import torch import librosa import gradio as gr from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor, AutoProcessor, SeamlessM4Tv2Model import numpy as np import soundfile as sf import tempfile # Load the models and processors asr_model = Wav2Vec2ForCTC.from_pretrained("jonatasgrosman/wav2vec2-large-xlsr-53-english") asr_processor = Wav2Vec2Processor.from_pretrained("jonatasgrosman/wav2vec2-large-xlsr-53-english") # Load the SeamlessM4T model and processor translator_model = SeamlessM4Tv2Model.from_pretrained("facebook/seamless-m4t-v2-large") translator_processor = AutoProcessor.from_pretrained("facebook/seamless-m4t-v2-large") tts = pipeline("text-to-speech", model="Baghdad99/hausa_voice_tts") def translate_speech(audio_file_path): # Load the audio file as a floating point time series audio_data, sample_rate = librosa.load(audio_file_path, sr=16000) # Prepare the input dictionary input_dict = asr_processor(audio_data, sampling_rate=16000, return_tensors="pt", padding=True) # Pass the resampled audio_data here # Use the ASR model to get the logits logits = asr_model(input_dict.input_values.to("cpu")).logits # Get the predicted IDs pred_ids = torch.argmax(logits, dim=-1)[0] # Decode the predicted IDs to get the transcription transcription = asr_processor.decode(pred_ids) print(f"Transcription: {transcription}") # Print the transcription # Prepare the input dictionary for the translator text_inputs = translator_processor(text=transcription, src_lang="eng", return_tensors="pt") # Use the translator model to translate the transcription translated_text = translator_model.generate(**text_inputs, tgt_lang="ha") # Change the target language to Hausa # Decode the translated text translated_text_str = translator_processor.decode(translated_text[0]) # Remove special tokens translated_text_str = translated_text_str.replace("", "").replace("", "").strip() print(f"Translated text string: {translated_text_str}") # Print the translated text string # Use the text-to-speech pipeline to synthesize the translated text synthesised_speech = tts(translated_text_str) # Check if the synthesised speech contains 'audio' if 'audio' in synthesised_speech: synthesised_speech_data = synthesised_speech['audio'] else: print("The synthesised speech does not contain 'audio'") return # Flatten the audio data synthesised_speech_data = synthesised_speech_data.flatten() # Scale the audio data to the range of int16 format synthesised_speech = (synthesised_speech_data * 32767).astype(np.int16) return 16000, synthesised_speech