forked from mrq/ai-voice-cloning
859 lines
29 KiB
Python
Executable File
859 lines
29 KiB
Python
Executable File
import os
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if 'XDG_CACHE_HOME' not in os.environ:
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os.environ['XDG_CACHE_HOME'] = os.path.realpath(os.path.join(os.getcwd(), './models/'))
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if 'TORTOISE_MODELS_DIR' not in os.environ:
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os.environ['TORTOISE_MODELS_DIR'] = os.path.realpath(os.path.join(os.getcwd(), './models/tortoise/'))
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if 'TRANSFORMERS_CACHE' not in os.environ:
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os.environ['TRANSFORMERS_CACHE'] = os.path.realpath(os.path.join(os.getcwd(), './models/transformers/'))
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import argparse
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import time
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import json
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import base64
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import re
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import urllib.request
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import signal
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import tqdm
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import torch
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import torchaudio
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import music_tag
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import gradio as gr
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import gradio.utils
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from datetime import datetime
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from tortoise.api import TextToSpeech, MODELS, get_model_path
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from tortoise.utils.audio import load_audio, load_voice, load_voices, get_voice_dir
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from tortoise.utils.text import split_and_recombine_text
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from tortoise.utils.device import get_device_name, set_device_name
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import whisper
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MODELS['dvae.pth'] = "https://huggingface.co/jbetker/tortoise-tts-v2/resolve/3704aea61678e7e468a06d8eea121dba368a798e/.models/dvae.pth"
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args = None
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tts = None
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webui = None
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voicefixer = None
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whisper_model = None
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def get_args():
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global args
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return args
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def setup_args():
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global args
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default_arguments = {
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'share': False,
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'listen': None,
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'check-for-updates': False,
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'models-from-local-only': False,
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'low-vram': False,
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'sample-batch-size': None,
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'embed-output-metadata': True,
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'latents-lean-and-mean': True,
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'voice-fixer': False, # getting tired of long initialization times in a Colab for downloading a large dataset for it
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'voice-fixer-use-cuda': True,
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'force-cpu-for-conditioning-latents': False,
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'defer-tts-load': False,
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'device-override': None,
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'whisper-model': "base",
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'autoregressive-model': None,
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'concurrency-count': 2,
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'output-sample-rate': 44100,
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'output-volume': 1,
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}
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if os.path.isfile('./config/exec.json'):
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with open(f'./config/exec.json', 'r', encoding="utf-8") as f:
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overrides = json.load(f)
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for k in overrides:
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default_arguments[k] = overrides[k]
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parser = argparse.ArgumentParser()
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parser.add_argument("--share", action='store_true', default=default_arguments['share'], help="Lets Gradio return a public URL to use anywhere")
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parser.add_argument("--listen", default=default_arguments['listen'], help="Path for Gradio to listen on")
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parser.add_argument("--check-for-updates", action='store_true', default=default_arguments['check-for-updates'], help="Checks for update on startup")
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parser.add_argument("--models-from-local-only", action='store_true', default=default_arguments['models-from-local-only'], help="Only loads models from disk, does not check for updates for models")
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parser.add_argument("--low-vram", action='store_true', default=default_arguments['low-vram'], help="Disables some optimizations that increases VRAM usage")
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parser.add_argument("--no-embed-output-metadata", action='store_false', default=not default_arguments['embed-output-metadata'], help="Disables embedding output metadata into resulting WAV files for easily fetching its settings used with the web UI (data is stored in the lyrics metadata tag)")
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parser.add_argument("--latents-lean-and-mean", action='store_true', default=default_arguments['latents-lean-and-mean'], help="Exports the bare essentials for latents.")
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parser.add_argument("--voice-fixer", action='store_true', default=default_arguments['voice-fixer'], help="Uses python module 'voicefixer' to improve audio quality, if available.")
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parser.add_argument("--voice-fixer-use-cuda", action='store_true', default=default_arguments['voice-fixer-use-cuda'], help="Hints to voicefixer to use CUDA, if available.")
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parser.add_argument("--force-cpu-for-conditioning-latents", default=default_arguments['force-cpu-for-conditioning-latents'], action='store_true', help="Forces computing conditional latents to be done on the CPU (if you constantyl OOM on low chunk counts)")
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parser.add_argument("--defer-tts-load", default=default_arguments['defer-tts-load'], action='store_true', help="Defers loading TTS model")
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parser.add_argument("--device-override", default=default_arguments['device-override'], help="A device string to override pass through Torch")
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parser.add_argument("--whisper-model", default=default_arguments['whisper-model'], help="Specifies which whisper model to use for transcription.")
