific role based on similarity"""
return "please implement query method"
def init_memory(self, role_id, llm):
self.role_id = role_id
self.llm = llm
@@ -0,0 +1,83 @@
import traceback
from ..base import MemoryProviderBase, logger
from mem0 import MemoryClient
from core.utils.util import check_model_key
TAG = __name__
class MemoryProvider(MemoryProviderBase):
def __init__(self, config):
super().__init__(config)
self.api_key = config.get("api_key", "")
self.api_version = config.get("api_version", "v1.1")
have_key = check_model_key("Mem0ai", self.api_key)
if not have_key :
self.use_mem0 = False
return
else:
self.use_mem0 = True
try:
self.client = MemoryClient(api_key=self.api_key)
logger.bind(tag=TAG).info("成功连接到 Mem0ai 服务")
except Exception as e:
logger.bind(tag=TAG).error(f"连接到 Mem0ai 服务时发生错误: {str(e)}")
logger.bind(tag=TAG).error(f"详细错误: {traceback.format_exc()}")
self.use_mem0 = False
async def save_memory(self, msgs):
if not self.use_mem0:
return None
if len(msgs) < 2:
return None
try:
# Format the content as a message list for mem0
messages = [
{"role": message.role, "content": message.content}
for message in msgs if message.role != "system"
]
result = self.client.add(messages, user_id=self.role_id, output_format=self.api_version)
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)}")
return None
async def query_memory(self, query: str)-> str:
if not self.use_mem0:
return ""
try:
results = self.client.search(
query,
user_id=self.role_id,
output_format=self.api_version
)
if not results or 'results' not in results:
return ""
# Format each memory entry with its update time up to minutes
memories = []
for entry in results['results']:
timestamp = entry.get('updated_at', '')
if timestamp:
try:
# Parse and reformat the timestamp
dt = timestamp.split('.')[0] # Remove milliseconds
formatted_time = dt.replace('T', ' ')
except:
formatted_time = timestamp
memory = entry.get('memory', '')
if timestamp and memory:
# Store tuple of (timestamp, formatted_string) for sorting
memories.append((timestamp, f"[{formatted_time}] {memory}"))
# Sort by timestamp in descending order (newest first)
memories.sort(key=lambda x: x[0], reverse=True)
# Extract only the formatted strings
memories_str = "\n".join(f"- {memory[1]}" for memory in memories)
logger.bind(tag=TAG).debug(f"Query results: {memories_str}")
return memories_str
except Exception as e:
logger.bind(tag=TAG).error(f"查询记忆失败: {str(e)}")
return ""
@@ -0,0 +1,156 @@
from ..base import MemoryProviderBase, logger
import time
import json
import os
import yaml
from core.utils.util import get_project_dir
short_term_memory_prompt = """
# 时空记忆编织者
## 核心使命
构建可生长的动态记忆网络,在有限空间内保留关键信息的同时,智能维护信息演变轨迹
根据对话记录,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
## 记忆法则
### 1. 三维度记忆评估(每次更新必执行)
| 维度 | 评估标准 | 权重分 |
|------------|---------------------------|--------|
| 时效性 | 信息新鲜度(按对话轮次) | 40% |
| 情感强度 | 含💖标记/重复提及次数 | 35% |
| 关联密度 | 与其他信息的连接数量 | 25% |
### 2. 动态更新机制
**名字变更处理示例:**
原始记忆:"曾用名": ["张三"], "现用名": "张三丰"
触发条件:当检测到「我叫X」「称呼我Y」等命名信号时
操作流程:
1. 将旧名移入"曾用名"列表
2. 记录命名时间轴:"2024-02-15 14:32:启用张三丰"
3. 在记忆立方追加:「从张三到张三丰的身份蜕变」
### 3. 空间优化策略
- **信息压缩术**:用符号体系提升密度
- ✅"张三丰[北/软工/🐱]"
- ❌"北京软件工程师,养猫"
- **淘汰预警**:当总字数≥900时触发
1. 删除权重分<60且3轮未提及的信息
2. 合并相似条目(保留时间戳最近的)
## 记忆结构
输出格式必须为可解析的json字符串,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
```json
{
"时空档案": {
"身份图谱": {
"现用名": "",
"特征标记": []
},
"记忆立方": [
{
"事件": "入职新公司",
"时间戳": "2024-03-20",
"情感值": 0.9,
"关联项": ["下午茶"],
"保鲜期": 30
}
]
},
"关系网络": {
"高频话题": {"职场": 12},
"暗线联系": [""]
},
"待响应": {
"紧急事项": ["需立即处理的任务"],
"潜在关怀": ["可主动提供的帮助"]
},
"高光语录": [
"最打动人心的瞬间,强烈的情感表达,user的原话"
]
}
```
"""
def extract_json_data(json_code):
start = json_code.find("```json")
# 从start开始找到下一个```结束
end = json_code.find("```", start+1)
#print("start:", start, "end:", end)
if start == -1 or end == -1:
try:
jsonData = json.loads(json_code)
