import gradio as gr import torch # import torchaudio from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan from datasets import load_dataset from transformers import pipeline def text_to_audio(text): # 这里是你的文本转音频的算法实现 # 加载处理器和模型 processor = SpeechT5Processor.from_pretrained("/home/tt/speecht5/speecht5_finetuned/checkpoint-19000") model = SpeechT5ForTextToSpeech.from_pretrained("/home/tt/speecht5/speecht5_finetuned/checkpoint-19000") # 加载说话者嵌入数据集 embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation") speaker_index = 0 speaker_embeddings = torch.tensor(embeddings_dataset[speaker_index]["xvector"]).unsqueeze(0) # 处理输入文本 inputs = processor(text=text, return_tensors="pt") # 加载声码器 vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan") # 生成语音 speech = model.generate_speech(inputs["input_ids"], speaker_embeddings, vocoder=vocoder) # 调整音频张量的形状以匹配 torchaudio.save 的要求 speech_2d = speech.unsqueeze(0) # 添加一个维度,使其成为二维张量 # 保存音频为 .wav 文件 # torchaudio.save("generated_speech1.wav", speech_2d, 16000) return speech def audio_to_text(audio): sr, audio_array = audio data_dict = {"array" : audio_array, "sampling_rate" : sr} device = "cuda:0" if torch.cuda.is_available() else "cpu" pipe = pipeline( "automatic-speech-recognition", model="/home/rouvling/temp_lug/gradio/asr/checkpoint-4000-old", device=device ) result=pipe(data_dict, max_new_tokens=256) return result['text'] text_to_audio_interface = gr.Interface( fn=text_to_audio, inputs="text", outputs="audio", title="Text to Audio", description="输入文字,生成对应音频", ) audio_to_text_interface = gr.Interface( fn=audio_to_text, inputs="audio", outputs="text", title="Audio to Text", description="上传音频,生成对应文字", ) demo = gr.TabbedInterface( [text_to_audio_interface, audio_to_text_interface], ["Text to Audio", "Audio to Text"], ) demo.launch(server_name="0.0.0.0", server_port=8080)