Spaces:
Sleeping
Sleeping
| # import part | |
| import streamlit as st | |
| from transformers import pipeline | |
| from gtts import gTTS | |
| import io | |
| # function part | |
| # img2text | |
| def img2text(url): | |
| image_to_text_model = pipeline("image-to-text", | |
| model="Salesforce/blip-image-captioning-base") | |
| text = image_to_text_model(url)[0]["generated_text"] | |
| return text | |
| # text2story | |
| def text2story(text): | |
| # 使用 Hugging Face 的 text-generation 模型生成故事 | |
| story_pipeline = pipeline("text-generation", model="agentica-org/DeepScaleR-1.5B-Preview") | |
| result = story_pipeline(text, max_length=200, num_return_sequences=1) | |
| story_text = result[0]['generated_text'] | |
| return story_text | |
| # text2audio | |
| def text2audio(story_text): | |
| # 使用 gTTS 将文本转换为音频 | |
| tts = gTTS(text=story_text, lang='en') | |
| # 创建一个内存中的字节流对象,用于存储音频数据 | |
| audio_file = io.BytesIO() | |
| # 将音频数据写入字节流 | |
| tts.write_to_fp(audio_file) | |
| # 将文件指针移动到文件开头,以便后续读取 | |
| audio_file.seek(0) | |
| return {'audio': audio_file, 'sampling_rate': 16000} # 返回音频数据和采样率 | |
| # main part | |
| st.set_page_config(page_title="Your Image to Audio Story", | |
| page_icon="🦜") | |
| st.header("Turn Your Image to Audio Story") | |
| uploaded_file = st.file_uploader("Select an Image...") | |
| if uploaded_file is not None: | |
| print(uploaded_file) | |
| bytes_data = uploaded_file.getvalue() | |
| with open(uploaded_file.name, "wb") as file: | |
| file.write(bytes_data) | |
| st.image(uploaded_file, caption="Uploaded Image", | |
| use_column_width=True) | |
| # Stage 1: Image to Text | |
| st.text('Processing img2text...') | |
| scenario = img2text(uploaded_file.name) | |
| st.write(scenario) | |
| # Stage 2: Text to Story | |
| st.text('Generating a story...') | |
| story = text2story(scenario) | |
| st.write(story) | |
| # Stage 3: Story to Audio data | |
| st.text('Generating audio data...') | |
| audio_data = text2audio(story) | |
| # Play button | |
| if st.button("Play Audio"): | |
| st.audio(audio_data['audio'], | |
| format="audio/wav", | |
| start_time=0, | |
| sample_rate=audio_data['sampling_rate']) |