import os if 'XDG_CACHE_HOME' not in os.environ: os.environ['XDG_CACHE_HOME'] = os.path.realpath(os.path.join(os.getcwd(), './models/')) if 'TORTOISE_MODELS_DIR' not in os.environ: os.environ['TORTOISE_MODELS_DIR'] = os.path.realpath(os.path.join(os.getcwd(), './models/tortoise/')) if 'TRANSFORMERS_CACHE' not in os.environ: os.environ['TRANSFORMERS_CACHE'] = os.path.realpath(os.path.join(os.getcwd(), './models/transformers/')) import argparse import time import json import base64 import re import urllib.request import signal import tqdm import torch import torchaudio import music_tag import gradio as gr import gradio.utils from datetime import datetime from tortoise.api import TextToSpeech, MODELS, get_model_path from tortoise.utils.audio import load_audio, load_voice, load_voices, get_voice_dir from tortoise.utils.text import split_and_recombine_text from tortoise.utils.device import get_device_name, set_device_name import whisper MODELS['dvae.pth'] = "https://huggingface.co/jbetker/tortoise-tts-v2/resolve/3704aea61678e7e468a06d8eea121dba368a798e/.models/dvae.pth" args = None tts = None webui = None voicefixer = None whisper_model = None def get_args(): global args return args def setup_args(): global args default_arguments = { 'share': False, 'listen': None, 'check-for-updates': False, 'models-from-local-only': False, 'low-vram': False, 'sample-batch-size': None, 'embed-output-metadata': True, 'latents-lean-and-mean': True, 'voice-fixer': False, # getting tired of long initialization times in a Colab for downloading a large dataset for it 'voice-fixer-use-cuda': True, 'force-cpu-for-conditioning-latents': False, 'defer-tts-load': False, 'device-override': None, 'whisper-model': "base", 'autoregressive-model': None, 'concurrency-count': 2, 'output-sample-rate': 44100, 'output-volume': 1, } if os.path.isfile('./config/exec.json'): with open(f'./config/exec.json', 'r', encoding="utf-8") as f: overrides = json.load(f) for k in overrides: default_arguments[k] = overrides[k] parser = argparse.ArgumentParser() parser.add_argument("--share", action='store_true', default=default_arguments['share'], help="Lets Gradio return a public URL to use anywhere") parser.add_argument("--listen", default=default_arguments['listen'], help="Path for Gradio to listen on") parser.add_argument("--check-for-updates", action='store_true', default=default_arguments['check-for-updates'], help="Checks for update on startup") 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") parser.add_argument("--low-vram", action='store_true', default=default_arguments['low-vram'], help="Disables some optimizations that increases VRAM usage") 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)") parser.add_argument("--latents-lean-and-mean", action='store_true', default=default_arguments['latents-lean-and-mean'], help="Exports the bare essentials for latents.") parser.add_argument("--voice-fixer", action='store_true', default=default_arguments['voice-fixer'], help="Uses python module 'voicefixer' to improve audio quality, if available.") 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.") 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)") parser.add_argument("--defer-tts-load", default=default_arguments['defer-tts-load'], action='store_true', help="Defers loading TTS model") parser.add_argument("--device-override", default=default_arguments['device-override'], help="A device string to override pass through Torch") parser.add_argument("--whisper-model", default=default_arguments['whisper-model'], help="Specifies which whisper model to use for transcription.") parser.add_argument("--autoregressive-model", default=default_arguments['autoregressive-model'], help="Specifies which autoregressive model to use for sampling.") 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") parser.add_argument("--concurrency-count", type=int, default=default_arguments['concurrency-count'], help="How many Gradio events to process at once") parser.add_argument("--output-sample-rate", type=int, default=default_arguments['output-sample-rate'], help="Sample rate to resample the output to (from 24KHz)") parser.add_argument("--output-volume", type=float, default=default_arguments['output-volume'], help="Adjusts volume of output") parser.add_argument("--os", default="unix", help="Specifies which OS, easily") args = parser.parse_args() args.embed_output_metadata = not args.no_embed_output_metadata set_device_name(args.device_override) args.listen_host = None args.listen_port = None args.listen_path = None if args.listen: try: match = re.findall(r"^(?:(.+?):(\d+))?(\/.+?)?