ai-voice-cloning/src/utils.py

3958 lines
126 KiB
Python

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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 math
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import json
import base64
import re
import urllib.request
import signal
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import gc
import subprocess
import psutil
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import yaml
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import hashlib
import string
import random
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from tqdm import tqdm
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import torch
import torchaudio
import music_tag
import gradio as gr
import gradio.utils
import pandas as pd
import numpy as np
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from glob import glob
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from datetime import datetime
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from datetime import timedelta
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from tortoise.api import TextToSpeech as TorToise_TTS, MODELS, get_model_path, pad_or_truncate
from tortoise.utils.audio import load_audio, load_voice, load_voices, get_voice_dir, get_voices
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from tortoise.utils.text import split_and_recombine_text
from tortoise.utils.device import get_device_name, set_device_name, get_device_count, get_device_vram, get_device_batch_size, do_gc
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MODELS['dvae.pth'] = "https://huggingface.co/jbetker/tortoise-tts-v2/resolve/3704aea61678e7e468a06d8eea121dba368a798e/.models/dvae.pth"
WHISPER_MODELS = ["tiny", "base", "small", "medium", "large", "large-v1", "large-v2"]
WHISPER_SPECIALIZED_MODELS = ["tiny.en", "base.en", "small.en", "medium.en"]
WHISPER_BACKENDS = ["openai/whisper", "lightmare/whispercpp", "m-bain/whisperx"]
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VOCODERS = ['univnet', 'bigvgan_base_24khz_100band', 'bigvgan_24khz_100band']
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TTSES = ['tortoise']
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INFERENCING = False
GENERATE_SETTINGS_ARGS = None
LEARNING_RATE_SCHEMES = {"Multistep": "MultiStepLR", "Cos. Annealing": "CosineAnnealingLR_Restart"}
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LEARNING_RATE_SCHEDULE = [ 2, 4, 9, 18, 25, 33, 50 ]
RESAMPLERS = {}
MIN_TRAINING_DURATION = 0.6
MAX_TRAINING_DURATION = 11.6097505669
MAX_TRAINING_CHAR_LENGTH = 200
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VALLE_ENABLED = False
BARK_ENABLED = False
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VERBOSE_DEBUG = True
import traceback
try:
from whisper.normalizers.english import EnglishTextNormalizer
from whisper.normalizers.basic import BasicTextNormalizer
from whisper.tokenizer import LANGUAGES
print("Whisper detected")
except Exception as e:
if VERBOSE_DEBUG:
print(traceback.format_exc())
pass
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try:
from vall_e.emb.qnt import encode as valle_quantize
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from vall_e.emb.g2p import encode as valle_phonemize
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from vall_e.inference import TTS as VALLE_TTS
import soundfile
print("VALL-E detected")
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VALLE_ENABLED = True
except Exception as e:
if VERBOSE_DEBUG:
print(traceback.format_exc())
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pass
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if VALLE_ENABLED:
TTSES.append('vall-e')
# torchaudio.set_audio_backend('soundfile')
try:
import bark
from bark import text_to_semantic
from bark.generation import SAMPLE_RATE as BARK_SAMPLE_RATE, ALLOWED_PROMPTS, preload_models, codec_decode, generate_coarse, generate_fine, generate_text_semantic, load_codec_model
from bark.api import generate_audio as bark_generate_audio
from encodec.utils import convert_audio
from scipy.io.wavfile import write as write_wav
print("Bark detected")
BARK_ENABLED = True
except Exception as e:
if VERBOSE_DEBUG:
print(traceback.format_exc())
pass
if BARK_ENABLED:
TTSES.append('bark')
def semantic_to_audio_tokens(
semantic_tokens,
history_prompt = None,
temp = 0.7,
silent = False,
output_full = False,
):
coarse_tokens = generate_coarse(
semantic_tokens, history_prompt=history_prompt, temp=temp, silent=silent, use_kv_caching=True
)
fine_tokens = generate_fine(coarse_tokens, history_prompt=history_prompt, temp=0.5)
if output_full:
full_generation = {
"semantic_prompt": semantic_tokens,
"coarse_prompt": coarse_tokens,
"fine_prompt": fine_tokens,
}
return full_generation
return fine_tokens
class Bark_TTS():
def __init__(self, small=False):
self.input_sample_rate = BARK_SAMPLE_RATE
self.output_sample_rate = BARK_SAMPLE_RATE # args.output_sample_rate
preload_models(
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text_use_gpu=True,
coarse_use_gpu=True,
fine_use_gpu=True,
codec_use_gpu=True,
text_use_small=small,
coarse_use_small=small,
fine_use_small=small,
force_reload=False
)
self.device = get_device_name()
try:
from vocos import Vocos
self.vocos_enabled = True
print("Vocos detected")
except Exception as e:
if VERBOSE_DEBUG:
print(traceback.format_exc())
self.vocos_enabled = False
try:
from hubert.hubert_manager import HuBERTManager
hubert_manager = HuBERTManager()
hubert_manager.make_sure_hubert_installed()
hubert_manager.make_sure_tokenizer_installed()
self.hubert_enabled = True
print("HuBERT detected")
except Exception as e:
if VERBOSE_DEBUG:
print(traceback.format_exc())
self.hubert_enabled = False
if self.vocos_enabled:
