Create app.py
Browse files
app.py
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import os
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import torch
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import gradio as gr
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from fastapi import FastAPI, UploadFile, File
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from fastapi.responses import JSONResponse
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import uvicorn
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
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import soundfile as sf
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import numpy as np
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import tempfile
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# Initialize FastAPI app
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app = FastAPI()
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# Initialize model and processor
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
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model_id = "nyrahealth/CrisperWhisper"
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model = AutoModelForSpeechSeq2Seq.from_pretrained(
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model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
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)
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model.to(device)
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processor = AutoProcessor.from_pretrained(model_id)
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pipe = pipeline(
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"automatic-speech-recognition",
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model=model,
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tokenizer=processor.tokenizer,
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feature_extractor=processor.feature_extractor,
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chunk_length_s=30,
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batch_size=16,
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return_timestamps='word',
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torch_dtype=torch_dtype,
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device=device,
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)
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def adjust_pauses_for_hf_pipeline_output(pipeline_output, split_threshold=0.12):
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"""
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Adjust pause timings by distributing pauses up to the threshold evenly between adjacent words.
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"""
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adjusted_chunks = pipeline_output["chunks"].copy()
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for i in range(len(adjusted_chunks) - 1):
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current_chunk = adjusted_chunks[i]
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next_chunk = adjusted_chunks[i + 1]
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current_start, current_end = current_chunk["timestamp"]
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next_start, next_end = next_chunk["timestamp"]
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pause_duration = next_start - current_end
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if pause_duration > 0:
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if pause_duration > split_threshold:
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distribute = split_threshold / 2
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else:
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distribute = pause_duration / 2
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adjusted_chunks[i]["timestamp"] = (current_start, current_end + distribute)
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adjusted_chunks[i + 1]["timestamp"] = (next_start - distribute, next_end)
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pipeline_output["chunks"] = adjusted_chunks
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return pipeline_output
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def process_audio(audio_path):
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"""Process audio file and return transcription with timestamps"""
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try:
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# Read audio file
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audio_data, sample_rate = sf.read(audio_path)
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# Convert to mono if stereo
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if len(audio_data.shape) > 1:
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audio_data = audio_data.mean(axis=1)
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# Process with pipeline
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result = pipe({"array": audio_data, "sampling_rate": sample_rate})
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# Adjust pauses
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adjusted_result = adjust_pauses_for_hf_pipeline_output(result)
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return adjusted_result
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except Exception as e:
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return {"error": str(e)}
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# FastAPI endpoint
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@app.post("/transcribe")
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async def transcribe_audio(file: UploadFile = File(...)):
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try:
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# Save uploaded file temporarily
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with tempfile.NamedTemporaryFile(delete=False, suffix=os.path.splitext(file.filename)[1]) as temp_file:
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content = await file.read()
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temp_file.write(content)
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temp_file_path = temp_file.name
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# Process the audio
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result = process_audio(temp_file_path)
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# Clean up temporary file
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os.unlink(temp_file_path)
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return JSONResponse(content=result)
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except Exception as e:
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return JSONResponse(
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status_code=500,
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content={"error": str(e)}
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)
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# Gradio interface
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def gradio_transcribe(audio):
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if audio is None:
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return "Please upload an audio file"
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result = process_audio(audio)
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return result
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# Create Gradio interface
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demo = gr.Interface(
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fn=gradio_transcribe,
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inputs=gr.Audio(type="filepath", label="Upload Audio (MP3 or WAV)"),
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outputs=gr.JSON(label="Transcription with Timestamps"),
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title="CrisperWhisper Audio Transcription",
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description="Upload an audio file to get transcription with word-level timestamps",
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examples=[],
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allow_flagging="never"
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)
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# Mount Gradio app
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app = gr.mount_gradio_app(app, demo, path="/")
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if __name__ == "__main__":
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uvicorn.run(app, host="0.0.0.0", port=7860)
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