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import spaces |
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import torch |
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import gradio as gr |
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import tempfile |
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import os |
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import uuid |
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import scipy.io.wavfile |
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import time |
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import numpy as np |
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, WhisperTokenizer, pipeline |
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import subprocess |
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subprocess.run( |
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"pip install flash-attn --no-build-isolation", |
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env={"FLASH_ATTENTION_SKIP_CUDA_BUILD": "TRUE"}, |
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shell=True, |
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) |
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device = "cuda" if torch.cuda.is_available() else "cpu" |
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torch_dtype = torch.float16 |
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MODEL_NAME = "primeline/whisper-large-v3-turbo-german" |
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model = AutoModelForSpeechSeq2Seq.from_pretrained( |
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MODEL_NAME, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True, attn_implementation="flash_attention_2" |
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) |
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model.to(device) |
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processor = AutoProcessor.from_pretrained(MODEL_NAME) |
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tokenizer = WhisperTokenizer.from_pretrained(MODEL_NAME) |
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pipe = pipeline( |
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task="automatic-speech-recognition", |
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model=model, |
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tokenizer=tokenizer, |
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feature_extractor=processor.feature_extractor, |
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chunk_length_s=10, |
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torch_dtype=torch_dtype, |
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device=device, |
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) |
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@spaces.GPU |
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def transcribe(inputs, previous_transcription): |
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start_time = time.time() |
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try: |
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filename = f"{uuid.uuid4().hex}.wav" |
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sample_rate, audio_data = inputs |
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scipy.io.wavfile.write(filename, sample_rate, audio_data) |
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transcription = pipe(filename)["text"] |
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previous_transcription += transcription |
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end_time = time.time() |
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latency = end_time - start_time |
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return previous_transcription, f"{latency:.2f}" |
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except Exception as e: |
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print(f"Error during Transcription: {e}") |
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return previous_transcription, "Error" |
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@spaces.GPU |
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def translate_and_transcribe(inputs, previous_transcription, target_language): |
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start_time = time.time() |
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try: |
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filename = f"{uuid.uuid4().hex}.wav" |
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sample_rate, audio_data = inputs |
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scipy.io.wavfile.write(filename, sample_rate, audio_data) |
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translation = pipe(filename, generate_kwargs={"task": "translate", "language": target_language} )["text"] |
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previous_transcription += translation |
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end_time = time.time() |
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latency = end_time - start_time |
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return previous_transcription, f"{latency:.2f}" |
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except Exception as e: |
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print(f"Error during Translation and Transcription: {e}") |
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return previous_transcription, "Error" |
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def clear(): |
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return "" |
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with gr.Blocks() as microphone: |
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with gr.Column(): |
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gr.Markdown(f"# Realtime Whisper Large V3 Turbo German: \n Transcribe Audio in Realtime. This Demo uses the Checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) and 🤗 Transformers.\n Note: The first token takes about 5 seconds. After that, it works flawlessly.") |
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with gr.Row(): |
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input_audio_microphone = gr.Audio(streaming=True) |
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output = gr.Textbox(label="Transcription", value="") |
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latency_textbox = gr.Textbox(label="Latency (seconds)", value="0.0", scale=0) |
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with gr.Row(): |
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clear_button = gr.Button("Clear Output") |
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input_audio_microphone.stream(transcribe, [input_audio_microphone, output], [output, latency_textbox], time_limit=45, stream_every=2, concurrency_limit=None) |
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clear_button.click(clear, outputs=[output]) |
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with gr.Blocks() as file: |
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with gr.Column(): |
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gr.Markdown(f"# Realtime Whisper Large V3 Turbo German: \n Transcribe Audio in Realtime. This Demo uses the Checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) and 🤗 Transformers.\n Note: The first token takes about 5 seconds. After that, it works flawlessly.") |
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with gr.Row(): |
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input_audio_microphone = gr.Audio(sources="upload", type="numpy") |
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output = gr.Textbox(label="Transcription", value="") |
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latency_textbox = gr.Textbox(label="Latency (seconds)", value="0.0", scale=0) |
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with gr.Row(): |
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submit_button = gr.Button("Submit") |
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clear_button = gr.Button("Clear Output") |
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submit_button.click(transcribe, [input_audio_microphone, output], [output, latency_textbox], concurrency_limit=None) |
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clear_button.click(clear, outputs=[output]) |
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with gr.Blocks(theme=gr.themes.Ocean()) as demo: |
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gr.TabbedInterface([microphone, file], ["Microphone", "Transcribe from file"]) |
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demo.launch() |