return json_code
except Exception as e:
print("Error:", e)
return ""
jsonData = json_code[start+7:end]
return jsonData
TAG = __name__
class MemoryProvider(MemoryProviderBase):
def __init__(self, config):
super().__init__(config)
self.short_momery = ""
self.memory_path = get_project_dir() + 'data/.memory.yaml'
self.load_memory()
def init_memory(self, role_id, llm):
super().init_memory(role_id, llm)
self.load_memory()
def load_memory(self):
all_memory = {}
if os.path.exists(self.memory_path):
with open(self.memory_path, 'r', encoding='utf-8') as f:
all_memory = yaml.safe_load(f) or {}
if self.role_id in all_memory:
self.short_momery = all_memory[self.role_id]
def save_memory_to_file(self):
all_memory = {}
if os.path.exists(self.memory_path):
with open(self.memory_path, 'r', encoding='utf-8') as f:
all_memory = yaml.safe_load(f) or {}
all_memory[self.role_id] = self.short_momery
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):
if self.llm is None:
logger.bind(tag=TAG).error("LLM is not set for memory provider")
return None
if len(msgs) < 2:
return None
msgStr = ""
for msg in msgs:
if msg.role == "user":
msgStr += f"User: {msg.content}\n"
elif msg.role== "assistant":
msgStr += f"Assistant: {msg.content}\n"
if len(self.short_momery) > 0:
msgStr+="历史记忆:\n"
msgStr+=self.short_momery
#当前时间
time_str = time.strftime("%Y-%m-%d %H:%M:%S", time.localtime())
msgStr += f"当前时间:{time_str}"
result = self.llm.response_no_stream(short_term_memory_prompt, msgStr)
json_str = extract_json_data(result)
try:
json_data = json.loads(json_str) # 检查json格式是否正确
self.short_momery = json_str
except Exception as e:
print("Error:", e)
self.save_memory_to_file()
logger.bind(tag=TAG).info(f"Save memory successful - Role: {self.role_id}")
return self.short_momery
async def query_memory(self, query: str)-> str:
return self.short_momery
@@ -0,0 +1,18 @@
'''
不使用记忆,可以选择此模块
'''
from ..base import MemoryProviderBase, logger
TAG = __name__
class MemoryProvider(MemoryProviderBase):
def __init__(self, config):
super().__init__(config)
async def save_memory(self, msgs):
logger.bind(tag=TAG).debug("nomem mode: No memory saving is performed.")
return None
async def query_memory(self, query: str)-> str:
logger.bind(tag=TAG).debug("nomem mode: No memory query is performed.")
return ""
@@ -0,0 +1,132 @@
import os
import uuid
import json
import hmac
import hashlib
import base64
import requests
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
import http.client
import urllib.parse
import time
import uuid
from urllib import parse
class AccessToken:
@staticmethod
def _encode_text(text):
encoded_text = parse.quote_plus(text)
return encoded_text.replace('+', '%20').replace('*', '%2A').replace('%7E', '~')
@staticmethod
def _encode_dict(dic):
keys = dic.keys()
dic_sorted = [(key, dic[key]) for key in sorted(keys)]
encoded_text = parse.urlencode(dic_sorted)
return encoded_text.replace('+', '%20').replace('*', '%2A').replace('%7E', '~')
@staticmethod
def create_token(access_key_id, access_key_secret):
parameters = {'AccessKeyId': access_key_id,
'Action': 'CreateToken',
'Format': 'JSON',
'RegionId': 'cn-shanghai',
'SignatureMethod': 'HMAC-SHA1',
'SignatureNonce': str(uuid.uuid1()),
'SignatureVersion': '1.0',
'Timestamp': time.strftime("%Y-%m-%dT%H:%M:%SZ", time.gmtime()),
'Version': '2019-02-28'}
# 构造规范化的请求字符串
query_string = AccessToken._encode_dict(parameters)
print('规范化的请求字符串: %s' % query_string)
# 构造待签名字符串
string_to_sign = 'GET' + '&' + AccessToken._encode_text('/') + '&' + AccessToken._encode_text(query_string)
print('待签名的字符串: %s' % string_to_sign)
# 计算签名
secreted_string = hmac.new(bytes(access_key_secret + '&', encoding='utf-8'),
bytes(string_to_sign, encoding='utf-8'),
hashlib.sha1).digest()
signature = base64.b64encode(secreted_string)
print('签名: %s' % signature)