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parser.add_argument("--autoregressive-model", default=default_arguments['autoregressive-model'], help="Specifies which autoregressive model to use for sampling.")
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parser.add_argument("--sample-batch-size", default=default_arguments['sample-batch-size'], type=int, help="Sets how many batches to use during the autoregressive samples pass")
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parser.add_argument("--concurrency-count", type=int, default=default_arguments['concurrency-count'], help="How many Gradio events to process at once")
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parser.add_argument("--output-sample-rate", type=int, default=default_arguments['output-sample-rate'], help="Sample rate to resample the output to (from 24KHz)")
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parser.add_argument("--output-volume", type=float, default=default_arguments['output-volume'], help="Adjusts volume of output")
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parser.add_argument("--os", default="unix", help="Specifies which OS, easily")
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args = parser.parse_args()
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args.embed_output_metadata = not args.no_embed_output_metadata
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set_device_name(args.device_override)
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args.listen_host = None
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args.listen_port = None
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args.listen_path = None
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if args.listen:
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try:
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match = re.findall(r"^(?:(.+?):(\d+))?(\/.+?)?$", args.listen)[0]
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args.listen_host = match[0] if match[0] != "" else "127.0.0.1"
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args.listen_port = match[1] if match[1] != "" else None
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args.listen_path = match[2] if match[2] != "" else "/"
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except Exception as e:
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pass
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if args.listen_port is not None:
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args.listen_port = int(args.listen_port)
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return args
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def pad(num, zeroes):
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s = ""
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for i in range(zeroes,0,-1):
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if num < 10 ** i:
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s = f"{s}0"
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return f"{s}{num}"
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def generate(
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text,
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delimiter,
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emotion,
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prompt,
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voice,
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mic_audio,
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voice_latents_chunks,
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seed,
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candidates,
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num_autoregressive_samples,
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diffusion_iterations,
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temperature,
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diffusion_sampler,
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breathing_room,
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cvvp_weight,
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top_p,
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diffusion_temperature,
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length_penalty,
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repetition_penalty,
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cond_free_k,
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experimental_checkboxes,
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progress=None
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):
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global args
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global tts
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if not tts:
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raise Exception("TTS is uninitialized or still initializing...")
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if voice != "microphone":
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voices = [voice]
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else:
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voices = []
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if voice == "microphone":
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if mic_audio is None:
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raise Exception("Please provide audio from mic when choosing `microphone` as a voice input")
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mic = load_audio(mic_audio, tts.input_sample_rate)
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voice_samples, conditioning_latents = [mic], None
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elif voice == "random":
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voice_samples, conditioning_latents = None, tts.get_random_conditioning_latents()
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else:
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progress(0, desc="Loading voice...")
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voice_samples, conditioning_latents = load_voice(voice)
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if voice_samples is not None and len(voice_samples) > 0:
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sample_voice = torch.cat(voice_samples, dim=-1).squeeze().cpu()
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conditioning_latents = tts.get_conditioning_latents(voice_samples, return_mels=not args.latents_lean_and_mean, progress=progress, slices=voice_latents_chunks, force_cpu=args.force_cpu_for_conditioning_latents)
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if len(conditioning_latents) == 4:
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conditioning_latents = (conditioning_latents[0], conditioning_latents[1], conditioning_latents[2], None)
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if voice != "microphone":
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torch.save(conditioning_latents, f'{get_voice_dir()}/{voice}/cond_latents.pth')
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voice_samples = None
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else:
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if conditioning_latents is not None:
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sample_voice, _ = load_voice(voice, load_latents=False)
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sample_voice = torch.cat(sample_voice, dim=-1).squeeze().cpu()
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else:
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sample_voice = None
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if seed == 0:
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seed = None
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if conditioning_latents is not None and len(conditioning_latents) == 2 and cvvp_weight > 0:
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print("Requesting weighing against CVVP weight, but voice latents are missing some extra data. Please regenerate your voice latents.")
