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"""demo 2/3.ipynb |
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Automatically generated by Colaboratory. |
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Original file is located at |
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https://colab.research.google.com/drive/1QeNS57tZzvJudeNjQczKJ-PbN0l1tK6V |
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# Import library |
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""" |
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import os |
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import librosa |
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import gradio as gr |
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import noisereduce as nr |
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from scipy.io import wavfile |
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from transformers import WhisperProcessor, WhisperForConditionalGeneration |
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"""# Load model""" |
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from google.colab import drive |
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import os |
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drive.mount('/content/gdrive') |
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processor = WhisperProcessor.from_pretrained("/content/gdrive/MyDrive/ColabNotebookShared/Speech2TextHuyenNguyen/Model/FPTVinTest2") |
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model = WhisperForConditionalGeneration.from_pretrained("/content/gdrive/MyDrive/ColabNotebookShared/Speech2TextHuyenNguyen/Model/FPTVinTest2/checkpoint-1332").to("cuda") |
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model.config.forced_decoder_ids = processor.get_decoder_prompt_ids(task = "transcribe") |
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"""# Slipt audio""" |
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from pydub import AudioSegment |
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def preprocessing(path): |
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type_file = path.split(".")[-1] |
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sound = AudioSegment.from_file(path, type_file) |
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path_list = [] |
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time_audio = int(sound.duration_seconds / 20) + 1 |
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for i in range(time_audio): |
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t1 = i * 20 * 1000 |
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t2 = (i+1) * 20 * 1000 |
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if i == (time_audio-1): |
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newAudio = sound[t1:] |
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else: |
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newAudio = sound[t1:t2] |
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newAudio = newAudio.split_to_mono()[0] |
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newAudio = newAudio.set_frame_rate(16000) |
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audio_path = '/content/new_audio' + str(i) + '.wav' |
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newAudio.export(audio_path, format="wav") |
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path_list.append(audio_path) |
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return path_list |
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"""# Capitalization""" |
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!git lfs install |
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!git clone https://github.com/huyenxam/Vicap.git |
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import os |
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from gec_model import GecBERTModel |
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cache_dir = "./" |
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model_cap = GecBERTModel( |
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vocab_path=os.path.join(cache_dir, "vocabulary"), |
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model_paths="dragonSwing/vibert-capu", |
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split_chunk=True |
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) |
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"""# Spelling Correction""" |
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from transformers import AutoTokenizer, AutoModelForSeq2SeqLM |
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tokenizer_spell = AutoTokenizer.from_pretrained("VietAI/vit5-base") |
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model_spell = AutoModelForSeq2SeqLM.from_pretrained("HuyenNguyen/Vi-test1") |
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model_spell.cuda() |
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def spelling_text(text): |
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encoding = tokenizer_spell(text, return_tensors="pt") |
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input_ids, attention_masks = encoding["input_ids"].to("cuda"), encoding["attention_mask"].to("cuda") |
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outputs = model_spell.generate( |
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input_ids=input_ids, attention_mask=attention_masks, |
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max_length=30, |
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) |
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for output in outputs: |
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line = tokenizer_spell.decode(output, skip_special_tokens=True, clean_up_tokenization_spaces=True) |
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return line |
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def spelling(transcription): |
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sentences = transcription.split(" ") |
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len_sen = int(len(sentences) / 25) + 1 |
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result = "" |
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for i in range(len_sen): |
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t1 = i * 24 |
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t2 = (i+1) * 24 |
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if i == (len_sen - 1): |
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text = " ".join(sentences[t1:]) |
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else: |
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text = " ".join(sentences[t1:t2]) |
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result = result + " " + spelling_text(text) |
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return result |
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"""# Speech To Text""" |
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import torch |
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import numpy as np |
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import gradio as gr |
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from scipy.io.wavfile import write |
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import pytube as pt |
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from transformers import pipeline |
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from huggingface_hub import model_info |
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def transcribe(microphone, file_upload): |
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warn_output = "" |
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if (microphone is not None) and (file_upload is not None): |
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warn_output = ( |
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"WARNING: You've uploaded an audio file and used the microphone. " |
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"The recorded file from the microphone will be used and the uploaded audio will be discarded.\n" |
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) |
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elif (microphone is None) and (file_upload is None): |
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return "ERROR: You have to either use the microphone or upload an audio file" |
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path = microphone if microphone is not None else file_upload |
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X_new, sr_new = librosa.load(path) |
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dst = "/content/audio.wav" |
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write(dst, sr_new, X_new) |
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transcription = "" |
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path_list = preprocessing(dst) |
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for audio_path in path_list: |
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X, sr = librosa.load(audio_path, sr=16000) |
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input_features = processor(X.astype('float16'), return_tensors="pt").input_features |
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predicted_ids = model.generate(input_features.to("cuda")) |
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text = processor.batch_decode(predicted_ids, skip_special_tokens = True)[0] |
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transcription = transcription + " " + text |
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transcription_spell = spelling(transcription) |
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transcription_cap = model_cap(transcription_spell)[0] |
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return transcription_cap |
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def _return_yt_html_embed(yt_url): |
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video_id = yt_url.split("?v=")[-1] |
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HTML_str = ( |
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f'<center> <iframe width="500" height="320" src="https://www.youtube.com/embed/{video_id}"> </iframe>' |
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" </center>" |
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) |
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return HTML_str |
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def yt_transcribe(yt_url): |
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return "ouput", 'This feature is temporarily locked' |
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demo = gr.Blocks() |
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mf_transcribe = gr.Interface( |
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fn=transcribe, |
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inputs=[ |
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gr.inputs.Audio(source="microphone", type="filepath", optional=True), |
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gr.inputs.Audio(source="upload", type="filepath", optional=True), |
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], |
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outputs="text", |
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layout="horizontal", |
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theme="huggingface", |
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title="PYLAB Demo: Transcribe Audio", |
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allow_flagging="never", |
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) |
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yt_transcribe = gr.Interface( |
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fn=yt_transcribe, |
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inputs=[gr.inputs.Textbox(lines=1, placeholder="Paste the URL to a YouTube video here", label="YouTube URL")], |
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outputs=["html", "text"], |
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layout="horizontal", |
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theme="huggingface", |
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title="PYLAB Demo: Transcribe YouTube", |
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allow_flagging="never", |
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) |
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with demo: |
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gr.TabbedInterface([mf_transcribe, yt_transcribe], ["Transcribe Audio", "Transcribe YouTube"]) |
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demo.launch(enable_queue=True) |
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