# 进行URL编码
signature = AccessToken._encode_text(signature)
print('URL编码后的签名: %s' % signature)
# 调用服务
full_url = 'http://nls-meta.cn-shanghai.aliyuncs.com/?Signature=%s&%s' % (signature, query_string)
print('url: %s' % full_url)
# 提交HTTP GET请求
response = requests.get(full_url)
if response.ok:
root_obj = response.json()
key = 'Token'
if key in root_obj:
token = root_obj[key]['Id']
expire_time = root_obj[key]['ExpireTime']
return token, expire_time
print(response.text)
return None, None
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
# 新增空值判断逻辑
access_key_id = config.get("access_key_id")
access_key_secret = config.get("access_key_secret")
if access_key_id and access_key_secret:
# 使用密钥对生成临时token
token, expire_time = AccessToken.create_token(access_key_id, access_key_secret)
else:
# 直接使用预生成的长期token
token = config.get("token")
expire_time = None
print('token: %s, expire time(s): %s' % (token, expire_time))
self.appkey = config.get("appkey")
self.token = token
self.format = config.get("format", "wav")
self.sample_rate = config.get("sample_rate", 16000)
self.voice = config.get("voice", "xiaoyun")
self.volume = config.get("volume", 50)
self.speech_rate = config.get("speech_rate", 0)
self.pitch_rate = config.get("pitch_rate", 0)
self.host = config.get("host", "nls-gateway-cn-shanghai.aliyuncs.com")
self.api_url = f"https://{self.host}/stream/v1/tts"
self.header = {
"Content-Type": "application/json"
}
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{__name__}{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
request_json = {
"appkey": self.appkey,
"token": self.token,
"text": text,
"format": self.format,
"sample_rate": self.sample_rate,
"voice": self.voice,
"volume": self.volume,
"speech_rate": self.speech_rate,
"pitch_rate": self.pitch_rate
}
print(self.api_url, json.dumps(request_json, ensure_ascii=False))
try:
resp = requests.post(self.api_url, json.dumps(request_json), headers=self.header)
# 检查返回请求数据的mime类型是否是audio/***,是则保存到指定路径下;返回的是binary格式的
if resp.headers['Content-Type'].startswith('audio/'):
with open(output_file, 'wb') as f:
f.write(resp.content)
return output_file
else:
raise Exception(f"{__name__} status_code: {resp.status_code} response: {resp.content}")
except Exception as e:
raise Exception(f"{__name__} error: {e}")
@@ -0,0 +1,123 @@
import asyncio
from config.logger import setup_logging
import os
import numpy as np
import opuslib_next
from pydub import AudioSegment
from abc import ABC, abstractmethod
import queue
TAG = __name__
logger = setup_logging()
class TTSProviderBase(ABC):
def __init__(self, config, delete_audio_file):
self.delete_audio_file = delete_audio_file
self.output_file = config.get("output_file")
@abstractmethod
def generate_filename(self):
pass
def to_tts(self, text):
tmp_file = self.generate_filename()
try:
max_repeat_time = 5
while not os.path.exists(tmp_file) and max_repeat_time > 0:
asyncio.run(self.text_to_speak(text, tmp_file))
if not os.path.exists(tmp_file):
max_repeat_time = max_repeat_time - 1
logger.bind(tag=TAG).error(f"语音生成失败: {text}:{tmp_file},再试{max_repeat_time}")
if max_repeat_time > 0:
logger.bind(tag=TAG).info(f"语音生成成功: {text}:{tmp_file},重试{5 - max_repeat_time}")
return tmp_file
except Exception as e:
logger.bind(tag=TAG).info(f": {e}")
return None
def to_tts_stream(self, text, queue: queue.Queue, text_index=0):
try:
asyncio.run(self.text_to_speak_stream(text, queue, text_index))
except Exception as e:
logger.bind(tag=TAG).info(f"Failed to generate TTS file: {e}")
return None
@abstractmethod
async def text_to_speak(self, text, output_file):
pass
async def text_to_speak_stream(self, text, queue: queue.Queue, text_index=0):
raise Exception("该TTS还没有实现stream模式")
def wav_to_opus_data(self, wav_file_path):
# 使用pydub加载PCM文件
# 获取文件后缀名
file_type = os.path.splitext(wav_file_path)[1]
if file_type:
file_type = file_type.lstrip('.')
