import torch import spaces import gradio as gr import os from pyannote.audio import Pipeline from pydub import AudioSegment # 获取 Hugging Face 认证令牌 HF_TOKEN = os.environ.get("HUGGINGFACE_READ_TOKEN") pipeline = None # 尝试加载 pyannote 模型 try: pipeline = Pipeline.from_pretrained( "pyannote/speaker-diarization-3.1", use_auth_token=HF_TOKEN ) device = torch.device("cuda" if torch.cuda.is_available() else "cpu") pipeline.to(device) except Exception as e: print(f"Error initializing pipeline: {e}") pipeline = None # 音频拼接函数:拼接目标音频和混合音频,返回目标音频的起始时间和结束时间作为字典 def combine_audio_with_time(target_audio, mixed_audio): if pipeline is None: return "错误: 模型未初始化" # 打印文件路径,确保文件正确传递 print(f"目标音频文件路径: {target_audio}") print(f"混合音频文件路径: {mixed_audio}") # 加载目标说话人的样本音频 try: target_audio_segment = AudioSegment.from_wav(target_audio) except Exception as e: return f"加载目标音频时出错: {e}" # 加载混合音频 try: mixed_audio_segment = AudioSegment.from_wav(mixed_audio) except Exception as e: return f"加载混合音频时出错: {e}" # 记录目标说话人音频的时间点(精确到0.01秒) target_start_time = len(mixed_audio_segment) / 1000 # 秒为单位,精确到 0.01 秒 # 目标音频的结束时间(拼接后的音频长度) target_end_time = target_start_time + len(target_audio_segment) / 1000 # 秒为单位 # 将目标说话人的音频片段添加到混合音频的最后 final_audio = mixed_audio_segment + target_audio_segment final_audio.export("final_output.wav", format="wav") # 返回目标音频的起始时间和结束时间 return {"start_time": target_start_time, "end_time": target_end_time} # 使用 pyannote/speaker-diarization 对拼接后的音频进行说话人分离 @spaces.GPU(duration=60 * 2) # 使用 GPU 加速,限制执行时间为 120 秒 def diarize_audio(temp_file): if pipeline is None: return "错误: 模型未初始化" try: diarization = pipeline(temp_file) except Exception as e: return f"处理音频时出错: {e}" print(diarization) print(type(diarization)) # 返回 diarization 输出 return str(diarization) # 将时间戳转换为秒 def timestamp_to_seconds(timestamp): try: h, m, s = map(float, timestamp.split(':')) return 3600 * h + 60 * m + s except ValueError as e: print(f"转换时间戳时出错: '{timestamp}'. 错误: {e}") return None # 计算时间段的重叠部分(单位:秒) def calculate_overlap(start1, end1, start2, end2): overlap_start = max(start1, start2) overlap_end = min(end1, end2) overlap_duration = max(0, overlap_end - overlap_start) return overlap_duration # 获取目标时间段和说话人时间段的重叠比例 def get_best_match(target_time, diarization_output): target_start_time = target_time['start_time'] target_end_time = target_time['end_time'] # 假设 diarization_output 是一个列表,包含说话人时间段和标签 speaker_segments = [] for line in diarization_output.strip().split('\n'): try: parts = line.strip()[1:-1].split(' --> ') start_time = parts[0].strip() end_time = parts[1].split(']')[0].strip() label = line.split()[-1].strip() start_seconds = timestamp_to_seconds(start_time) end_seconds = timestamp_to_seconds(end_time) # 计算目标音频时间段和说话人时间段的重叠时间 overlap = calculate_overlap(target_start_time, target_end_time, start_seconds, end_seconds) overlap_ratio = overlap / (end_seconds - start_seconds) # 记录说话人标签和重叠比例 speaker_segments.append((label, overlap_ratio, start_seconds, end_seconds)) except Exception as e: print(f"处理行时出错: '{line.strip()}'. 错误: {e}") # 按照重叠比例排序,返回重叠比例最大的一段 best_match = max(speaker_segments, key=lambda x: x[1], default=None) return best_match # 处理音频文件并返回输出 def process_audio(target_audio, mixed_audio): # 打印文件路径,确保传入的文件有效 print(f"处理音频:目标音频: {target_audio}, 混合音频: {mixed_audio}") # 进行音频拼接并返回目标音频的起始和结束时间(作为字典) time_dict = combine_audio_with_time(target_audio, mixed_audio) # 执行说话人分离 diarization_result = diarize_audio("final_output.wav") if diarization_result.startswith("错误"): return diarization_result, None # 出错时返回错误信息 else: # 获取最佳匹配的说话人标签和时间段 best_match = get_best_match(time_dict, diarization_result) if best_match: # 返回最佳匹配说话人的标签和时间段 return best_match[0], best_match[2], best_match[3] # Gradio 接口 with gr.Blocks() as demo: gr.Markdown(""" # 🗣️ 音频拼接与说话人分类 🗣️ 上传目标音频和混合音频,拼接并进行说话人分类。结果包括最佳匹配说话人的时间段。 """) mixed_audio_input = gr.Audio(type="filepath", label="上传混合音频") target_audio_input = gr.Audio(type="filepath", label="上传目标说话人音频") process_button = gr.Button("处理音频") # 输出结果 diarization_output = gr.Textbox(label="最佳匹配说话人") time_range_output = gr.Textbox(label="最佳匹配时间段") # 点击按钮时触发处理音频 process_button.click( fn=process_audio, inputs=[target_audio_input, mixed_audio_input], outputs=[diarization_output, time_range_output] ) demo.launch(share=True)