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cvvp_weight = 0
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settings = {
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'temperature': float(temperature),
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'top_p': float(top_p),
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'diffusion_temperature': float(diffusion_temperature),
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'length_penalty': float(length_penalty),
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'repetition_penalty': float(repetition_penalty),
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'cond_free_k': float(cond_free_k),
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'num_autoregressive_samples': num_autoregressive_samples,
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'sample_batch_size': args.sample_batch_size,
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'diffusion_iterations': diffusion_iterations,
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'voice_samples': voice_samples,
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'conditioning_latents': conditioning_latents,
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'use_deterministic_seed': seed,
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'return_deterministic_state': True,
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'k': candidates,
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'diffusion_sampler': diffusion_sampler,
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'breathing_room': breathing_room,
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'progress': progress,
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'half_p': "Half Precision" in experimental_checkboxes,
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'cond_free': "Conditioning-Free" in experimental_checkboxes,
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'cvvp_amount': cvvp_weight,
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}
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if delimiter == "\\n":
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delimiter = "\n"
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if delimiter != "" and delimiter in text:
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texts = text.split(delimiter)
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else:
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texts = split_and_recombine_text(text)
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full_start_time = time.time()
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outdir = f"./results/{voice}/"
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os.makedirs(outdir, exist_ok=True)
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audio_cache = {}
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resample = None
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# not a ternary in the event for some reason I want to rely on librosa's upsampling interpolator rather than torchaudio's, for some reason
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if tts.output_sample_rate != args.output_sample_rate:
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resampler = torchaudio.transforms.Resample(
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tts.output_sample_rate,
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args.output_sample_rate,
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lowpass_filter_width=16,
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rolloff=0.85,
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resampling_method="kaiser_window",
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beta=8.555504641634386,
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)
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volume_adjust = torchaudio.transforms.Vol(gain=args.output_volume, gain_type="amplitude") if args.output_volume != 1 else None
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idx = 0
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idx_cache = {}
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for i, file in enumerate(os.listdir(outdir)):
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filename = os.path.basename(file)
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extension = os.path.splitext(filename)[1]
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if extension != ".json" and extension != ".wav":
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continue
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match = re.findall(rf"^{voice}_(\d+)(?:.+?)?{extension}$", filename)
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key = int(match[0])
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idx_cache[key] = True
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if len(idx_cache) > 0:
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keys = sorted(list(idx_cache.keys()))
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idx = keys[-1] + 1
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# I know there's something to pad I don't care
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idx = pad(idx, 4)
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def get_name(line=0, candidate=0, combined=False):
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name = f"{idx}"
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if combined:
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name = f"{name}_combined"
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elif len(texts) > 1:
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name = f"{name}_{line}"
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if candidates > 1:
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name = f"{name}_{candidate}"
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return name
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for line, cut_text in enumerate(texts):
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if emotion == "Custom":
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if prompt.strip() != "":
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cut_text = f"[{prompt},] {cut_text}"
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else:
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cut_text = f"[I am really {emotion.lower()},] {cut_text}"
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progress.msg_prefix = f'[{str(line+1)}/{str(len(texts))}]'
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print(f"{progress.msg_prefix} Generating line: {cut_text}")
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start_time = time.time()
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gen, additionals = tts.tts(cut_text, **settings )
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seed = additionals[0]
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run_time = time.time()-start_time
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print(f"Generating line took {run_time} seconds")
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if not isinstance(gen, list):
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gen = [gen]
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for j, g in enumerate(gen):
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audio = g.squeeze(0).cpu()
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name = get_name(line=line, candidate=j)
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audio_cache[name] = {
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'audio': audio,
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'text': cut_text,
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'time': run_time
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}
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# save here in case some error happens mid-batch
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torchaudio.save(f'{outdir}/{voice}_{name}.wav', audio, tts.output_sample_rate)
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for k in audio_cache:
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audio = audio_cache[k]['audio']
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if resampler is not None:
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audio = resampler(audio)
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if volume_adjust is not None:
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audio = volume_adjust(audio)
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audio_cache[k]['audio'] = audio
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torchaudio.save(f'{outdir}/{voice}_{k}.wav', audio, args.output_sample_rate)
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output_voices = []
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for candidate in range(candidates):
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if len(texts) > 1:
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audio_clips = []
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for line in range(len(texts)):
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name = get_name(line=line, candidate=candidate)
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audio = audio_cache[name]['audio']
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audio_clips.append(audio)
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name = get_name(candidate=candidate, combined=True)
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audio = torch.cat(audio_clips, dim=-1)
