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Harveenchadha
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89702e3
Create app.py
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app.py
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import soundfile as sf
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import torch
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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from transformers import M2M100ForConditionalGeneration, M2M100Tokenizer
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import gradio as gr
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import sox
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import numpy as np
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import yaml
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import tensorflow as tf
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from tensorflow_tts.inference import TFAutoModel
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from tensorflow_tts.inference import AutoProcessor
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# initialize fastspeech2 model.
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fastspeech2 = TFAutoModel.from_pretrained("tensorspeech/tts-fastspeech2-ljspeech-en")
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# initialize mb_melgan model
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mb_melgan = TFAutoModel.from_pretrained("tensorspeech/tts-mb_melgan-ljspeech-en")
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# inference
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processor = AutoProcessor.from_pretrained("tensorspeech/tts-fastspeech2-ljspeech-en")
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def tts(text):
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input_ids = processor.text_to_sequence(text)
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# fastspeech inference
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mel_before, mel_after, duration_outputs, _, _ = fastspeech2.inference(
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input_ids=tf.expand_dims(tf.convert_to_tensor(input_ids, dtype=tf.int32), 0),
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speaker_ids=tf.convert_to_tensor([0], dtype=tf.int32),
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speed_ratios=tf.convert_to_tensor([1.0], dtype=tf.float32),
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f0_ratios =tf.convert_to_tensor([1.0], dtype=tf.float32),
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energy_ratios =tf.convert_to_tensor([1.0], dtype=tf.float32),
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)
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# melgan inference
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audio_before = mb_melgan.inference(mel_before)[0, :, 0]
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audio_after = mb_melgan.inference(mel_after)[0, :, 0]
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# save to file
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sf.write('./audio_before.wav', audio_before, 22050, "PCM_16")
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sf.write('./audio_after.wav', audio_after, 22050, "PCM_16")
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return './audio_after.wav'
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def convert(inputfile, outfile):
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sox_tfm = sox.Transformer()
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sox_tfm.set_output_format(
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file_type="wav", channels=1, encoding="signed-integer", rate=16000, bits=16
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)
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sox_tfm.build(inputfile, outfile)
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model_translate = M2M100ForConditionalGeneration.from_pretrained("facebook/m2m100_418M")
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tokenizer_translate = M2M100Tokenizer.from_pretrained("facebook/m2m100_418M")
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inlang='hi'
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outlang='en'
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tokenizer_translate.src_lang = inlang
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def translate(text):
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encoded_hi = tokenizer_translate(text, return_tensors="pt")
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generated_tokens = model_translate.generate(**encoded_hi, forced_bos_token_id=tokenizer_translate.get_lang_id(outlang))
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return tokenizer_translate.batch_decode(generated_tokens, skip_special_tokens=True)[0]
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processor = Wav2Vec2Processor.from_pretrained("Harveenchadha/vakyansh-wav2vec2-hindi-him-4200")
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model = Wav2Vec2ForCTC.from_pretrained("Harveenchadha/vakyansh-wav2vec2-hindi-him-4200")
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def parse_transcription(wav_file):
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filename = wav_file.name.split('.')[0]
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convert(wav_file.name, filename + "16k.wav")
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speech, _ = sf.read(filename + "16k.wav")
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input_values = processor(speech, sampling_rate=16_000, return_tensors="pt").input_values
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.decode(predicted_ids[0], skip_special_tokens=True)
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translation = translate(transcription)
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return transcription, translation, tts(translation)
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output1 = gr.outputs.Textbox(label="Hindi Output from ASR")
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output2 = gr.outputs.Textbox(label="English Translated Output")
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input_ = gr.inputs.Audio(source="microphone", type="file")
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output_audio = gr.outputs.Audio(type="file", label="Output Audio")
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gr.Interface(parse_transcription, inputs = input_, outputs=[output1, output2, output_audio], analytics_enabled=False,
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show_tips=False,
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theme='huggingface',
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layout='vertical',
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title="Vakyansh: Speech To text for Indic Languages",
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description="This is a live demo for Speech to Speech Translation. Speak in Hindi and get output in English", enable_queue=True).launch( inline=False)
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