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https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-22 07:03:53 +08:00
update:优化tts异常捕获
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@@ -30,13 +30,15 @@ class TTSProviderBase(ABC):
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asyncio.run(self.text_to_speak(text, tmp_file))
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except Exception as e:
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logger.bind(tag=TAG).error(f"语音生成失败: {text},错误: {e}")
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if not os.path.exists(tmp_file):
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max_repeat_time -= 1
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if max_repeat_time > 0:
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logger.bind(tag=TAG).error(f"再试{max_repeat_time}次")
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asyncio.sleep(max_repeat_time / 10)
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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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logger.bind(tag=TAG).info(
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f"语音生成成功: {text}:{tmp_file},重试{5 - max_repeat_time}次"
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)
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return tmp_file
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except Exception as e:
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@@ -52,9 +54,11 @@ class TTSProviderBase(ABC):
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# 获取文件后缀名
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file_type = os.path.splitext(audio_file_path)[1]
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if file_type:
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file_type = file_type.lstrip('.')
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file_type = file_type.lstrip(".")
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# 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
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audio = AudioSegment.from_file(audio_file_path, format=file_type, parameters=["-nostdin"])
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audio = AudioSegment.from_file(
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audio_file_path, format=file_type, parameters=["-nostdin"]
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)
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# 转换为单声道/16kHz采样率/16位小端编码(确保与编码器匹配)
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audio = audio.set_channels(1).set_frame_rate(16000).set_sample_width(2)
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@@ -76,11 +80,11 @@ class TTSProviderBase(ABC):
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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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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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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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