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torchaudio.save(f'{outdir}/{voice}_{name}.wav', audio, args.output_sample_rate)
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audio = audio.squeeze(0).cpu()
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audio_cache[name] = {
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'audio': audio,
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'text': text,
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'time': time.time()-full_start_time,
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'output': True
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}
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else:
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name = get_name(candidate=candidate)
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audio_cache[name]['output'] = True
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info = {
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'text': text,
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'delimiter': '\\n' if delimiter == "\n" else delimiter,
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'emotion': emotion,
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'prompt': prompt,
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'voice': voice,
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'seed': seed,
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'candidates': candidates,
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'num_autoregressive_samples': num_autoregressive_samples,
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'diffusion_iterations': diffusion_iterations,
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'temperature': temperature,
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'diffusion_sampler': diffusion_sampler,
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'breathing_room': breathing_room,
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'cvvp_weight': cvvp_weight,
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'top_p': top_p,
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'diffusion_temperature': diffusion_temperature,
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'length_penalty': length_penalty,
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'repetition_penalty': repetition_penalty,
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'cond_free_k': cond_free_k,
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'experimentals': experimental_checkboxes,
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'time': time.time()-full_start_time,
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}
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# kludgy yucky codesmells
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for name in audio_cache:
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if 'output' not in audio_cache[name]:
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continue
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output_voices.append(f'{outdir}/{voice}_{name}.wav')
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with open(f'{outdir}/{voice}_{name}.json', 'w', encoding="utf-8") as f:
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f.write(json.dumps(info, indent='\t') )
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if args.voice_fixer and voicefixer is not None:
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fixed_output_voices = []
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for path in progress.tqdm(output_voices, desc="Running voicefix..."):
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fixed = path.replace(".wav", "_fixed.wav")
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voicefixer.restore(
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input=path,
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output=fixed,
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cuda=get_device_name() == "cuda" and args.voice_fixer_use_cuda,
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#mode=mode,
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)
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fixed_output_voices.append(fixed)
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output_voices = fixed_output_voices
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if voice is not None and conditioning_latents is not None:
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with open(f'{get_voice_dir()}/{voice}/cond_latents.pth', 'rb') as f:
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info['latents'] = base64.b64encode(f.read()).decode("ascii")
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if args.embed_output_metadata:
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for name in progress.tqdm(audio_cache, desc="Embedding metadata..."):
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info['text'] = audio_cache[name]['text']
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info['time'] = audio_cache[name]['time']
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metadata = music_tag.load_file(f"{outdir}/{voice}_{name}.wav")
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metadata['lyrics'] = json.dumps(info)
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metadata.save()
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if sample_voice is not None:
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sample_voice = (tts.input_sample_rate, sample_voice.numpy())
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print(f"Generation took {info['time']} seconds, saved to '{output_voices[0]}'\n")
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info['seed'] = settings['use_deterministic_seed']
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if 'latents' in info:
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del info['latents']
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os.makedirs('./config/', exist_ok=True)
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with open(f'./config/generate.json', 'w', encoding="utf-8") as f:
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f.write(json.dumps(info, indent='\t') )
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stats = [
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[ seed, "{:.3f}".format(info['time']) ]
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]
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return (
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sample_voice,
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output_voices,
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stats,
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)
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import subprocess
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training_process = None
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def run_training(config_path):
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try:
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print("Unloading TTS to save VRAM.")
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global tts
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del tts
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tts = None
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except Exception as e:
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pass
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global training_process
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torch.multiprocessing.freeze_support()
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cmd = ['call' '.\\train.bat', config_path] if os.name == "nt" else ['bash', './train.sh', config_path]
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print("Spawning process: ", " ".join(cmd))
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training_process = subprocess.Popen(cmd, stdout=subprocess.PIPE, stderr=subprocess.STDOUT, universal_newlines=True)
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buffer=[]
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for line in iter(training_process.stdout.readline, ""):
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buffer.append(line)
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print(line[:-1])
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yield "".join(buffer)
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training_process.stdout.close()
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return_code = training_process.wait()
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training_process = None
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#if return_code:
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# raise subprocess.CalledProcessError(return_code, cmd)
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|
|
|
|
|
def stop_training():
|
|
if training_process is None:
|
|
return "No training in progress"
|
|
training_process.kill()
|
|
training_process = None
|
|
return "Training cancelled"
|
|
|
|
def setup_voicefixer(restart=False):
|
|
global voicefixer
|
|
if restart:
|
|
del voicefixer
|
|
voicefixer = None
|
|
|
|
try:
|
|
print("Initializating voice-fixer")
|
|
from voicefixer import VoiceFixer
|
|
voicefixer = VoiceFixer()
|
|
print("initialized voice-fixer")
|
|
except Exception as e:
|
|
print(f"Error occurred while tring to initialize voicefixer: {e}")
|
|
|
|
def setup_tortoise(restart=False):
|
|
global args
|
|
global tts
|
|
|
|
if args.voice_fixer and not restart:
|
|
setup_voicefixer(restart=restart)
|
|
|
|
if restart:
|
|
del tts
|
|
tts = None
|
|
|
|
print("Initializating TorToiSe...")