self.vocos = Vocos.from_pretrained("charactr/vocos-encodec-24khz").to(self.device)
def create_voice( self, voice ):
transcription_json = f'./training/{voice}/whisper.json'
if not os.path.exists(transcription_json):
raise f"Transcription for voice not found: {voice}"
transcriptions = json.load(open(transcription_json, 'r', encoding="utf-8"))
candidates = []
for file in transcriptions:
result = transcriptions[file]
added = 0
for segment in result['segments']:
path = file.replace(".wav", f"_{pad(segment['id'], 4)}.wav")
# check if the slice actually exists
if not os.path.exists(f'./training/{voice}/audio/{path}'):
continue
entry = (
path,
segment['end'] - segment['start'],
segment['text']
)
candidates.append(entry)
added = added + 1
# if nothing got added (assuming because nothign was sliced), use the master file
if added == 0: # added < len(result['segments']):
start = 0
end = 0
for segment in result['segments']:
start = max( start, segment['start'] )
end = max( end, segment['end'] )
entry = (
file,
end - start,
result['text']
)
candidates.append(entry)
candidates.sort(key=lambda x: x[1])
candidate = random.choice(candidates)
audio_filepath = f'./training/{voice}/audio/{candidate[0]}'
text = candidate[-1]
print("Using as reference:", audio_filepath, text)
# Load and pre-process the audio waveform
model = load_codec_model(use_gpu=True)
wav, sr = torchaudio.load(audio_filepath)
wav = convert_audio(wav, sr, model.sample_rate, model.channels)
# generate semantic tokens
if self.hubert_enabled:
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from hubert.pre_kmeans_hubert import CustomHubert
from hubert.customtokenizer import CustomTokenizer
wav = wav.to(self.device)
# Extract discrete codes from EnCodec
with torch.no_grad():
encoded_frames = model.encode(wav.unsqueeze(0))
codes = torch.cat([encoded[0] for encoded in encoded_frames], dim=-1).squeeze() # [n_q, T]
# get seconds of audio
seconds = wav.shape[-1] / model.sample_rate
# Load the HuBERT model
hubert_model = CustomHubert(checkpoint_path='./data/models/hubert/hubert.pt').to(self.device)
# Load the CustomTokenizer model
tokenizer = CustomTokenizer.load_from_checkpoint('./data/models/hubert/tokenizer.pth').to(self.device)
semantic_vectors = hubert_model.forward(wav, input_sample_hz=model.sample_rate)
semantic_tokens = tokenizer.get_token(semantic_vectors)
# move codes to cpu
codes = codes.cpu().numpy()
# move semantic tokens to cpu
semantic_tokens = semantic_tokens.cpu().numpy()
else:
wav = wav.unsqueeze(0).to(self.device)
# Extract discrete codes from EnCodec
with torch.no_grad():
encoded_frames = model.encode(wav)
codes = torch.cat([encoded[0] for encoded in encoded_frames], dim=-1).squeeze().cpu().numpy() # [n_q, T]
# get seconds of audio
seconds = wav.shape[-1] / model.sample_rate
# generate semantic tokens
semantic_tokens = generate_text_semantic(text, max_gen_duration_s=seconds, top_k=50, top_p=.95, temp=0.7)
# print(bark.__file__)
bark_location = os.path.dirname(os.path.relpath(bark.__file__)) # './modules/bark/bark/'
output_path = f'./{bark_location}/assets/prompts/' + voice.replace("/", "_") + '.npz'
np.savez(output_path, fine_prompt=codes, coarse_prompt=codes[:2, :], semantic_prompt=semantic_tokens)
def inference( self, text, voice, text_temp=0.7, waveform_temp=0.7 ):
if voice == "random":
voice = None
else:
if not os.path.exists('./modules/bark/bark/assets/prompts/' + voice + '.npz'):
self.create_voice( voice )
voice = voice.replace("/", "_")
if voice not in ALLOWED_PROMPTS:
ALLOWED_PROMPTS.add( voice )
semantic_tokens = text_to_semantic(text, history_prompt=voice, temp=text_temp, silent=False)
audio_tokens = semantic_to_audio_tokens( semantic_tokens, history_prompt=voice, temp=waveform_temp, silent=False, output_full=False )
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if self.vocos_enabled:
audio_tokens_torch = torch.from_numpy(audio_tokens).to(self.device)
features = self.vocos.codes_to_features(audio_tokens_torch)
wav = self.vocos.decode(features, bandwidth_id=torch.tensor([2], device=self.device))
else:
wav = codec_decode( audio_tokens )
return ( wav, BARK_SAMPLE_RATE )
# return (bark_generate_audio(text, history_prompt=voice, text_temp=text_temp, waveform_temp=waveform_temp), BARK_SAMPLE_RATE)
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args = None
tts = None
tts_loading = False
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webui = None
voicefixer = None
whisper_model = None
whisper_align_model = None
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training_state = None
current_voice = None
def cleanup_voice_name( name ):
return name.split("/")[-1]
def resample( waveform, input_rate, output_rate=44100 ):
# mono-ize
waveform = torch.mean(waveform, dim=0, keepdim=True)
if input_rate == output_rate:
return waveform, output_rate
key = f'{input_rate}:{output_rate}'
if not key in RESAMPLERS:
RESAMPLERS[key] = torchaudio.transforms.Resample(
input_rate,
output_rate,
lowpass_filter_width=16,
rolloff=0.85,
resampling_method="kaiser_window",
beta=8.555504641634386,
)
return RESAMPLERS[key]( waveform ), output_rate
def generate(**kwargs):
if args.tts_backend == "tortoise":
return generate_tortoise(**kwargs)
if args.tts_backend == "vall-e":
return generate_valle(**kwargs)
if args.tts_backend == "bark":
return generate_bark(**kwargs)
def generate_bark(**kwargs):
parameters = {}
parameters.update(kwargs)
voice = parameters['voice']
progress = parameters['progress'] if 'progress' in parameters else None
if parameters['seed'] == 0:
parameters['seed'] = None
usedSeed = parameters['seed']
global args
global tts
unload_whisper()
unload_voicefixer()
if not tts:
# should check if it's loading or unloaded, and load it if it's unloaded
if tts_loading:
raise Exception("TTS is still initializing...")
