import io
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import wave
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import json
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import base64
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import asyncio
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import websockets
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import numpy as np
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from datetime import datetime
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from config.logger import setup_logging
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from core.providers.tts.base import TTSProviderBase
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TAG = __name__
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logger = setup_logging()
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class TTSProvider(TTSProviderBase):
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TTS_PARAM_CONFIG = [
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("ttsVolume", "volume", 0, 3, 1.0, lambda v: round(float(v), 1)),
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("ttsRate", "speed", 0, 3, 1.0, lambda v: round(float(v), 1)),
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]
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def __init__(self, config, delete_audio_file):
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super().__init__(config, delete_audio_file)
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self.url = config.get("url", "ws://192.168.1.10:8092/paddlespeech/tts/streaming")
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self.protocol = config.get("protocol", "websocket")
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if config.get("private_voice"):
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self.spk_id = int(config.get("private_voice"))
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else:
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self.spk_id = int(config.get("spk_id", "0"))
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speed = config.get("speed", 1.0)
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self.speed = float(speed) if speed else 1.0
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volume = config.get("volume", 1.0)
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self.volume = float(volume) if volume else 1.0
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self.delete_audio_file = config.get("delete_audio", True)
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# 应用百分比调整（如果存在），否则使用公有化配置
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self._apply_percentage_params(config)
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if not self.delete_audio_file:
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timestamp = datetime.now().strftime("%Y%m%d_%H%M%S")
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save_path = config.get("save_path")
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if save_path:
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if not save_path.endswith('.wav'):
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save_path = f"{save_path}_{timestamp}.wav"
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else:
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other_path = save_path[:-4]
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save_path = f"{other_path}_{timestamp}.wav"
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self.save_path = save_path
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else:
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self.save_path = f"./streaming_tts_{timestamp}.wav"
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else:
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self.save_path = None
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async def pcm_to_wav(self, pcm_data: bytes, sample_rate: int = 24000, num_channels: int = 1,
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bits_per_sample: int = 16) -> bytes:
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"""
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将 PCM 数据转换为 WAV 文件并返回字节数据
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:param pcm_data: PCM 数据（原始字节流）
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:param sample_rate: 音频采样率，默认为24000
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:param num_channels: 声道数，默认为单声道
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:param bits_per_sample: 每个样本的位数，默认为16
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:return: WAV 格式的字节数据
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"""
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byte_data = np.frombuffer(pcm_data, dtype=np.int16) # 16位PCM
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wav_io = io.BytesIO()
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with wave.open(wav_io, "wb") as wav_file:
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wav_file.setnchannels(num_channels)
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wav_file.setsampwidth(bits_per_sample // 8)
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wav_file.setframerate(sample_rate)
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wav_file.writeframes(byte_data.tobytes())
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return wav_io.getvalue()
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async def text_to_speak(self, text, output_file):
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if self.protocol == "websocket":
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return await self.text_streaming(text, output_file)
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else:
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raise ValueError("Unsupported protocol. Please use 'websocket' or 'http'.")
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async def text_streaming(self, text, output_file):
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try:
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# 使用 websockets 异步连接到 WebSocket 服务器
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async with websockets.connect(self.url) as ws:
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# 发送开始请求
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start_request = {
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"task": "tts",
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"signal": "start"
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}
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await ws.send(json.dumps(start_request))
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# 接收开始响应并提取 session_id
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start_response = await ws.recv()
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start_response = json.loads(start_response) # 解析 JSON 响应
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if start_response.get("status") != 0:
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raise Exception(f"连接失败: {start_response.get('signal')}")
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session_id = start_response.get("session")
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# 发送待合成的文本数据
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data_request = {
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"text": text,
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"spk_id": self.spk_id,
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}
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await ws.send(json.dumps(data_request))
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audio_chunks = b""
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timeout_seconds = self.tts_timeout
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try:
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while True:
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response = await asyncio.wait_for(ws.recv(), timeout=timeout_seconds)
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response = json.loads(response) # 解析 JSON 响应
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status = response.get("status")
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if status == 2: # 最后一个数据包
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break
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else:
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# 拼接音频数据（base64 编码的 PCM 数据）
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audio_chunks += base64.b64decode(response.get("audio"))
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except asyncio.TimeoutError:
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raise Exception(f"WebSocket 超时：等待音频数据超过 {timeout_seconds} 秒")
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# 将拼接后的 PCM 数据转换为 WAV 格式
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wav_data = await self.pcm_to_wav(audio_chunks)
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# 结束请求
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end_request = {
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"task": "tts",
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"signal": "end",
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"session": session_id # 会话 ID 必须与开始请求中的一致
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}
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await ws.send(json.dumps(end_request))
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# 接收结束响应避免服务抛出异常
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await ws.recv()
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# 根据配置决定是否保存文件
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if not self.delete_audio_file and self.save_path:
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with open(self.save_path, "wb") as f:
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f.write(wav_data)
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logger.bind(tag=TAG).info(f"音频文件已保存到: {self.save_path}")
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# 返回或保存音频数据
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if output_file:
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with open(output_file, "wb") as file_to_save:
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file_to_save.write(wav_data)
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else:
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return wav_data
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except Exception as e:
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raise Exception(f"Error during TTS WebSocket request: {e} while processing text: {text}")
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