|
|
tts = TextToSpeech(minor_optimizations=not args.low_vram, autoregressive_model_path=args.autoregressive_model)
|
|
get_model_path('dvae.pth')
|
|
print("TorToiSe initialized, ready for generation.")
|
|
return tts
|
|
|
|
def save_training_settings( batch_size=None, learning_rate=None, print_rate=None, save_rate=None, name=None, dataset_name=None, dataset_path=None, validation_name=None, validation_path=None ):
|
|
settings = {
|
|
"batch_size": batch_size if batch_size else 128,
|
|
"learning_rate": learning_rate if learning_rate else 1e-5,
|
|
"print_rate": print_rate if print_rate else 50,
|
|
"save_rate": save_rate if save_rate else 50,
|
|
"name": name if name else "finetune",
|
|
"dataset_name": dataset_name if dataset_name else "finetune",
|
|
"dataset_path": dataset_path if dataset_path else "./training/finetune/train.txt",
|
|
"validation_name": validation_name if validation_name else "finetune",
|
|
"validation_path": validation_path if validation_path else "./training/finetune/train.txt",
|
|
}
|
|
outfile = f'./training/{settings["name"]}.yaml'
|
|
|
|
with open(f'./models/.template.yaml', 'r', encoding="utf-8") as f:
|
|
yaml = f.read()
|
|
|
|
for k in settings:
|
|
yaml = yaml.replace(f"${{{k}}}", str(settings[k]))
|
|
|
|
with open(outfile, 'w', encoding="utf-8") as f:
|
|
f.write(yaml)
|
|
|
|
return f"Training settings saved to: {outfile}"
|
|
|
|
def prepare_dataset( files, outdir, language=None, progress=None ):
|
|
global whisper_model
|
|
if whisper_model is None:
|
|
notify_progress(f"Loading Whisper model: {args.whisper_model}", progress)
|
|
whisper_model = whisper.load_model(args.whisper_model)
|
|
|
|
os.makedirs(outdir, exist_ok=True)
|
|
|
|
idx = 0
|
|
results = {}
|
|
transcription = []
|
|
|
|
for file in enumerate_progress(files, desc="Iterating through voice files", progress=progress):
|
|
print(f"Transcribing file: {file}")
|
|
|
|
result = whisper_model.transcribe(file, language=language if language else "English")
|
|
results[os.path.basename(file)] = result
|
|
|
|
print(f"Transcribed file: {file}, {len(result['segments'])} found.")