if progress is not None:
notify_progress("Initializing TTS...", progress=progress)
load_tts()
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
do_gc()
voice_samples = None
conditioning_latents = None
sample_voice = None
voice_cache = {}
def get_settings( override=None ):
settings = {
'voice': parameters['voice'],
'text_temp': float(parameters['temperature']),
'waveform_temp': float(parameters['temperature']),
}
# could be better to just do a ternary on everything above, but i am not a professional
selected_voice = voice
if override is not None:
if 'voice' in override:
selected_voice = override['voice']
for k in override:
if k not in settings:
continue
settings[k] = override[k]
return settings
if not parameters['delimiter']:
parameters['delimiter'] = "\n"
elif parameters['delimiter'] == "\\n":
parameters['delimiter'] = "\n"
if parameters['delimiter'] and parameters['delimiter'] != "" and parameters['delimiter'] in parameters['text']:
texts = parameters['text'].split(parameters['delimiter'])
else:
texts = split_and_recombine_text(parameters['text'])
full_start_time = time.time()
outdir = f"{args.results_folder}/{voice}/"
os.makedirs(outdir, exist_ok=True)
audio_cache = {}
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][1:]
if extension != "json" and extension != "wav":
continue
match = re.findall(rf"^{cleanup_voice_name(voice)}_(\d+)(?:.+?)?{extension}$", filename)
if match and len(match) > 0:
key = int(match[0])
idx_cache[key] = True
if len(idx_cache) > 0:
keys = sorted(list(idx_cache.keys()))
idx = keys[-1] + 1
idx = pad(idx, 4)
def get_name(line=0, candidate=0, combined=False):
name = f"{idx}"
if combined:
name = f"{name}_combined"
elif len(texts) > 1:
name = f"{name}_{line}"
if parameters['candidates'] > 1:
name = f"{name}_{candidate}"
return name
def get_info( voice, settings = None, latents = True ):
info = {}
info.update(parameters)
info['time'] = time.time()-full_start_time
info['datetime'] = datetime.now().isoformat()
info['progress'] = None
del info['progress']
if info['delimiter'] == "\n":
info['delimiter'] = "\\n"
if settings is not None:
for k in settings:
if k in info:
info[k] = settings[k]
return info
INFERENCING = True
for line, cut_text in enumerate(texts):
tqdm_prefix = f'[{str(line+1)}/{str(len(texts))}]'
print(f"{tqdm_prefix} Generating line: {cut_text}")
start_time = time.time()
# do setting editing
match = re.findall(r'^(\{.+\}) (.+?)$', cut_text)
override = None
if match and len(match) > 0:
match = match[0]
try:
override = json.loads(match[0])
cut_text = match[1].strip()
except Exception as e:
raise Exception("Prompt settings editing requested, but received invalid JSON")
settings = get_settings( override=override )
gen = tts.inference(cut_text, **settings )
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):
wav, sr = g
name = get_name(line=line, candidate=j)
settings['text'] = cut_text
settings['time'] = run_time
settings['datetime'] = datetime.now().isoformat()
# save here in case some error happens mid-batch
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if tts.vocos_enabled:
torchaudio.save(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav', wav.cpu(), sr)
else:
write_wav(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav', sr, wav)
wav, sr = torchaudio.load(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
audio_cache[name] = {
'audio': wav,
'settings': get_info(voice=override['voice'] if override and 'voice' in override else voice, settings=settings)
}
del gen
do_gc()
INFERENCING = False
for k in audio_cache:
audio = audio_cache[k]['audio']
audio, _ = resample(audio, tts.output_sample_rate, args.output_sample_rate)
if volume_adjust is not None:
audio = volume_adjust(audio)
audio_cache[k]['audio'] = audio
torchaudio.save(f'{outdir}/{cleanup_voice_name(voice)}_{k}.wav', audio, args.output_sample_rate)
output_voices = []
for candidate in range(parameters['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}/{cleanup_voice_name(voice)}_{name}.wav', audio, args.output_sample_rate)
audio = audio.squeeze(0).cpu()
audio_cache[name] = {
'audio': audio,
'settings': get_info(voice=voice),
'output': True
}
else:
try:
name = get_name(candidate=candidate)
audio_cache[name]['output'] = True
except Exception as e:
for name in audio_cache:
audio_cache[name]['output'] = True
if args.voice_fixer:
if not voicefixer:
notify_progress("Loading voicefix...", progress=progress)
load_voicefixer()
try:
fixed_cache = {}
for name in tqdm(audio_cache, desc="Running voicefix..."):
del audio_cache[name]['audio']
if 'output' not in audio_cache[name] or not audio_cache[name]['output']:
continue
path = f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav'
fixed = f'{outdir}/{cleanup_voice_name(voice)}_{name}_fixed.wav'
voicefixer.restore(
input=path,
output=fixed,
cuda=get_device_name() == "cuda" and args.voice_fixer_use_cuda,
#mode=mode,
)
fixed_cache[f'{name}_fixed'] = {
'settings': audio_cache[name]['settings'],
'output': True
}
audio_cache[name]['output'] = False
for name in fixed_cache:
audio_cache[name] = fixed_cache[name]
except Exception as e:
print(e)
print("\nFailed to run Voicefixer")
for name in audio_cache:
if 'output' not in audio_cache[name] or not audio_cache[name]['output']:
if args.prune_nonfinal_outputs:
audio_cache[name]['pruned'] = True
os.remove(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
continue
output_voices.append(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
if not args.embed_output_metadata:
with open(f'{outdir}/{cleanup_voice_name(voice)}_{name}.json', 'w', encoding="utf-8") as f:
f.write(json.dumps(audio_cache[name]['settings'], indent='\t') )
if args.embed_output_metadata:
for name in tqdm(audio_cache, desc="Embedding metadata..."):
if 'pruned' in audio_cache[name] and audio_cache[name]['pruned']:
continue
metadata = music_tag.load_file(f"{outdir}/{cleanup_voice_name(voice)}_{name}.wav")
metadata['lyrics'] = json.dumps(audio_cache[name]['settings'])
metadata.save()
if sample_voice is not None:
sample_voice = (tts.input_sample_rate, sample_voice.numpy())
info = get_info(voice=voice, latents=False)
print(f"Generation took {info['time']} seconds, saved to '{output_voices[0]}'\n")
info['seed'] = usedSeed
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 = [
[ parameters['seed'], "{:.3f}".format(info['time']) ]
]
return (
sample_voice,
output_voices,
stats,
)
def generate_valle(**kwargs):
parameters = {}
parameters.update(kwargs)
voice = parameters['voice']
progress = parameters['progress'] if 'progress' in parameters else None
if parameters['seed'] == 0:
parameters['seed'] = None
usedSeed = parameters['seed']
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global args
global tts
unload_whisper()
unload_voicefixer()
if not tts:
# should check if it's loading or unloaded, and load it if it's unloaded
if tts_loading:
raise Exception("TTS is still initializing...")
