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重新划分目录 (#214)
* 🦄 refactor(web): 修改zhikongtaiweb到web * 🦄 refactor: 重写前端 路由守护尚未写完 * 🦄 refactor: 标准化路由 * update:前端重写,保留后端 * update:添加前端代码 * update:pip转成poetry启动 * update:增加mem0ai包依赖 * update:文档增加mem0ai的描述 * feat: play online mp3 (#181) Co-authored-by: 欣南科技 <huangrongzhuang@xin-nan.com> * 修改前端代码 * update:调整项目目录 * update:优化 * update:配置文件去除8002端口 * update:增加开发说明 * update:更新开发协议 --------- Co-authored-by: kalicyh <34980061+kaliCYH@users.noreply.github.com> Co-authored-by: hrz <1710360675@qq.com> Co-authored-by: freshlife001 <talent@mises.site> Co-authored-by: CGD <3030332422@qq.com>
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co-authored by
kalicyh
hrz
freshlife001
CGD
parent
8b151d07c2
commit
0e43748fdc
@@ -0,0 +1,84 @@
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import asyncio
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from config.logger import setup_logging
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import os
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import numpy as np
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import opuslib_next
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from pydub import AudioSegment
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from abc import ABC, abstractmethod
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TAG = __name__
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logger = setup_logging()
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class TTSProviderBase(ABC):
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def __init__(self, config, delete_audio_file):
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self.delete_audio_file = delete_audio_file
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self.output_file = config.get("output_file")
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@abstractmethod
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def generate_filename(self):
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pass
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def to_tts(self, text):
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tmp_file = self.generate_filename()
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try:
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max_repeat_time = 5
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while not os.path.exists(tmp_file) and max_repeat_time > 0:
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asyncio.run(self.text_to_speak(text, tmp_file))
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if not os.path.exists(tmp_file):
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max_repeat_time = max_repeat_time - 1
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logger.bind(tag=TAG).error(f"语音生成失败: {text}:{tmp_file},再试{max_repeat_time}次")
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if max_repeat_time > 0:
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logger.bind(tag=TAG).info(f"语音生成成功: {text}:{tmp_file},重试{5 - max_repeat_time}次")
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return tmp_file
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except Exception as e:
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logger.bind(tag=TAG).info(f"Failed to generate TTS file: {e}")
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return None
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@abstractmethod
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async def text_to_speak(self, text, output_file):
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pass
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def wav_to_opus_data(self, wav_file_path):
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# 使用pydub加载PCM文件
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# 获取文件后缀名
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file_type = os.path.splitext(wav_file_path)[1]
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if file_type:
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file_type = file_type.lstrip('.')
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audio = AudioSegment.from_file(wav_file_path, format=file_type)
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duration = len(audio) / 1000.0
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# 转换为单声道和16kHz采样率(确保与编码器匹配)
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audio = audio.set_channels(1).set_frame_rate(16000)
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# 获取原始PCM数据(16位小端)
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raw_data = audio.raw_data
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# 初始化Opus编码器
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encoder = opuslib_next.Encoder(16000, 1, opuslib_next.APPLICATION_AUDIO)
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# 编码参数
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frame_duration = 60 # 60ms per frame
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frame_size = int(16000 * frame_duration / 1000) # 960 samples/frame
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opus_datas = []
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# 按帧处理所有音频数据(包括最后一帧可能补零)
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for i in range(0, len(raw_data), frame_size * 2): # 16bit=2bytes/sample
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# 获取当前帧的二进制数据
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chunk = raw_data[i:i + frame_size * 2]
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# 如果最后一帧不足,补零
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if len(chunk) < frame_size * 2:
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chunk += b'\x00' * (frame_size * 2 - len(chunk))
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# 转换为numpy数组处理
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np_frame = np.frombuffer(chunk, dtype=np.int16)
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# 编码Opus数据
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opus_data = encoder.encode(np_frame.tobytes(), frame_size)
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opus_datas.append(opus_data)
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return opus_datas, duration
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