|
|
|
|
waveform, sampling_rate = torchaudio.load(file)
|
|
num_channels, num_frames = waveform.shape
|
|
|
|
for segment in result['segments']: # enumerate_progress(result['segments'], desc="Segmenting voice file", progress=progress):
|
|
start = int(segment['start'] * sampling_rate)
|
|
end = int(segment['end'] * sampling_rate)
|
|
|
|
sliced_waveform = waveform[:, start:end]
|
|
sliced_name = f"{pad(idx, 4)}.wav"
|
|
|
|
torchaudio.save(f"{outdir}/{sliced_name}", sliced_waveform, sampling_rate)
|
|
|
|
transcription.append(f"{sliced_name}|{segment['text'].strip()}")
|
|
idx = idx + 1
|
|
|
|
with open(f'{outdir}/whisper.json', 'w', encoding="utf-8") as f:
|
|
f.write(json.dumps(results, indent='\t'))
|
|
|
|
with open(f'{outdir}/train.txt', 'w', encoding="utf-8") as f:
|
|
f.write("\n".join(transcription))
|
|
|
|
return f"Processed dataset to: {outdir}"
|
|
|
|
def reset_generation_settings():
|
|
with open(f'./config/generate.json', 'w', encoding="utf-8") as f:
|
|
f.write(json.dumps({}, indent='\t') )
|
|
return import_generate_settings()
|
|
|
|
def import_voices(files, saveAs=None, progress=None):
|
|
global args
|
|
|
|
if not isinstance(files, list):
|
|
files = [files]
|
|
|
|
for file in enumerate_progress(files, desc="Importing voice files", progress=progress):
|
|
j, latents = read_generate_settings(file, read_latents=True)
|
|
|
|
if j is not None and saveAs is None:
|
|
saveAs = j['voice']
|
|
if saveAs is None or saveAs == "":
|
|
raise Exception("Specify a voice name")
|
|
|
|
outdir = f'{get_voice_dir()}/{saveAs}/'
|
|
os.makedirs(outdir, exist_ok=True)
|
|
|
|
if latents:
|
|
print(f"Importing latents to {latents}")
|
|
with open(f'{outdir}/cond_latents.pth', 'wb') as f:
|
|
f.write(latents)
|
|
latents = f'{outdir}/cond_latents.pth'
|
|
print(f"Imported latents to {latents}")
|
|
else:
|
|
filename = file.name
|
|
if filename[-4:] != ".wav":
|
|
raise Exception("Please convert to a WAV first")
|
|
|
|
path = f"{outdir}/{os.path.basename(filename)}"
|
|
print(f"Importing voice to {path}")
|
|
|
|
waveform, sampling_rate = torchaudio.load(filename)
|
|
|
|
if args.voice_fixer and voicefixer is not None:
|
|
# resample to best bandwidth since voicefixer will do it anyways through librosa
|
|
if sampling_rate != 44100:
|
|
print(f"Resampling imported voice sample: {path}")
|
|
resampler = torchaudio.transforms.Resample(
|
|
sampling_rate,
|
|
44100,
|
|
lowpass_filter_width=16,
|
|
rolloff=0.85,
|
|
resampling_method="kaiser_window",
|
|
beta=8.555504641634386,
|
|
)
|
|
waveform = resampler(waveform)
|
|
sampling_rate = 44100
|
|
|
|
torchaudio.save(path, waveform, sampling_rate)
|
|
|
|
print(f"Running 'voicefixer' on voice sample: {path}")
|
|
voicefixer.restore(
|
|
input = path,
|
|
output = path,
|
|
cuda=get_device_name() == "cuda" and args.voice_fixer_use_cuda,
|
|
#mode=mode,
|
|
)
|
|
else:
|
|
torchaudio.save(path, waveform, sampling_rate)
|
|
|
|
print(f"Imported voice to {path}")
|
|
|
|
def import_generate_settings(file="./config/generate.json"):
|
|
settings, _ = read_generate_settings(file, read_latents=False)
|
|
|
|
if settings is None:
|
|
return None
|
|
|
|
return (
|
|
None if 'text' not in settings else settings['text'],
|
|
None if 'delimiter' not in settings else settings['delimiter'],
|
|
None if 'emotion' not in settings else settings['emotion'],
|
|
None if 'prompt' not in settings else settings['prompt'],
|
|
None if 'voice' not in settings else settings['voice'],
|
|
None,
|
|
None,
|
|
None if 'seed' not in settings else settings['seed'],
|
|
None if 'candidates' not in settings else settings['candidates'],
|
|
None if 'num_autoregressive_samples' not in settings else settings['num_autoregressive_samples'],
|
|
None if 'diffusion_iterations' not in settings else settings['diffusion_iterations'],