if progress is not None:
notify_progress("Initializing TTS...", progress=progress)
load_tts()
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
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do_gc()
voice_samples = None
conditioning_latents = None
sample_voice = None
voice_cache = {}
def fetch_voice( voice ):
if voice in voice_cache:
return voice_cache[voice]
"""
voice_dir = f'./training/{voice}/audio/'
if not os.path.isdir(voice_dir) or len(os.listdir(voice_dir)) == 0:
voice_dir = f'./voices/{voice}/'
files = [ f'{voice_dir}/{d}' for d in os.listdir(voice_dir) if d[-4:] == ".wav" ]
"""
if os.path.isdir(f'./training/{voice}/audio/'):
files = get_voice(name="audio", dir=f"./training/{voice}/", load_latents=False)
else:
files = get_voice(name=voice, load_latents=False)
# return files
voice_cache[voice] = random.sample(files, k=min(3, len(files)))
return voice_cache[voice]
def get_settings( override=None ):
settings = {
'ar_temp': float(parameters['temperature']),
'nar_temp': float(parameters['temperature']),
'max_ar_steps': parameters['num_autoregressive_samples'],
}
# could be better to just do a ternary on everything above, but i am not a professional
selected_voice = voice
if override is not None:
if 'voice' in override:
selected_voice = override['voice']
for k in override:
if k not in settings:
continue
settings[k] = override[k]
settings['references'] = fetch_voice(voice=selected_voice) # [ fetch_voice(voice=selected_voice) for _ in range(3) ]
return settings
if not parameters['delimiter']:
parameters['delimiter'] = "\n"
elif parameters['delimiter'] == "\\n":
parameters['delimiter'] = "\n"
if parameters['delimiter'] and parameters['delimiter'] != "" and parameters['delimiter'] in parameters['text']:
texts = parameters['text'].split(parameters['delimiter'])
else:
texts = split_and_recombine_text(parameters['text'])
full_start_time = time.time()
outdir = f"{args.results_folder}/{voice}/"
os.makedirs(outdir, exist_ok=True)
audio_cache = {}
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][1:]
if extension != "json" and extension != "wav":
continue
match = re.findall(rf"^{voice}_(\d+)(?:.+?)?{extension}$", filename)
if match and len(match) > 0:
key = int(match[0])
idx_cache[key] = True
if len(idx_cache) > 0:
keys = sorted(list(idx_cache.keys()))
idx = keys[-1] + 1
idx = pad(idx, 4)
def get_name(line=0, candidate=0, combined=False):
name = f"{idx}"
if combined:
name = f"{name}_combined"
elif len(texts) > 1:
name = f"{name}_{line}"
if parameters['candidates'] > 1:
name = f"{name}_{candidate}"
return name
def get_info( voice, settings = None, latents = True ):
info = {}
info.update(parameters)
info['time'] = time.time()-full_start_time
info['datetime'] = datetime.now().isoformat()
info['progress'] = None
del info['progress']
if info['delimiter'] == "\n":
info['delimiter'] = "\\n"
if settings is not None:
for k in settings:
if k in info:
info[k] = settings[k]
return info
INFERENCING = True
for line, cut_text in enumerate(texts):
tqdm_prefix = f'[{str(line+1)}/{str(len(texts))}]'
print(f"{tqdm_prefix} Generating line: {cut_text}")
start_time = time.time()
# do setting editing
match = re.findall(r'^(\{.+\}) (.+?)$', cut_text)
override = None
if match and len(match) > 0:
match = match[0]
try:
override = json.loads(match[0])
cut_text = match[1].strip()
except Exception as e:
raise Exception("Prompt settings editing requested, but received invalid JSON")
name = get_name(line=line, candidate=0)
settings = get_settings( override=override )
references = settings['references']
settings.pop("references")
settings['out_path'] = f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav'
gen = tts.inference(cut_text, references, **settings )
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):
wav, sr = g
name = get_name(line=line, candidate=j)
settings['text'] = cut_text
settings['time'] = run_time
settings['datetime'] = datetime.now().isoformat()
# save here in case some error happens mid-batch
#torchaudio.save(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav', wav.cpu(), sr)
#soundfile.write(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav', wav.cpu()[0,0], sr)
wav, sr = torchaudio.load(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
audio_cache[name] = {
'audio': wav,
'settings': get_info(voice=override['voice'] if override and 'voice' in override else voice, settings=settings)
}
del gen
do_gc()
INFERENCING = False
for k in audio_cache:
audio = audio_cache[k]['audio']
audio, _ = resample(audio, tts.output_sample_rate, args.output_sample_rate)
if volume_adjust is not None:
audio = volume_adjust(audio)
audio_cache[k]['audio'] = audio
torchaudio.save(f'{outdir}/{cleanup_voice_name(voice)}_{k}.wav', audio, args.output_sample_rate)
output_voices = []
for candidate in range(parameters['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}/{cleanup_voice_name(voice)}_{name}.wav', audio, args.output_sample_rate)
audio = audio.squeeze(0).cpu()
audio_cache[name] = {
'audio': audio,
'settings': get_info(voice=voice),
'output': True
}
else:
name = get_name(candidate=candidate)
audio_cache[name]['output'] = True
if args.voice_fixer:
if not voicefixer:
notify_progress("Loading voicefix...", progress=progress)
load_voicefixer()
try:
fixed_cache = {}
for name in tqdm(audio_cache, desc="Running voicefix..."):
del audio_cache[name]['audio']
if 'output' not in audio_cache[name] or not audio_cache[name]['output']:
continue
path = f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav'
fixed = f'{outdir}/{cleanup_voice_name(voice)}_{name}_fixed.wav'
voicefixer.restore(
input=path,
output=fixed,
cuda=get_device_name() == "cuda" and args.voice_fixer_use_cuda,
#mode=mode,
)
fixed_cache[f'{name}_fixed'] = {
'settings': audio_cache[name]['settings'],
'output': True
}
audio_cache[name]['output'] = False
for name in fixed_cache:
audio_cache[name] = fixed_cache[name]
except Exception as e:
print(e)
print("\nFailed to run Voicefixer")
for name in audio_cache:
if 'output' not in audio_cache[name] or not audio_cache[name]['output']:
if args.prune_nonfinal_outputs:
audio_cache[name]['pruned'] = True
os.remove(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
continue
output_voices.append(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
if not args.embed_output_metadata:
with open(f'{outdir}/{cleanup_voice_name(voice)}_{name}.json', 'w', encoding="utf-8") as f:
f.write(json.dumps(audio_cache[name]['settings'], indent='\t') )
if args.embed_output_metadata:
for name in tqdm(audio_cache, desc="Embedding metadata..."):
if 'pruned' in audio_cache[name] and audio_cache[name]['pruned']:
continue
metadata = music_tag.load_file(f"{outdir}/{cleanup_voice_name(voice)}_{name}.wav")
metadata['lyrics'] = json.dumps(audio_cache[name]['settings'])
metadata.save()
if sample_voice is not None:
sample_voice = (tts.input_sample_rate, sample_voice.numpy())
info = get_info(voice=voice, latents=False)
print(f"Generation took {info['time']} seconds, saved to '{output_voices[0]}'\n")
info['seed'] = usedSeed
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 = [
[ parameters['seed'], "{:.3f}".format(info['time']) ]
]
return (
sample_voice,
output_voices,
stats,
)
def generate_tortoise(**kwargs):
parameters = {}
parameters.update(kwargs)
voice = parameters['voice']
progress = parameters['progress'] if 'progress' in parameters else None
if parameters['seed'] == 0:
parameters['seed'] = None
usedSeed = parameters['seed']
global args
global tts
unload_whisper()
unload_voicefixer()
if not tts:
# should check if it's loading or unloaded, and load it if it's unloaded
if tts_loading:
raise Exception("TTS is still initializing...")
load_tts()
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
do_gc()
voice_samples = None
conditioning_latents = None
sample_voice = None
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voice_cache = {}
def fetch_voice( voice ):
cache_key = f'{voice}:{tts.autoregressive_model_hash[:8]}'
if cache_key in voice_cache:
return voice_cache[cache_key]
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print(f"Loading voice: {voice} with model {tts.autoregressive_model_hash[:8]}")
sample_voice = None
if voice == "microphone":
if parameters['mic_audio'] is None:
raise Exception("Please provide audio from mic when choosing `microphone` as a voice input")
voice_samples, conditioning_latents = [load_audio(parameters['mic_audio'], tts.input_sample_rate)], None
elif voice == "random":
voice_samples, conditioning_latents = None, tts.get_random_conditioning_latents()
else:
if progress is not None:
notify_progress(f"Loading voice: {voice}", progress=progress)
voice_samples, conditioning_latents = load_voice(voice, model_hash=tts.autoregressive_model_hash)
if voice_samples and len(voice_samples) > 0:
if conditioning_latents is None:
conditioning_latents = compute_latents(voice=voice, voice_samples=voice_samples, voice_latents_chunks=parameters['voice_latents_chunks'])
sample_voice = torch.cat(voice_samples, dim=-1).squeeze().cpu()
voice_samples = None
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voice_cache[cache_key] = (voice_samples, conditioning_latents, sample_voice)
return voice_cache[cache_key]
def get_settings( override=None ):
settings = {
'temperature': float(parameters['temperature']),
'top_p': float(parameters['top_p']),
'diffusion_temperature': float(parameters['diffusion_temperature']),
'length_penalty': float(parameters['length_penalty']),
'repetition_penalty': float(parameters['repetition_penalty']),
'cond_free_k': float(parameters['cond_free_k']),
'num_autoregressive_samples': parameters['num_autoregressive_samples'],
'sample_batch_size': args.sample_batch_size,
'diffusion_iterations': parameters['diffusion_iterations'],
'voice_samples': None,
'conditioning_latents': None,
'use_deterministic_seed': parameters['seed'],
'return_deterministic_state': True,
'k': parameters['candidates'],
'diffusion_sampler': parameters['diffusion_sampler'],
'breathing_room': parameters['breathing_room'],
'half_p': "Half Precision" in parameters['experimentals'],
'cond_free': "Conditioning-Free" in parameters['experimentals'],
'cvvp_amount': parameters['cvvp_weight'],
'autoregressive_model': args.autoregressive_model,
'diffusion_model': args.diffusion_model,
'tokenizer_json': args.tokenizer_json,
}
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# could be better to just do a ternary on everything above, but i am not a professional
selected_voice = voice
if override is not None:
if 'voice' in override:
selected_voice = override['voice']
for k in override:
if k not in settings:
continue
settings[k] = override[k]
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if settings['autoregressive_model'] is not None:
if settings['autoregressive_model'] == "auto":
settings['autoregressive_model'] = deduce_autoregressive_model(selected_voice)
tts.load_autoregressive_model(settings['autoregressive_model'])
if settings['diffusion_model'] is not None:
if settings['diffusion_model'] == "auto":
settings['diffusion_model'] = deduce_diffusion_model(selected_voice)
tts.load_diffusion_model(settings['diffusion_model'])
if settings['tokenizer_json'] is not None:
tts.load_tokenizer_json(settings['tokenizer_json'])
settings['voice_samples'], settings['conditioning_latents'], _ = fetch_voice(voice=selected_voice)
# clamp it down for the insane users who want this
# it would be wiser to enforce the sample size to the batch size, but this is what the user wants
settings['sample_batch_size'] = args.sample_batch_size
if not settings['sample_batch_size']:
settings['sample_batch_size'] = tts.autoregressive_batch_size
if settings['num_autoregressive_samples'] < settings['sample_batch_size']:
settings['sample_batch_size'] = settings['num_autoregressive_samples']
if settings['conditioning_latents'] is not None and len(settings['conditioning_latents']) == 2 and settings['cvvp_amount'] > 0:
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print("Requesting weighing against CVVP weight, but voice latents are missing some extra data. Please regenerate your voice latents with 'Slimmer voice latents' unchecked.")