audio = AudioSegment.from_file(wav_file_path, format=file_type)
duration = len(audio) / 1000.0
# 转换为单声道和16kHz采样率(确保与编码器匹配)
audio = audio.set_channels(1).set_frame_rate(16000)
# 获取原始PCM数据(16位小端)
raw_data = audio.raw_data
# 初始化Opus编码器
encoder = opuslib_next.Encoder(16000, 1, opuslib_next.APPLICATION_AUDIO)
# 编码参数
frame_duration = 60 # 60ms per frame
frame_size = int(16000 * frame_duration / 1000) # 960 samples/frame
opus_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]
# 如果最后一帧不足,补零
if len(chunk) < frame_size * 2:
chunk += b'\x00' * (frame_size * 2 - len(chunk))
# 转换为numpy数组处理
np_frame = np.frombuffer(chunk, dtype=np.int16)
# 编码Opus数据
opus_data = encoder.encode(np_frame.tobytes(), frame_size)
opus_datas.append(opus_data)
return opus_datas, duration
def wav_to_opus_data_audio_raw(self, raw_data):
# 初始化Opus编码器
encoder = opuslib_next.Encoder(16000, 1, opuslib_next.APPLICATION_AUDIO)
# 编码参数
frame_duration = 60 # 60ms per frame
frame_size = int(16000 * frame_duration / 1000) # 960 samples/frame
opus_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]
# 如果最后一帧不足,补零
if len(chunk) < frame_size * 2:
# logger.bind(tag=TAG).info("开始补0")
chunk += b'\x00' * (frame_size * 2 - len(chunk))
# 转换为numpy数组处理
np_frame = np.frombuffer(chunk, dtype=np.int16)
# 编码Opus数据
opus_data = encoder.encode(np_frame.tobytes(), frame_size)
opus_datas.append(opus_data)
return opus_datas
@@ -0,0 +1,38 @@
import os
import uuid
import json
import base64
import requests
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.model = config.get("model")
self.access_token = config.get("access_token")
self.voice = config.get("voice")
self.response_format = config.get("response_format")
self.host = "api.coze.cn"
self.api_url = f"https://{self.host}/v1/audio/speech"
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
request_json = {
"model": self.model,
"input": text,
"voice_id": self.voice,
"response_format": self.response_format,
}
headers = {
"Authorization": f"Bearer {self.access_token}",
"Content-Type": "application/json"
}
response = requests.request("POST", self.api_url, json=request_json, headers=headers)
data = response.content
file_to_save = open(output_file, "wb")
file_to_save.write(data)
@@ -0,0 +1,62 @@
import os
import uuid
import json
import base64
import requests
from datetime import datetime
from core.utils.util import check_model_key
from core.providers.tts.base import TTSProviderBase
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.appid = config.get("appid")
self.access_token = config.get("access_token")
self.cluster = config.get("cluster")
self.voice = config.get("voice")
self.api_url = config.get("api_url")
self.authorization = config.get("authorization")
self.header = {"Authorization": f"{self.authorization}{self.access_token}"}
check_model_key("TTS", self.access_token)
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
request_json = {
"app": {
"appid": f"{self.appid}",
"token": "access_token",
"cluster": self.cluster
},
"user": {
"uid": "1"
},
"audio": {
"voice_type": self.voice,
"encoding": "wav",
"speed_ratio": 1.0,
"volume_ratio": 1.0,
"pitch_ratio": 1.0,
},
"request": {
"reqid": str(uuid.uuid4()),
"text": text,
"text_type": "plain",
"operation": "query",
"with_frontend": 1,
"frontend_type": "unitTson"
}
}
try:
resp = requests.post(self.api_url, json.dumps(request_json), headers=self.header)
if "data" in resp.json():
data = resp.json()["data"]
file_to_save = open(output_file, "wb")
file_to_save.write(base64.b64decode(data))
else:
raise Exception(f"{__name__} status_code: {resp.status_code} response: {resp.content}")
except Exception as e:
raise Exception(f"{__name__} error: {e}")
@@ -0,0 +1,18 @@
import os
import uuid
import edge_tts
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.voice = config.get("voice")
def generate_filename(self, extension=".mp3"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
communicate = edge_tts.Communicate(text, voice=self.voice) # Use your preferred voice
await communicate.save(output_file)
@@ -0,0 +1,268 @@
import base64
import os
import traceback
import uuid
import queue
import io
import numpy as np
import requests
import ormsgpack
from pathlib import Path
