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# !git clone https://github.com/Edresson/Coqui-TTS -b multilingual-torchaudio-SE TTS | |
import os | |
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 | |
from scipy.io import wavfile | |
import subprocess | |
import whisper | |
model = whisper.load_model("base") | |
os.system('pip install voicefixer --upgrade') | |
from voicefixer import VoiceFixer | |
voicefixer = VoiceFixer() | |
import openai | |
import torchaudio | |
from speechbrain.pretrained import SpectralMaskEnhancement | |
os.system('pip install -U numpy==1.21') | |
from TTS.api import TTS | |
tts = TTS(model_name="tts_models/zh-CN/baker/tacotron2-DDC-GST", progress_bar=False, gpu=True) | |
enhance_model = SpectralMaskEnhancement.from_hparams( | |
source="speechbrain/metricgan-plus-voicebank", | |
savedir="pretrained_models/metricgan-plus-voicebank", | |
run_opts={"device":"cuda"}, | |
) | |
mes1 = [ | |
{"role": "system", "content": "You are a TOEFL examiner. Help me improve my oral Englsih and give me feedback."} | |
] | |
mes2 = [ | |
{"role": "system", "content": "You are a mental health therapist. Respond to me only in Chinese. Your name is Tina."} | |
] | |
mes3 = [ | |
{"role": "system", "content": "You are my personal assistant. Respond to me only in Chinese. Your name is Alice."} | |
] | |
res = [] | |
''' | |
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(apikey, ta, audio, choice1): | |
openai.api_key = apikey | |
# load audio and pad/trim it to fit 30 seconds | |
audio = whisper.load_audio(audio) | |
audio = whisper.pad_or_trim(audio) | |
# make log-Mel spectrogram and move to the same device as the model | |
mel = whisper.log_mel_spectrogram(audio).to(model.device) | |
# detect the spoken language | |
_, probs = model.detect_language(mel) | |
print(f"Detected language: {max(probs, key=probs.get)}") | |
# decode the audio | |
options = whisper.DecodingOptions() | |
result = whisper.decode(model, mel, options) | |
res.append(result.text) | |
if choice1 == "TOEFL": | |
messages = mes1 | |
elif choice1 == "Therapist": | |
messages = mes2 | |
elif choice1 == "Alice": | |
messages = mes3 | |
# chatgpt | |
n = len(res) | |
content = res[n-1] | |
messages.append({"role": "user", "content": content}) | |
completion = openai.ChatCompletion.create( | |
model = "gpt-3.5-turbo", | |
messages = messages | |
) | |
chat_response = completion.choices[0].message.content | |
messages.append({"role": "assistant", "content": chat_response}) | |
tts.tts_to_file(chat_response + "。", file_path="output.wav") | |
target_audio = 'target.wav' | |
reference_audio = 'reference.wav' | |
driving_audio = 'driving.wav' | |
rate1, data1 = wavfile.read("output.wav") | |
write(target_audio, ta[0], ta[1]) | |
write(reference_audio, rate1, data1) | |
write(driving_audio, rate1, data1) | |
# !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)) | |
voicefixer.restore(input=ref_wav_voc, # input wav file path | |
output="audio1.wav", # output wav file path | |
cuda=True, # whether to use gpu acceleration | |
mode = 0) # You can try out mode 0, 1, or 2 to find out the best result | |
noisy = enhance_model.load_audio( | |
"audio1.wav" | |
).unsqueeze(0) | |
enhanced = enhance_model.enhance_batch(noisy, lengths=torch.tensor([1.])) | |
torchaudio.save("enhanced.wav", enhanced.cpu(), 16000) | |
return [result.text, chat_response, "enhanced.wav"] | |
c1=gr.Interface( | |
fn=voice_conversion, | |
inputs=[ | |
gr.Textbox(lines=1, label = "请填写您的OpenAI-API-key"), | |
gr.Audio(source="upload", label = "请上传您喜欢的声音(wav文件)", type="filepath"), | |
gr.Audio(source="microphone", label = "和您的专属AI聊天吧!", type="filepath"), | |
gr.Radio(["TOEFL", "Therapist", "Alice"], label="TOEFL Examiner, Therapist Tina, or Assistant Alice?"), | |
], | |
outputs=[ | |
gr.Textbox(label="Speech to Text"), gr.Textbox(label="ChatGPT Output"), gr.Audio(label="Audio with Custom Voice"), | |
], | |
#theme="huggingface", | |
description = "🤖 - 让有人文关怀的AI造福每一个人!AI向善,文明璀璨!TalktoAI - Enable the future!", | |
) | |
c2=gr.Interface( | |
fn=voice_conversion, | |
inputs=[ | |
gr.Textbox(lines=1, label = "请填写您的OpenAI-API-key"), | |
gr.Audio(source="microphone", label = "请上传您喜欢的声音,并尽量避免噪音", type="filepath"), | |
gr.Audio(source="microphone", label = "和您的专属AI聊天吧!", type="filepath"), | |
gr.Radio(["TOEFL", "Therapist", "Alice"], label="TOEFL Examiner, Therapist Tina, or Assistant Alice?"), | |
], | |
outputs=[ | |
gr.Textbox(label="Speech to Text"), gr.Textbox(label="ChatGPT Output"), gr.Audio(label="Audio with Custom Voice"), | |
], | |
#theme="huggingface", | |
description = "🤖 - 让有人文关怀的AI造福每一个人!AI向善,文明璀璨!TalktoAI - Enable the future!", | |
) | |
demo = gr.TabbedInterface([c1, c2], ["wav文件上传", "麦克风上传"], title = '🥳💬💕 - TalktoAI,随时随地,谈天说地!') | |
demo.launch(debug = True) |