settings['cvvp_amount'] = 0
return settings
if not parameters['delimiter']:
parameters['delimiter'] = "\n"
elif parameters['delimiter'] == "\\n":
parameters['delimiter'] = "\n"
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if parameters['delimiter'] and parameters['delimiter'] != "" and parameters['delimiter'] in parameters['text']:
texts = parameters['text'].split(parameters['delimiter'])
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else:
texts = split_and_recombine_text(parameters['text'])
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full_start_time = time.time()
outdir = f"{args.results_folder}/{voice}/"
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os.makedirs(outdir, exist_ok=True)
audio_cache = {}
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][1:]
if extension != "json" and extension != "wav":
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continue
match = re.findall(rf"^{voice}_(\d+)(?:.+?)?{extension}$", filename)
if match and len(match) > 0:
key = int(match[0])
idx_cache[key] = True
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if len(idx_cache) > 0:
keys = sorted(list(idx_cache.keys()))
idx = keys[-1] + 1
idx = pad(idx, 4)
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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 parameters['candidates'] > 1:
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name = f"{name}_{candidate}"
return name
def get_info( voice, settings = None, latents = True ):
info = {}
info.update(parameters)
info['time'] = time.time()-full_start_time
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info['datetime'] = datetime.now().isoformat()
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info['model'] = tts.autoregressive_model_path
info['model_hash'] = tts.autoregressive_model_hash
info['progress'] = None
del info['progress']
if info['delimiter'] == "\n":
info['delimiter'] = "\\n"
if settings is not None:
for k in settings:
if k in info:
info[k] = settings[k]
if 'half_p' in settings and 'cond_free' in settings:
info['experimentals'] = []
if settings['half_p']:
info['experimentals'].append("Half Precision")
if settings['cond_free']:
info['experimentals'].append("Conditioning-Free")
if latents and "latents" not in info:
voice = info['voice']
model_hash = settings["model_hash"][:8] if settings is not None and "model_hash" in settings else tts.autoregressive_model_hash[:8]
dir = f'{get_voice_dir()}/{voice}/'
latents_path = f'{dir}/cond_latents_{model_hash}.pth'
if voice == "random" or voice == "microphone":
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if latents and settings is not None and settings['conditioning_latents']:
os.makedirs(dir, exist_ok=True)
torch.save(conditioning_latents, latents_path)
if latents_path and os.path.exists(latents_path):
try:
with open(latents_path, 'rb') as f:
info['latents'] = base64.b64encode(f.read()).decode("ascii")
except Exception as e:
pass
return info
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INFERENCING = True
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for line, cut_text in enumerate(texts):
if should_phonemize():
cut_text = phonemizer( cut_text )
if parameters['emotion'] == "Custom":
if parameters['prompt'] and parameters['prompt'].strip() != "":
cut_text = f"[{parameters['prompt']},] {cut_text}"
elif parameters['emotion'] != "None" and parameters['emotion']:
cut_text = f"[I am really {parameters['emotion'].lower()},] {cut_text}"
tqdm_prefix = f'[{str(line+1)}/{str(len(texts))}]'
print(f"{tqdm_prefix} Generating line: {cut_text}")
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start_time = time.time()
# do setting editing
match = re.findall(r'^(\{.+\}) (.+?)$', cut_text)
override = None
if match and len(match) > 0:
match = match[0]
try:
override = json.loads(match[0])
cut_text = match[1].strip()
except Exception as e:
raise Exception("Prompt settings editing requested, but received invalid JSON")
settings = get_settings( override=override )
gen, additionals = tts.tts(cut_text, **settings )
parameters['seed'] = additionals[0]
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run_time = time.time()-start_time
print(f"Generating line took {run_time} seconds")
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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)
settings['text'] = cut_text
settings['time'] = run_time
settings['datetime'] = datetime.now().isoformat()
if args.tts_backend == "tortoise":
settings['model'] = tts.autoregressive_model_path
settings['model_hash'] = tts.autoregressive_model_hash
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audio_cache[name] = {
'audio': audio,
'settings': get_info(voice=override['voice'] if override and 'voice' in override else voice, settings=settings)
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}
# save here in case some error happens mid-batch
torchaudio.save(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav', audio, tts.output_sample_rate)
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del gen
do_gc()
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INFERENCING = False
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for k in audio_cache:
audio = audio_cache[k]['audio']
audio, _ = resample(audio, tts.output_sample_rate, args.output_sample_rate)
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if volume_adjust is not None:
audio = volume_adjust(audio)
audio_cache[k]['audio'] = audio
torchaudio.save(f'{outdir}/{cleanup_voice_name(voice)}_{k}.wav', audio, args.output_sample_rate)
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output_voices = []
for candidate in range(parameters['candidates']):