import torch
import torchaudio
from pydantic import BaseModel, Field, conint, model_validator
from pydub import AudioSegment
from typing_extensions import Annotated
from datetime import datetime
from typing import Literal
from core.utils.util import check_model_key
from core.providers.tts.base import TTSProviderBase
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class ServeReferenceAudio(BaseModel):
audio: bytes
text: str
@model_validator(mode="before")
def decode_audio(cls, values):
audio = values.get("audio")
if (
isinstance(audio, str) and len(audio) > 255
): # Check if audio is a string (Base64)
try:
values["audio"] = base64.b64decode(audio)
except Exception as e:
# If the audio is not a valid base64 string, we will just ignore it and let the server handle it
pass
return values
def __repr__(self) -> str:
return f"ServeReferenceAudio(text={self.text!r}, audio_size={len(self.audio)})"
class ServeTTSRequest(BaseModel):
text: str
chunk_length: Annotated[int, conint(ge=100, le=300, strict=True)] = 200
# Audio format
format: Literal["wav", "pcm", "mp3"] = "wav"
# References audios for in-context learning
references: list[ServeReferenceAudio] = []
# Reference id
# For example, if you want use https://fish.audio/m/7f92f8afb8ec43bf81429cc1c9199cb1/
# Just pass 7f92f8afb8ec43bf81429cc1c9199cb1
reference_id: str | None = None
seed: int | None = None
use_memory_cache: Literal["on", "off"] = "off"
# Normalize text for en & zh, this increase stability for numbers
normalize: bool = True
# not usually used below
streaming: bool = False
max_new_tokens: int = 1024
top_p: Annotated[float, Field(ge=0.1, le=1.0, strict=True)] = 0.7
repetition_penalty: Annotated[float, Field(ge=0.9, le=2.0, strict=True)] = 1.2
temperature: Annotated[float, Field(ge=0.1, le=1.0, strict=True)] = 0.7
class Config:
# Allow arbitrary types for pytorch related types
arbitrary_types_allowed = True
def audio_to_bytes(file_path):
if not file_path or not Path(file_path).exists():
return None
with open(file_path, "rb") as wav_file:
wav = wav_file.read()
return wav
def read_ref_text(ref_text):
path = Path(ref_text)
if path.exists() and path.is_file():
with path.open("r", encoding="utf-8") as file:
return file.read()
return ref_text
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.reference_id = config.get("reference_id")
self.reference_audio = config.get("reference_audio", [])
self.reference_text = config.get("reference_text", [])
self.format = config.get("format", "wav")
self.channels = config.get("channels", 1)
self.rate = config.get("rate", 44100)
self.api_key = config.get("api_key", "YOUR_API_KEY")
have_key = check_model_key("FishSpeech TTS", self.api_key)
if not have_key:
return
self.normalize = config.get("normalize", True)
self.max_new_tokens = config.get("max_new_tokens", 1024)
self.chunk_length = config.get("chunk_length", 200)
self.top_p = config.get("top_p", 0.7)
self.repetition_penalty = config.get("repetition_penalty", 1.2)
self.temperature = config.get("temperature", 0.7)
self.streaming = config.get("streaming", False)
self.use_memory_cache = config.get("use_memory_cache", "on")
self.seed = config.get("seed")
self.api_url = config.get("api_url", "http://127.0.0.1:8080/v1/tts")
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
# Prepare reference data
byte_audios = [audio_to_bytes(ref_audio) for ref_audio in self.reference_audio]
ref_texts = [read_ref_text(ref_text) for ref_text in self.reference_text]
data = {
"text": text,
"references": [
ServeReferenceAudio(
audio=audio if audio else b"", text=text
)
for text, audio in zip(ref_texts, byte_audios)
],
"reference_id": self.reference_id,
"normalize": self.normalize,
"format": self.format,
"max_new_tokens": self.max_new_tokens,
"chunk_length": self.chunk_length,
"top_p": self.top_p,
"repetition_penalty": self.repetition_penalty,
"temperature": self.temperature,
"streaming": self.streaming,
"use_memory_cache": self.use_memory_cache,
"seed": self.seed,
}
pydantic_data = ServeTTSRequest(**data)
response = requests.post(
self.api_url,
data=ormsgpack.packb(pydantic_data, option=ormsgpack.OPT_SERIALIZE_PYDANTIC),
headers={
"Authorization": f"Bearer {self.api_key}",