$", args.listen)[0] args.listen_host = match[0] if match[0] != "" else "127.0.0.1" args.listen_port = match[1] if match[1] != "" else None args.listen_path = match[2] if match[2] != "" else "/" except Exception as e: pass if args.listen_port is not None: args.listen_port = int(args.listen_port) return args def pad(num, zeroes): s = "" for i in range(zeroes,0,-1): if num < 10 ** i: s = f"{s}0" return f"{s}{num}" def generate( text, delimiter, emotion, prompt, voice, mic_audio, voice_latents_chunks, seed, candidates, num_autoregressive_samples, diffusion_iterations, temperature, diffusion_sampler, breathing_room, cvvp_weight, top_p, diffusion_temperature, length_penalty, repetition_penalty, cond_free_k, experimental_checkboxes, progress=None ): global args global tts if not tts: raise Exception("TTS is uninitialized or still initializing...") if voice != "microphone": voices = [voice] else: voices = [] if voice == "microphone": if mic_audio is None: raise Exception("Please provide audio from mic when choosing `microphone` as a voice input") mic = load_audio(mic_audio, tts.input_sample_rate) voice_samples, conditioning_latents = [mic], None elif voice == "random": voice_samples, conditioning_latents = None, tts.get_random_conditioning_latents() else: progress(0, desc="Loading voice...") voice_samples, conditioning_latents = load_voice(voice) if voice_samples and len(voice_samples) > 0: sample_voice = torch.cat(voice_samples, dim=-1).squeeze().cpu() 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) if len(conditioning_latents) == 4: conditioning_latents = (conditioning_latents[0], conditioning_latents[1], conditioning_latents[2], None) if voice != "microphone": torch.save(conditioning_latents, f'{get_voice_dir()}/{voice}/cond_latents.pth') voice_samples = None else: if conditioning_latents is not None: sample_voice, _ = load_voice(voice, load_latents=False) if sample_voice and len(sample_voice) > 0: sample_voice = torch.cat(sample_voice, dim=-1).squeeze().cpu() else: sample_voice = None if seed == 0: seed = None if conditioning_latents is not None and len(conditioning_latents) == 2 and cvvp_weight > 0: print("Requesting weighing against CVVP weight, but voice latents are missing some extra data. Please regenerate your voice latents.") cvvp_weight = 0 settings = { 'temperature': float(temperature), 'top_p': float(top_p), 'diffusion_temperature': float(diffusion_temperature), 'length_penalty': float(length_penalty), 'repetition_penalty': float(repetition_penalty), 'cond_free_k': float(cond_free_k), 'num_autoregressive_samples': num_autoregressive_samples, 'sample_batch_size': args.sample_batch_size, 'diffusion_iterations': diffusion_iterations, 'voice_samples': voice_samples, 'conditioning_latents': conditioning_latents, 'use_deterministic_seed': seed, 'return_deterministic_state': True, 'k': candidates, 'diffusion_sampler': diffusion_sampler, 'breathing_room': breathing_room, 'progress': progress, 'half_p': "Half Precision" in experimental_checkboxes, 'cond_free': "Conditioning-Free" in experimental_checkboxes, 'cvvp_amount': cvvp_weight, } if delimiter == "\\n": delimiter = "\n" if delimiter != "" and delimiter in text: texts = text.split(delimiter) else: texts = split_and_recombine_text(text) full_start_time = time.time() outdir = f"./results/{voice}/" os.makedirs(outdir, exist_ok=True) audio_cache = {} resample = None # 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 if tts.output_sample_rate != args.output_sample_rate: resampler = torchaudio.transforms.Resample( tts.output_sample_rate, args.output_sample_rate, lowpass_filter_width=16, rolloff=0.85, resampling_method="kaiser_window", beta=8.555504641634386, ) volume_adjust = torchaudio.transforms.Vol(gain=args.output_volume, gain_type="amplitude") if args.output_volume != 1 else None idx = 0 idx_cache = {} for i, file in enumerate(os.listdir(outdir)): filename = os.path.basename(file) extension = os.path.splitext(filename)[1] if extension != ".json" and extension != ".wav": continue match = re.findall(rf"^{voice}_(\d+)(?:.+?)?{extension}$", filename) key = int(match[0]) idx_cache[key] = True if len(idx_cache) > 0: keys = sorted(list(idx_cache.keys())) idx = keys[-1] + 1 # I know there's something to pad I don't care idx = pad(idx, 4) def get_name(line=0, candidate=0, combined=False): name = f"{idx}" if combined: name = f"{name}_combined" elif len(texts) > 1: name = f"{name}_{line}" if candidates > 1: name = f"{name}_{candidate}" return name for line, cut_text in enumerate(texts): if emotion == "Custom": if prompt.strip() != "": cut_text = f"[{prompt},] {cut_text}" else: cut_text = f"[I am really {emotion.lower()},] {cut_text}" progress.msg_prefix = f'[{str(line+1)}/{str(len(texts))}]' print(f"{progress.msg_prefix} Generating line: {cut_text}") start_time = time.time() gen, additionals = tts.tts(cut_text, **settings ) seed = additionals[0] run_time = time.time()-start_time print(f"Generating line took {run_time} seconds") if not isinstance(gen, list): gen = [gen] for j, g in enumerate(gen): audio = g.squeeze(0).cpu() name = get_name(line=line, candidate=j) audio_cache[name] = { 'audio': audio, 'text': cut_text, 'time': run_time } # save here in case some error happens mid-batch