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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}/{cleanup_voice_name(voice)}_{name}.wav', audio, args.output_sample_rate)
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audio = audio.squeeze(0).cpu()
audio_cache[name] = {
'audio': audio,
'settings': get_info(voice=voice),
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'output': True
}
else:
name = get_name(candidate=candidate)
audio_cache[name]['output'] = True
if args.voice_fixer:
if not voicefixer:
notify_progress("Loading voicefix...", progress=progress)
load_voicefixer()
try:
fixed_cache = {}
for name in tqdm(audio_cache, desc="Running voicefix..."):
del audio_cache[name]['audio']
if 'output' not in audio_cache[name] or not audio_cache[name]['output']:
continue
path = f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav'
fixed = f'{outdir}/{cleanup_voice_name(voice)}_{name}_fixed.wav'
voicefixer.restore(
input=path,
output=fixed,
cuda=get_device_name() == "cuda" and args.voice_fixer_use_cuda,
#mode=mode,
)
fixed_cache[f'{name}_fixed'] = {
'settings': audio_cache[name]['settings'],
'output': True
}
audio_cache[name]['output'] = False
for name in fixed_cache:
audio_cache[name] = fixed_cache[name]
except Exception as e:
print(e)
print("\nFailed to run Voicefixer")
for name in audio_cache:
if 'output' not in audio_cache[name] or not audio_cache[name]['output']:
if args.prune_nonfinal_outputs:
audio_cache[name]['pruned'] = True
os.remove(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
continue
output_voices.append(f'{outdir}/{cleanup_voice_name(voice)}_{name}.wav')
if not args.embed_output_metadata:
with open(f'{outdir}/{cleanup_voice_name(voice)}_{name}.json', 'w', encoding="utf-8") as f:
f.write(json.dumps(audio_cache[name]['settings'], indent='\t') )
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if args.embed_output_metadata:
for name in tqdm(audio_cache, desc="Embedding metadata..."):
if 'pruned' in audio_cache[name] and audio_cache[name]['pruned']:
continue
metadata = music_tag.load_file(f"{outdir}/{cleanup_voice_name(voice)}_{name}.wav")
metadata['lyrics'] = json.dumps(audio_cache[name]['settings'])
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metadata.save()
if sample_voice is not None:
sample_voice = (tts.input_sample_rate, sample_voice.numpy())
info = get_info(voice=voice, latents=False)
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print(f"Generation took {info['time']} seconds, saved to '{output_voices[0]}'\n")
info['seed'] = usedSeed
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if 'latents' in info:
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:
f.write(json.dumps(info, indent='\t') )
stats = [
[ parameters['seed'], "{:.3f}".format(info['time']) ]
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]
return (
sample_voice,
output_voices,
stats,
)
def cancel_generate():
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if not INFERENCING:
return
import tortoise.api
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tortoise.api.STOP_SIGNAL = True
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def hash_file(path, algo="md5", buffer_size=0):
hash = None
if algo == "md5":
hash = hashlib.md5()
elif algo == "sha1":
hash = hashlib.sha1()
else:
raise Exception(f'Unknown hash algorithm specified: {algo}')
if not os.path.exists(path):
raise Exception(f'Path not found: {path}')
with open(path, 'rb') as f:
if buffer_size > 0:
while True:
data = f.read(buffer_size)
if not data:
break
hash.update(data)
else:
hash.update(f.read())
return "{0}".format(hash.hexdigest())
def update_baseline_for_latents_chunks( voice ):
global current_voice
current_voice = voice
path = f'{get_voice_dir()}/{voice}/'
if not os.path.isdir(path):
return 1
dataset_file = f'./training/{voice}/train.txt'
if os.path.exists(dataset_file):
return 0 # 0 will leverage using the LJspeech dataset for computing latents
files = os.listdir(path)
total = 0
total_duration = 0
for file in files:
if file[-4:] != ".wav":
continue
metadata = torchaudio.info(f'{path}/{file}')
duration = metadata.num_frames / metadata.sample_rate
total_duration += duration
total = total + 1
# brain too fried to figure out a better way
if args.autocalculate_voice_chunk_duration_size == 0:
return int(total_duration / total) if total > 0 else 1
return int(total_duration / args.autocalculate_voice_chunk_duration_size) if total_duration > 0 else 1
def compute_latents(voice=None, voice_samples=None, voice_latents_chunks=0, original_ar=False, original_diffusion=False):
global tts
global args
unload_whisper()
unload_voicefixer()
if not tts:
if tts_loading:
raise Exception("TTS is still initializing...")
load_tts()
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
if args.tts_backend == "bark":
tts.create_voice( voice )
return
if args.autoregressive_model == "auto":
tts.load_autoregressive_model(deduce_autoregressive_model(voice))
if voice:
load_from_dataset = voice_latents_chunks == 0
if load_from_dataset:
dataset_path = f'./training/{voice}/train.txt'
if not os.path.exists(dataset_path):
load_from_dataset = False
else:
with open(dataset_path, 'r', encoding="utf-8") as f:
lines = f.readlines()
print("Leveraging dataset for computing latents")
voice_samples = []
max_length = 0
for line in lines:
filename = f'./training/{voice}/{line.split("|")[0]}'
waveform = load_audio(filename, 22050)
max_length = max(max_length, waveform.shape[-1])
voice_samples.append(waveform)
for i in range(len(voice_samples)):
voice_samples[i] = pad_or_truncate(voice_samples[i], max_length)
voice_latents_chunks = len(voice_samples)
if voice_latents_chunks == 0:
print("Dataset is empty!")