"Content-Type": "application/msgpack",
},
)
if response.status_code == 200:
audio_content = response.content
with open(output_file, "wb") as audio_file:
audio_file.write(audio_content)
else:
print(f"Request failed with status code {response.status_code}")
print(response.json())
def _get_audio_from_tts(self, data_bytes):
tts_speech = torch.from_numpy(np.array(np.frombuffer(data_bytes, dtype=np.int16))).unsqueeze(dim=0)
with io.BytesIO() as bf:
torchaudio.save(bf, tts_speech, 44100, format="wav")
audio = AudioSegment.from_file(bf, format="wav")
audio = audio.set_channels(1).set_frame_rate(16000)
return audio
async def text_to_speak_stream(self, text, queue: queue.Queue, text_index=0):
try:
# Prepare reference data
byte_audios = [audio_to_bytes(ref_audio) for ref_audio in self.reference_audio]
ref_texts = [read_ref_text(ref_text) for ref_text in self.reference_text]
data = {
"text": text,
"references": [
ServeReferenceAudio(
audio=audio if audio else b"", text=text
)
for text, audio in zip(ref_texts, byte_audios)
],
"reference_id": self.reference_id,
"normalize": self.normalize,
"format": self.format,
"max_new_tokens": self.max_new_tokens,
"chunk_length": self.chunk_length,
"top_p": self.top_p,
"repetition_penalty": self.repetition_penalty,
"temperature": self.temperature,
"streaming": self.streaming,
"use_memory_cache": self.use_memory_cache,
"seed": self.seed,
}
pydantic_data = ServeTTSRequest(**data)
audio_buff = None
chunk_total = b''
last_raw = b''
audio_raw = b''
print("请求tts")
with requests.post(
self.api_url,
data=ormsgpack.packb(pydantic_data, option=ormsgpack.OPT_SERIALIZE_PYDANTIC),
headers={
"Authorization": f"Bearer {self.api_key}",
"Content-Type": "application/msgpack",
},
) as response:
if response.status_code == 200:
for chunk in response.iter_content():
# 拼接当前块和上一块数据
chunk_total += chunk
# 最后一个是静音,说明是一个完整的音频
if len(chunk_total) % 2 == 0 and chunk_total[-2:] == b'\x00\x00':
audio = self._get_audio_from_tts(chunk_total)
audio_raw = audio_raw + audio.raw_data
#长度凑够2贞开始发送,60ms*4=240ms
if len(audio_raw) >= 7680:
duration = 60 * len(audio_raw) // 1920
if (len(audio_raw) % 1920) > 0:
duration += 60
duration = duration / 1000.0
logger.bind(tag=TAG).info(f'发送数据长度:{len(audio_raw)}')
opus_datas = self.wav_to_opus_data_audio_raw(audio_raw)
queue.put({
"data": opus_datas,
"duration": duration,
"end": False,
"text_index": text_index
})
audio_raw = b''
chunk_total = b''
if len(chunk_total) > 0:
audio = self._get_audio_from_tts(chunk_total)
audio_raw = audio_raw + audio.raw_data
duration = 60 * len(audio_raw) // 1920
if (len(audio_raw) % 1920) > 0:
duration += 60
duration = duration / 1000.0
# 把 audio 转成 opus
logger.bind(tag=TAG).info(f'发送数据长度:{len(audio_raw)}')
opus_datas = self.wav_to_opus_data_audio_raw(audio_raw)
queue.put({
"data": opus_datas,
"duration": duration,
"end": False
})
else:
print('请求失败:', response.status_code, response.text)
queue.put({
"data": None,
"end": True
})
except Exception as e:
logger.bind(tag=TAG).error("tts发生错误")
traceback.print_exc()
raise e
@@ -0,0 +1,67 @@
import os
import uuid
import json
import base64
import requests
from config.logger import setup_logging
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
TAG = __name__
logger = setup_logging()
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.url = config.get("url")
self.text_lang = config.get("text_lang", "zh")
self.ref_audio_path = config.get("ref_audio_path")
self.prompt_text = config.get("prompt_text")
self.prompt_lang = config.get("prompt_lang", "zh")
self.top_k = config.get("top_k", 5)
self.top_p = config.get("top_p", 1)
self.temperature = config.get("temperature", 1)
self.text_split_method = config.get("text_split_method", "cut0")
self.batch_size = config.get("batch_size", 1)
self.batch_threshold = config.get("batch_threshold", 0.75)
self.split_bucket = config.get("split_bucket", True)
self.return_fragment = config.get("return_fragment", False)
self.speed_factor = config.get("speed_factor", 1.0)
self.streaming_mode = config.get("streaming_mode", False)
self.seed = config.get("seed", -1)
self.parallel_infer = config.get("parallel_infer", True)
self.repetition_penalty = config.get("repetition_penalty", 1.35)
