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| import torch | |
| import time | |
| import gradio as gr | |
| import spaces | |
| from transformers import pipeline | |
| from transformers.pipelines.audio_utils import ffmpeg_read | |
| DEFAULT_MODEL_NAME = "openai/whisper-tiny" | |
| BATCH_SIZE = 8 | |
| device = 0 if torch.cuda.is_available() else "cpu" | |
| def load_pipeline(model_name): | |
| return pipeline( | |
| task="automatic-speech-recognition", | |
| model=model_name, | |
| chunk_length_s=30, | |
| device=device, | |
| ) | |
| pipe = load_pipeline(DEFAULT_MODEL_NAME) | |
| def transcribe(inputs, task, model_name): | |
| if inputs is None: | |
| raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting your request.") | |
| global pipe | |
| if model_name != pipe.model.name_or_path: | |
| pipe = load_pipeline(model_name) | |
| start_time = time.time() # Record the start time | |
| text = pipe(inputs, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=True)["text"] | |
| end_time = time.time() # Record the end time | |
| transcription_time = end_time - start_time # Calculate the transcription time | |
| # Create the transcription time output with additional information | |
| transcription_time_output = ( | |
| f"Transcription Time: {transcription_time:.2f} seconds\n" | |
| f"Model Used: {model_name}\n" | |
| f"Device Used: {'GPU' if torch.cuda.is_available() else 'CPU'}" | |
| ) | |
| return text, transcription_time_output | |
| demo = gr.Blocks() | |
| mf_transcribe = gr.Interface( | |
| fn=transcribe, | |
| inputs=[ | |
| gr.Audio(type="filepath"), | |
| gr.Radio(["transcribe", "translate"], label="Task", value="transcribe"), | |
| gr.Textbox( | |
| label="Model Name", | |
| value=DEFAULT_MODEL_NAME, | |
| placeholder="Enter the model name", | |
| info="Some available models: distil-whisper/distil-large-v3 distil-whisper/distil-medium.en Systran/faster-distil-whisper-large-v3 Systran/faster-whisper-large-v3 Systran/faster-whisper-medium openai/whisper-tiny , openai/whisper-base, openai/whisper-medium, openai/whisper-large-v3" | |
| ), | |
| ], | |
| outputs=[gr.TextArea(label="Transcription"), gr.TextArea(label="Transcription Info")], | |
| theme="huggingface", | |
| title="Whisper Transcription", | |
| description=( | |
| "Transcribe long-form microphone or audio inputs with the click of a button! Demo uses the specified OpenAI Whisper" | |
| " checkpoint and 🤗 Transformers to transcribe audio files of arbitrary length." | |
| ), | |
| allow_flagging="never", | |
| ) | |
| file_transcribe = gr.Interface( | |
| fn=transcribe, | |
| inputs=[ | |
| gr.Audio(type="filepath", label="Audio file"), | |
| gr.Radio(["transcribe", "translate"], label="Task", value="transcribe"), | |
| gr.Textbox( | |
| label="Model Name", | |
| value=DEFAULT_MODEL_NAME, | |
| placeholder="Enter the model name", | |
| info="Some available models: openai/whisper-tiny, openai/whisper-base, openai/whisper-medium, openai/whisper-large-v2" | |
| ), | |
| ], | |
| outputs=[gr.TextArea(label="Transcription"), gr.TextArea(label="Transcription Info")], | |
| theme="huggingface", | |
| title="Whisper Transcription", | |
| description=( | |
| "Transcribe long-form microphone or audio inputs with the click of a button! Demo uses the specified OpenAI Whisper" | |
| " checkpoint and 🤗 Transformers to transcribe audio files of arbitrary length." | |
| ), | |
| allow_flagging="never", | |
| ) | |
| with demo: | |
| gr.TabbedInterface([mf_transcribe, file_transcribe], ["Microphone", "Audio file"]) | |
| demo.launch(share=True) |