|
|
0.8 if 'temperature' not in settings else settings['temperature'],
|
|
"DDIM" if 'diffusion_sampler' not in settings else settings['diffusion_sampler'],
|
|
8 if 'breathing_room' not in settings else settings['breathing_room'],
|
|
0.0 if 'cvvp_weight' not in settings else settings['cvvp_weight'],
|
|
0.8 if 'top_p' not in settings else settings['top_p'],
|
|
1.0 if 'diffusion_temperature' not in settings else settings['diffusion_temperature'],
|
|
1.0 if 'length_penalty' not in settings else settings['length_penalty'],
|
|
2.0 if 'repetition_penalty' not in settings else settings['repetition_penalty'],
|
|
2.0 if 'cond_free_k' not in settings else settings['cond_free_k'],
|
|
None if 'experimentals' not in settings else settings['experimentals'],
|
|
)
|
|
|
|
def curl(url):
|
|
try:
|
|
req = urllib.request.Request(url, headers={'User-Agent': 'Python'})
|
|
conn = urllib.request.urlopen(req)
|
|
data = conn.read()
|
|
data = data.decode()
|
|
data = json.loads(data)
|
|
conn.close()
|
|
return data
|
|
except Exception as e:
|
|
print(e)
|
|
return None
|
|
|
|
def check_for_updates():
|
|
if not os.path.isfile('./.git/FETCH_HEAD'):
|
|
print("Cannot check for updates: not from a git repo")
|
|
return False
|
|
|
|
with open(f'./.git/FETCH_HEAD', 'r', encoding="utf-8") as f:
|
|
head = f.read()
|
|
|
|
match = re.findall(r"^([a-f0-9]+).+?https:\/\/(.+?)\/(.+?)\/(.+?)\n", head)
|
|
if match is None or len(match) == 0:
|
|
print("Cannot check for updates: cannot parse FETCH_HEAD")
|
|
return False
|
|
|
|
match = match[0]
|
|
|
|
local = match[0]
|
|
host = match[1]
|
|
owner = match[2]
|
|
repo = match[3]
|
|
|
|
res = curl(f"https://{host}/api/v1/repos/{owner}/{repo}/branches/") #this only works for gitea instances
|
|
|
|
if res is None or len(res) == 0:
|
|
print("Cannot check for updates: cannot fetch from remote")
|
|
return False
|
|
|
|
remote = res[0]["commit"]["id"]
|
|
|
|
if remote != local:
|
|
print(f"New version found: {local[:8]} => {remote[:8]}")
|
|
return True
|
|
|
|
return False
|
|
|
|
def reload_tts():
|
|
setup_tortoise(restart=True)
|
|
|
|
def cancel_generate():
|
|
tortoise.api.STOP_SIGNAL = True
|
|
|
|
def get_voice_list(dir=get_voice_dir()):
|
|
os.makedirs(dir, exist_ok=True)
|
|
return sorted([d for d in os.listdir(dir) if os.path.isdir(os.path.join(dir, d)) and len(os.listdir(os.path.join(dir, d))) > 0 ]) + ["microphone", "random"]
|
|
|
|
def get_autoregressive_models(dir="./models/finetuned/"):
|
|
os.makedirs(dir, exist_ok=True)
|
|
return [get_model_path('autoregressive.pth')] + sorted([d for d in os.listdir(dir) if os.path.isdir(os.path.join(dir, d)) and len(os.listdir(os.path.join(dir, d))) > 0 ])
|
|
|
|
|
|
def update_autoregressive_model(path_name):
|
|
|
|
global tts
|
|
if not tts:
|
|
raise Exception("TTS is uninitialized or still initializing...")
|
|
|
|
print(f"Loading model: {path_name}")
|
|
if hasattr(tts, 'load_autoregressive_model') and tts.load_autoregressive_model(path_name):
|
|
args.autoregressive_model = path_name
|
|
save_args_settings()
|
|
# polyfill in case a user did NOT update the packages
|
|
else:
|
|
from tortoise.models.autoregressive import UnifiedVoice
|
|
|
|
previous_path = tts.autoregressive_model_path
|
|
tts.autoregressive_model_path = path_name if path_name and os.path.exists(path_name) else get_model_path('autoregressive.pth')
|
|
|
|
del tts.autoregressive
|
|
tts.autoregressive = UnifiedVoice(max_mel_tokens=604, max_text_tokens=402, max_conditioning_inputs=2, layers=30,
|
|
model_dim=1024,
|
|
heads=16, number_text_tokens=255, start_text_token=255, checkpointing=False,
|
|
train_solo_embeddings=False).cpu().eval()
|
|
tts.autoregressive.load_state_dict(torch.load(tts.autoregressive_model_path))
|
|
