import os os.system("pip install git+https://github.com/suno-ai/bark.git") from bark.generation import SUPPORTED_LANGS from bark import SAMPLE_RATE, generate_audio from scipy.io.wavfile import write as write_wav from datetime import datetime import shutil import gradio as gr import sys import string import time import argparse import json import numpy as np # import IPython # from IPython.display import Audio import torch from TTS.tts.utils.synthesis import synthesis from TTS.tts.utils.text.symbols import make_symbols, phonemes, symbols try: from TTS.utils.audio import AudioProcessor except: from TTS.utils.audio import AudioProcessor from TTS.tts.models import setup_model from TTS.config import load_config from TTS.tts.models.vits import * from TTS.tts.utils.speakers import SpeakerManager from pydub import AudioSegment # from google.colab import files import librosa from scipy.io.wavfile import write, read import subprocess ''' from google.colab import drive drive.mount('/content/drive') src_path = os.path.join(os.path.join(os.path.join(os.path.join(os.getcwd(), 'drive'), 'MyDrive'), 'Colab Notebooks'), 'best_model_latest.pth.tar') dst_path = os.path.join(os.getcwd(), 'best_model.pth.tar') shutil.copy(src_path, dst_path) ''' TTS_PATH = "TTS/" # add libraries into environment sys.path.append(TTS_PATH) # set this if TTS is not installed globally # Paths definition OUT_PATH = 'out/' # create output path os.makedirs(OUT_PATH, exist_ok=True) # model vars MODEL_PATH = 'best_model.pth.tar' CONFIG_PATH = 'config.json' TTS_LANGUAGES = "language_ids.json" TTS_SPEAKERS = "speakers.json" USE_CUDA = torch.cuda.is_available() # load the config C = load_config(CONFIG_PATH) # load the audio processor ap = AudioProcessor(**C.audio) speaker_embedding = None C.model_args['d_vector_file'] = TTS_SPEAKERS C.model_args['use_speaker_encoder_as_loss'] = False model = setup_model(C) model.language_manager.set_language_ids_from_file(TTS_LANGUAGES) # print(model.language_manager.num_languages, model.embedded_language_dim) # print(model.emb_l) cp = torch.load(MODEL_PATH, map_location=torch.device('cpu')) # remove speaker encoder model_weights = cp['model'].copy() for key in list(model_weights.keys()): if "speaker_encoder" in key: del model_weights[key] model.load_state_dict(model_weights) model.eval() if USE_CUDA: model = model.cuda() # synthesize voice use_griffin_lim = False # Paths definition CONFIG_SE_PATH = "config_se.json" CHECKPOINT_SE_PATH = "SE_checkpoint.pth.tar" # Load the Speaker encoder SE_speaker_manager = SpeakerManager(encoder_model_path=CHECKPOINT_SE_PATH, encoder_config_path=CONFIG_SE_PATH, use_cuda=USE_CUDA) # Define helper function def compute_spec(ref_file): y, sr = librosa.load(ref_file, sr=ap.sample_rate) spec = ap.spectrogram(y) spec = torch.FloatTensor(spec).unsqueeze(0) return spec def voice_conversion(ta, ra, da): target_audio = 'target.wav' reference_audio = 'reference.wav' driving_audio = 'driving.wav' write(target_audio, ta[0], ta[1]) write(reference_audio, ra[0], ra[1]) write(driving_audio, da[0], da[1]) # !ffmpeg-normalize $target_audio -nt rms -t=-27 -o $target_audio -ar 16000 -f # !ffmpeg-normalize $reference_audio -nt rms -t=-27 -o $reference_audio -ar 16000 -f # !ffmpeg-normalize $driving_audio -nt rms -t=-27 -o $driving_audio -ar 16000 -f files = [target_audio, reference_audio, driving_audio] for file in files: subprocess.run(["ffmpeg-normalize", file, "-nt", "rms", "-t=-27", "-o", file, "-ar", "16000", "-f"]) # ta_ = read(target_audio) target_emb = SE_speaker_manager.compute_d_vector_from_clip([target_audio]) target_emb = torch.FloatTensor(target_emb).unsqueeze(0) driving_emb = SE_speaker_manager.compute_d_vector_from_clip([reference_audio]) driving_emb = torch.FloatTensor(driving_emb).unsqueeze(0) # Convert the voice driving_spec = compute_spec(driving_audio) y_lengths = torch.tensor([driving_spec.size(-1)]) if USE_CUDA: ref_wav_voc, _, _ = model.voice_conversion(driving_spec.cuda(), y_lengths.cuda(), driving_emb.cuda(), target_emb.cuda()) ref_wav_voc = ref_wav_voc.squeeze().cpu().detach().numpy() else: ref_wav_voc, _, _ = model.voice_conversion(driving_spec, y_lengths, driving_emb, target_emb) ref_wav_voc = ref_wav_voc.squeeze().detach().numpy() # print("Reference Audio after decoder:") # IPython.display.display(Audio(ref_wav_voc, rate=ap.sample_rate)) return (ap.sample_rate, ref_wav_voc) def generate_text_to_speech(text_prompt, selected_speaker, text_temp, waveform_temp): audio_array = generate_audio(text_prompt, selected_speaker, text_temp, waveform_temp) now = datetime.now() date_str = now.strftime("%m-%d-%Y") time_str = now.strftime("%H-%M-%S") outputs_folder = os.path.join(os.getcwd(), "outputs") if not os.path.exists(outputs_folder): os.makedirs(outputs_folder) sub_folder = os.path.join(outputs_folder, date_str) if not os.path.exists(sub_folder): os.makedirs(sub_folder) file_name = f"audio_{time_str}.wav" file_path = os.path.join(sub_folder, file_name) write_wav(file_path, SAMPLE_RATE, audio_array) return file_path speakers_list = [] for lang, code in SUPPORTED_LANGS: for n in range(10): speakers_list.append(f"{code}_speaker_{n}") examples1 = [["ref.wav", "Bark.wav", "Bark.wav"]] with gr.Blocks() as demo: gr.Markdown( f""" #