import torch # Add this line import gradio as gr from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor, pipeline, AutoTokenizer import numpy as np import librosa # Load the models and processors asr_model = Wav2Vec2ForCTC.from_pretrained("Akashpb13/Hausa_xlsr") asr_processor = Wav2Vec2Processor.from_pretrained("Akashpb13/Hausa_xlsr") translator = pipeline("text2text-generation", model="Baghdad99/saad-hausa-text-to-english-text") tts = pipeline("text-to-speech", model="Baghdad99/english_voice_tts") def translate_speech(audio_input): # Load the audio file as a floating point time series audio_data, sample_rate = librosa.load(audio_input, sr=16000) # Prepare the input dictionary input_dict = asr_processor(audio_data, sampling_rate=sample_rate, return_tensors="pt", padding=True) # 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 # Use the translation pipeline to translate the transcription translated_text = translator(transcription, return_tensors="pt") print(f"Translated text: {translated_text}") # Print the translated text # Check if the translated text contains 'generated_token_ids' if 'generated_token_ids' in translated_text[0]: # Decode the tokens into text translated_text_str = translator.tokenizer.decode(translated_text[0]['generated_token_ids']) # Remove padding tokens translated_text_str = translated_text_str.replace("", "").strip() print(f"Translated text string: {translated_text_str}") # Print the translated text string else: print("The translated text does not contain 'generated_token_ids'") return # 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 # Define the Gradio interface iface = gr.Interface( fn=translate_speech, inputs=gr.inputs.Audio(type="filepath"), # Change this line outputs=gr.outputs.Audio(type="numpy"), title="Hausa to English Translation", description="Realtime demo for Hausa to English translation using speech recognition and text-to-speech synthesis." ) iface.launch()