torchaudio.save(f'{outdir}/{voice}_{name}.wav', audio, tts.output_sample_rate) for k in audio_cache: audio = audio_cache[k]['audio'] if resampler is not None: audio = resampler(audio) if volume_adjust is not None: audio = volume_adjust(audio) audio_cache[k]['audio'] = audio torchaudio.save(f'{outdir}/{voice}_{k}.wav', audio, args.output_sample_rate) output_voices = [] for candidate in range(candidates): if len(texts) > 1: audio_clips = [] for line in range(len(texts)): name = get_name(line=line, candidate=candidate) audio = audio_cache[name]['audio'] audio_clips.append(audio) name = get_name(candidate=candidate, combined=True) audio = torch.cat(audio_clips, dim=-1) torchaudio.save(f'{outdir}/{voice}_{name}.wav', audio, args.output_sample_rate) audio = audio.squeeze(0).cpu() audio_cache[name] = { 'audio': audio, 'text': text, 'time': time.time()-full_start_time, 'output': True } else: name = get_name(candidate=candidate) audio_cache[name]['output'] = True info = { 'text': text, 'delimiter': '\\n' if delimiter == "\n" else delimiter, 'emotion': emotion, 'prompt': prompt, 'voice': voice, 'seed': seed, 'candidates': candidates, 'num_autoregressive_samples': num_autoregressive_samples, 'diffusion_iterations': diffusion_iterations, 'temperature': temperature, 'diffusion_sampler': diffusion_sampler, 'breathing_room': breathing_room, 'cvvp_weight': cvvp_weight, 'top_p': top_p, 'diffusion_temperature': diffusion_temperature, 'length_penalty': length_penalty, 'repetition_penalty': repetition_penalty, 'cond_free_k': cond_free_k, 'experimentals': experimental_checkboxes, 'time': time.time()-full_start_time, } # kludgy yucky codesmells for name in audio_cache: if 'output' not in audio_cache[name]: continue output_voices.append(f'{outdir}/{voice}_{name}.wav') with open(f'{outdir}/{voice}_{name}.json', 'w', encoding="utf-8") as f: f.write(json.dumps(info, indent='\t') ) if args.voice_fixer and voicefixer is not None: fixed_output_voices = [] for path in progress.tqdm(output_voices, desc="Running voicefix..."): fixed = path.replace(".wav", "_fixed.wav") voicefixer.restore( input=path, output=fixed, cuda=get_device_name() == "cuda" and args.voice_fixer_use_cuda, #mode=mode, ) fixed_output_voices.append(fixed) output_voices = fixed_output_voices if voice and voice != "random" and conditioning_latents is not None: with open(f'{get_voice_dir()}/{voice}/cond_latents.pth', 'rb') as f: info['latents'] = base64.b64encode(f.read()).decode("ascii") if args.embed_output_metadata: for name in progress.tqdm(audio_cache, desc="Embedding metadata..."): info['text'] = audio_cache[name]['text'] info['time'] = audio_cache[name]['time'] metadata = music_tag.load_file(f"{outdir}/{voice}_{name}.wav") metadata['lyrics'] = json.dumps(info) metadata.save() if sample_voice is not None: sample_voice = (tts.input_sample_rate, sample_voice.numpy()) print(f"Generation took {info['time']} seconds, saved to '{output_voices[0]}'\n") info['seed'] = settings['use_deterministic_seed'] if 'latents' in info: del info['latents'] os.makedirs('./config/', exist_ok=True) with open(f'./config/generate.json', 'w', encoding="utf-8") as f: f.write(json.dumps(info, indent='\t') ) stats = [ [ seed, "{:.3f}".format(info['time']) ] ] return ( sample_voice, output_voices, stats, ) import subprocess training_process = None def run_training(config_path): try: print("Unloading TTS to save VRAM.") global tts del tts tts = None except Exception as e: pass global training_process torch.multiprocessing.freeze_support() cmd = ['train.bat', config_path] if os.name == "nt" else ['bash', './train.sh', config_path] print("Spawning process: ", " ".join(cmd)) training_process = subprocess.Popen(cmd, stdout=subprocess.PIPE, stderr=subprocess.STDOUT, universal_newlines=True) buffer=[] for line in iter(training_process.stdout.readline, ""): buffer.append(f'[{datetime.now().isoformat()}] {line}') print(f"[Training] {line[:-1]}") yield "".join(buffer[-8:]) training_process.stdout.close() return_code = training_process.wait() training_process = None #if return_code: # raise subprocess.CalledProcessError(return_code, cmd) 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( iterations=None, 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, output_name=None ): settings = { "iterations": iterations if iterations else 500, "batch_size": batch_size if batch_size else 64, "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", } if not output_name: output_name = f'{settings["name"]}.yaml' outfile = f'./training/{output_name}' 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 get_dataset_list(dir="./training/"): 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 and "train.txt" in os.listdir(os.path.join(dir, d)) ]) def get_training_list(dir="./training/"): return sorted([f'./training/{d}/train.yaml' 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 and "train.yaml" in os.listdir(os.path.join(dir, d)) ]) 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)