tts.autoregressive.post_init_gpt2_config(kv_cache=tts.use_kv_cache)
|
|
if tts.preloaded_tensors:
|
|
tts.autoregressive = tts.autoregressive.to(tts.device)
|
|
|
|
if previous_path != tts.autoregressive_model_path:
|
|
args.autoregressive_model = path_name
|
|
save_args_settings()
|
|
|
|
print(f"Loaded model: {tts.autoregressive_model_path}")
|
|
|
|
return path_name
|
|
|
|
def update_args( listen, share, check_for_updates, models_from_local_only, low_vram, embed_output_metadata, latents_lean_and_mean, voice_fixer, voice_fixer_use_cuda, force_cpu_for_conditioning_latents, defer_tts_load, device_override, sample_batch_size, concurrency_count, output_sample_rate, output_volume ):
|
|
global args
|
|
|
|
args.listen = listen
|
|
args.share = share
|
|
args.check_for_updates = check_for_updates
|
|
args.models_from_local_only = models_from_local_only
|
|
args.low_vram = low_vram
|
|
args.force_cpu_for_conditioning_latents = force_cpu_for_conditioning_latents
|
|
args.defer_tts_load = defer_tts_load
|
|
args.device_override = device_override
|
|
args.sample_batch_size = sample_batch_size
|
|
args.embed_output_metadata = embed_output_metadata
|
|
args.latents_lean_and_mean = latents_lean_and_mean
|
|
args.voice_fixer = voice_fixer
|
|
args.voice_fixer_use_cuda = voice_fixer_use_cuda
|
|
args.concurrency_count = concurrency_count
|
|
args.output_sample_rate = output_sample_rate
|
|
args.output_volume = output_volume
|
|
|
|
save_args_settings()
|
|
|
|
def save_args_settings():
|
|
settings = {
|
|
'listen': None if args.listen else args.listen,
|
|
'share': args.share,
|
|
'low-vram':args.low_vram,
|
|
'check-for-updates':args.check_for_updates,
|
|
'models-from-local-only':args.models_from_local_only,
|
|
'force-cpu-for-conditioning-latents': args.force_cpu_for_conditioning_latents,
|
|
'defer-tts-load': args.defer_tts_load,
|
|
'device-override': args.device_override,
|
|
'whisper-model': args.whisper_model,
|
|
'autoregressive-model': args.autoregressive_model,
|
|
'sample-batch-size': args.sample_batch_size,
|
|
'embed-output-metadata': args.embed_output_metadata,
|
|
'latents-lean-and-mean': args.latents_lean_and_mean,
|
|
'voice-fixer': args.voice_fixer,
|
|
'voice-fixer-use-cuda': args.voice_fixer_use_cuda,
|
|
'concurrency-count': args.concurrency_count,
|
|
'output-sample-rate': args.output_sample_rate,
|
|
'output-volume': args.output_volume,
|
|
}
|
|
|
|
os.makedirs('./config/', exist_ok=True)
|
|
with open(f'./config/exec.json', 'w', encoding="utf-8") as f:
|
|
f.write(json.dumps(settings, indent='\t') )
|
|
|
|
def read_generate_settings(file, read_latents=True, read_json=True):
|
|
j = None
|
|
latents = None
|
|
|
|
if file is not None:
|
|
if hasattr(file, 'name'):
|
|
file = file.name
|
|
|
|
if file[-4:] == ".wav":
|
|
metadata = music_tag.load_file(file)
|
|
if 'lyrics' in metadata:
|
|
j = json.loads(str(metadata['lyrics']))
|
|
elif file[-5:] == ".json":
|
|
with open(file, 'r') as f:
|
|
j = json.load(f)
|
|
|
|
if j is None:
|
|
print("No metadata found in audio file to read")
|
|
else:
|
|
if 'latents' in j:
|
|
if read_latents:
|
|
latents = base64.b64decode(j['latents'])
|
|
del j['latents']
|
|
|
|
|
|
if "time" in j:
|
|
j["time"] = "{:.3f}".format(j["time"])
|
|
|
|
return (
|
|
j,
|
|
latents,
|
|
)
|
|
|
|
def enumerate_progress(iterable, desc=None, progress=None, verbose=None):
|
|
if verbose and desc is not None:
|
|
print(desc)
|
|
|
|
if progress is None:
|
|
return tqdm(iterable, disable=not verbose)
|
|
return progress.tqdm(iterable, desc=f'{progress.msg_prefix} {desc}' if hasattr(progress, 'msg_prefix') else desc, track_tqdm=True)
|
|
|
|
def notify_progress(message, progress=None, verbose=True):
|
|
if verbose:
|
|
print(message)
|
|
|
|
if progress is None:
|
|
return
|
|
|
|
progress(0, desc=message) |