self.aux_ref_audio_paths = config.get("aux_ref_audio_paths", [])
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
request_json = {
"text": text,
"text_lang": self.text_lang,
"ref_audio_path": self.ref_audio_path,
"aux_ref_audio_paths": self.aux_ref_audio_paths,
"prompt_text": self.prompt_text,
"prompt_lang": self.prompt_lang,
"top_k": self.top_k,
"top_p": self.top_p,
"temperature": self.temperature,
"text_split_method": self.text_split_method,
"batch_size": self.batch_size,
"batch_threshold": self.batch_threshold,
"split_bucket": self.split_bucket,
"return_fragment": self.return_fragment,
"speed_factor": self.speed_factor,
"streaming_mode": self.streaming_mode,
"seed": self.seed,
"parallel_infer": self.parallel_infer,
"repetition_penalty": self.repetition_penalty
}
resp = requests.post(self.url, json=request_json)
if resp.status_code == 200:
with open(output_file, "wb") as file:
file.write(resp.content)
else:
logger.bind(tag=TAG).error(f"GPT_SoVITS_V2 TTS请求失败: {resp.status_code} - {resp.text}")
@@ -0,0 +1,52 @@
import os
import uuid
import requests
from config.logger import setup_logging
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
TAG = __name__
logger = setup_logging()
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.url = config.get("url")
self.text_lang = config.get("text_lang", "audo")
self.ref_audio_path = config.get("ref_audio_path")
self.prompt_lang = config.get("prompt_lang")
self.prompt_text = config.get("prompt_text")
self.top_k = config.get("top_k", 5)
self.top_p = config.get("top_p", 1)
self.temperature = config.get("temperature", 1)
self.sample_steps = config.get("sample_steps", 16)
self.media_type = config.get("media_type", "wav")
self.streaming_mode = config.get("streaming_mode", False)
self.threshold = config.get("threshold", 30)
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
request_params = {
"text": text,
"text_lang": self.text_lang,
"ref_audio_path": self.ref_audio_path,
"prompt_lang": self.prompt_lang,
"prompt_text": self.prompt_text,
"top_k": self.top_k,
"top_p": self.top_p,
"temperature": self.temperature,
"sample_steps": self.sample_steps,
"media_type": self.media_type,
"streaming_mode": self.streaming_mode,
"threshold": self.threshold,
}
resp = requests.get(self.url, params=request_params)
if resp.status_code == 200:
with open(output_file, "wb") as file:
file.write(resp.content)
else:
logger.bind(tag=TAG).error(f"GPT_SoVITS_V3 TTS请求失败: {resp.status_code} - {resp.text}")
@@ -0,0 +1,77 @@
import os
import uuid
import json
import requests
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.group_id = config.get("group_id")
self.api_key = config.get("api_key")
self.model = config.get("model")
self.voice_id = config.get("voice_id")
default_voice_setting = {
"voice_id": "female-shaonv",
"speed": 1,
"vol": 1,
"pitch": 0,
"emotion": "happy"
}
default_pronunciation_dict = {
"tone": [
"处理/(chu3)(li3)", "危险/dangerous"
]
}
defult_audio_setting = {
"sample_rate": 32000,
"bitrate": 128000,
"format": "mp3",
"channel": 1
}
self.voice_setting = {**default_voice_setting, **config.get("voice_setting", {})}
self.pronunciation_dict = {**default_pronunciation_dict, **config.get("pronunciation_dict", {})}
self.audio_setting = {**defult_audio_setting, **config.get("audio_setting", {})}
self.timber_weights = config.get("timber_weights", [])
if self.voice_id:
self.voice_setting["voice_id"] = self.voice_id
self.host = "api.minimax.chat"
self.api_url = f"https://{self.host}/v1/t2a_v2?GroupId={self.group_id}"
self.header = {
"Content-Type": "application/json",
"Authorization": f"Bearer {self.api_key}"
}
def generate_filename(self, extension=".mp3"):
return os.path.join(self.output_file, f"tts-{__name__}{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
request_json = {
"model": self.model,
"text": text,
"stream": False,
"voice_setting": self.voice_setting,
"pronunciation_dict": self.pronunciation_dict,
"audio_setting": self.audio_setting,
}
if type(self.timber_weights) is list and len(self.timber_weights) > 0:
request_json["timber_weights"] = self.timber_weights
request_json["voice_setting"]["voice_id"] = ""
try:
resp = requests.post(self.api_url, json.dumps(request_json), headers=self.header)
# 检查返回请求数据的status_code是否为0
if resp.json()["base_resp"]["status_code"] == 0:
data = resp.json()['data']['audio']
file_to_save = open(output_file, "wb")
file_to_save.write(bytes.fromhex(data))
else:
raise Exception(f"{__name__} status_code: {resp.status_code} response: {resp.content}")
except Exception as e:
raise Exception(f"{__name__} error: {e}")
@@ -0,0 +1,40 @@
import os
import uuid
import requests
from datetime import datetime
from core.utils.util import check_model_key
from core.providers.tts.base import TTSProviderBase
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.api_key = config.get("api_key")
self.api_url = config.get("api_url", "https://api.openai.com/v1/audio/speech")
self.model = config.get("model", "tts-1")
self.voice = config.get("voice", "alloy")
self.response_format = "wav"
self.speed = config.get("speed", 1.0)
self.output_file = config.get("output_file", "tmp/")
check_model_key("TTS", self.api_key)
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
headers = {
"Authorization": f"Bearer {self.api_key}",
"Content-Type": "application/json"
}
data = {
"model": self.model,
"input": text,
"voice": self.voice,
"response_format": "wav",
"speed": self.speed
}
response = requests.post(self.api_url, json=data, headers=headers)
if response.status_code == 200:
with open(output_file, "wb") as audio_file:
audio_file.write(response.content)
else:
raise Exception(f"OpenAI TTS请求失败: {response.status_code} - {response.text}")
@@ -0,0 +1,39 @@
import os
import uuid
import requests
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.model = config.get("model")
self.access_token = config.get("access_token")
self.voice = config.get("voice")
self.response_format = config.get("response_format")
self.sample_rate = config.get("sample_rate")
self.speed = config.get("speed")
self.gain = config.get("gain")
self.host = "api.siliconflow.cn"
self.api_url = f"https://{self.host}/v1/audio/speech"
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
request_json = {
"model": self.model,
"input": text,
"voice": self.voice,
"response_format": self.response_format,
}
headers = {
"Authorization": f"Bearer {self.access_token}",
"Content-Type": "application/json"
}
response = requests.request("POST", self.api_url, json=request_json, headers=headers)
data = response.content
file_to_save = open(output_file, "wb")
file_to_save.write(data)
@@ -0,0 +1,64 @@
import os
import uuid
import json
import requests
import shutil
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.url = config.get("url", "https://u95167-bd74-2aef8085.westx.seetacloud.com:8443/flashsummary/tts?token=")
self.voice_id = config.get("voice_id", 1695)
self.token = config.get("token")
self.to_lang = config.get("to_lang")
self.volume_change_dB = config.get("volume_change_dB", 0)
self.speed_factor = config.get("speed_factor", 1)
self.stream = config.get("stream", False)
self.output_file = config.get("output_file")
self.pitch_factor = config.get("pitch_factor", 0)
self.format = config.get("format", "mp3")
self.emotion = config.get("emotion", 1)
self.header = {
"Content-Type": "application/json"
}
def generate_filename(self, extension=".mp3"):
return os.path.join(self.output_file, f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file):
url = f'{self.url}{self.token}'
result = "firefly"
payload = json.dumps({
"to_lang": self.to_lang,
"text": text,
"emotion": self.emotion,
"format": self.format,
"volume_change_dB": self.volume_change_dB,
"voice_id": self.voice_id,
"pitch_factor": self.pitch_factor,
"speed_factor": self.speed_factor,
"token": self.token
})
resp = requests.request("POST", url, data=payload)
if resp.status_code != 200:
return None
resp_json = resp.json()
try:
result = resp_json['url'] + ':' + str(
resp_json[
'port']) + '/flashsummary/retrieveFileData?stream=True&token=' + self.token + '&voice_audio_path=' + \
resp_json['voice_path']
except Exception as e:
print("error:", e)
audio_content = requests.get(result)
with open(output_file, "wb") as f:
f.write(audio_content.content)
return True
voice_path = resp_json.get("voice_path")
des_path = output_file
shutil.move(voice_path, des_path)