load_from_dataset = True
if not load_from_dataset:
voice_samples, _ = load_voice(voice, load_latents=False)
if voice_samples is None:
return
conditioning_latents = tts.get_conditioning_latents(voice_samples, return_mels=not args.latents_lean_and_mean, slices=voice_latents_chunks, force_cpu=args.force_cpu_for_conditioning_latents, original_ar=original_ar, original_diffusion=original_diffusion)
if len(conditioning_latents) == 4:
conditioning_latents = (conditioning_latents[0], conditioning_latents[1], conditioning_latents[2], None)
outfile = f'{get_voice_dir()}/{voice}/cond_latents_{tts.autoregressive_model_hash[:8]}.pth'
torch.save(conditioning_latents, outfile)
print(f'Saved voice latents: {outfile}')
return conditioning_latents
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# superfluous, but it cleans up some things
class TrainingState():
def __init__(self, config_path, keep_x_past_checkpoints=0, start=True):
self.killed = False
self.training_dir = os.path.dirname(config_path)
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with open(config_path, 'r') as file:
self.yaml_config = yaml.safe_load(file)
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self.json_config = json.load(open(f"{self.training_dir}/train.json", 'r', encoding="utf-8"))
self.dataset_path = f"{self.training_dir}/train.txt"
with open(self.dataset_path, 'r', encoding="utf-8") as f:
self.dataset_size = len(f.readlines())
self.batch_size = self.json_config["batch_size"]
self.save_rate = self.json_config["save_rate"]
self.epoch = 0
self.epochs = self.json_config["epochs"]
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self.it = 0
self.its = calc_iterations( self.epochs, self.dataset_size, self.batch_size )
self.step = 0
self.steps = int(self.its / self.dataset_size)
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self.checkpoint = 0
self.checkpoints = int((self.its - self.it) / self.save_rate)
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self.gpus = self.json_config['gpus']
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self.buffer = []
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self.open_state = False
self.training_started = False
self.info = {}
self.it_rate = ""
self.it_rates = 0
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self.epoch_rate = ""
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self.eta = "?"
self.eta_hhmmss = "?"
self.nan_detected = False
self.last_info_check_at = 0
self.statistics = {
'loss': [],
'lr': [],
'grad_norm': [],
}
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self.losses = []
self.metrics = {
'step': "",
'rate': "",
'loss': "",
}
self.loss_milestones = [ 1.0, 0.15, 0.05 ]
if args.tts_backend=="vall-e":
self.valle_last_it = 0
self.valle_steps = 0
if keep_x_past_checkpoints > 0:
self.cleanup_old(keep=keep_x_past_checkpoints)
if start:
self.spawn_process(config_path=config_path, gpus=self.gpus)
def spawn_process(self, config_path, gpus=1):
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if args.tts_backend == "vall-e":
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self.cmd = ['deepspeed', f'--num_gpus={gpus}', '--module', 'vall_e.train', f'yaml="{config_path}"']
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else:
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self.cmd = ['train.bat', config_path] if os.name == "nt" else ['./train.sh', config_path]
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print("Spawning process: ", " ".join(self.cmd))
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self.process = subprocess.Popen(self.cmd, stdin=subprocess.PIPE, stdout=subprocess.PIPE, stderr=subprocess.STDOUT, universal_newlines=True)
def parse_metrics(self, data):
if isinstance(data, str):
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if line.find('Training Metrics:') >= 0:
data = json.loads(line.split("Training Metrics:")[-1])
data['mode'] = "training"
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elif line.find('Validation Metrics:') >= 0:
data = json.loads(line.split("Validation Metrics:")[-1])
data['mode'] = "validation"
else:
return
self.info = data
if 'epoch' in self.info:
self.epoch = int(self.info['epoch'])
if 'it' in self.info:
self.it = int(self.info['it'])
if 'step' in self.info:
self.step = int(self.info['step'])
if 'steps' in self.info:
self.steps = int(self.info['steps'])
if 'elapsed_time' in self.info:
self.info['iteration_rate'] = self.info['elapsed_time']
del self.info['elapsed_time']
if 'iteration_rate' in self.info:
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it_rate = self.info['iteration_rate']
self.it_rate = f'{"{:.3f}".format(1/it_rate)}it/s' if 0 < it_rate and it_rate < 1 else f'{"{:.3f}".format(it_rate)}s/it'
self.it_rates += it_rate
if self.it_rates > 0 and self.it * self.steps > 0:
epoch_rate = self.it_rates / self.it * self.steps
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self.epoch_rate = f'{"{:.3f}".format(1/epoch_rate)}epoch/s' if 0 < epoch_rate and epoch_rate < 1 else f'{"{:.3f}".format(epoch_rate)}s/epoch'
try:
self.eta = (self.its - self.it) * (self.it_rates / self.it)
eta = str(timedelta(seconds=int(self.eta)))
self.eta_hhmmss = eta
except Exception as e:
self.eta_hhmmss = "?"
pass
self.metrics['step'] = [f"{self.epoch}/{self.epochs}"]
if self.epochs != self.its:
self.metrics['step'].append(f"{self.it}/{self.its}")
if self.steps > 1:
self.metrics['step'].append(f"{self.step}/{self.steps}")
self.metrics['step'] = ", ".join(self.metrics['step'])
if args.tts_backend == "tortoise":
epoch = self.epoch + (self.step / self.steps)
else:
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epoch = self.info['epoch'] if 'epoch' in self.info else self.it
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if self.it > 0:
# probably can double for-loop but whatever
keys = {
'lrs': ['lr'],
'losses': ['loss_text_ce', 'loss_mel_ce'],
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'accuracies': [],
'precisions': [],
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'grad_norms': [],
}
if args.tts_backend == "vall-e":
keys['lrs'] = [
'ar.lr', 'nar.lr',
]
keys['losses'] = [
# 'ar.loss', 'nar.loss', 'ar+nar.loss',
'ar.loss.nll', 'nar.loss.nll',
]
keys['accuracies'] = [
'ar.loss.acc', 'nar.loss.acc',
'ar.stats.acc', 'nar.loss.acc',
]
keys['precisions'] = [ 'ar.loss.precision', 'nar.loss.precision', ]
keys['grad_norms'] = ['ar.grad_norm', 'nar.grad_norm']
for k in keys['lrs']:
if k not in self.info:
continue
self.statistics['lr'].append({'epoch': epoch, 'it': self.it, 'value': self.info[k], 'type': k})
for k in keys['accuracies']:
if k not in self.info:
continue
self.statistics['loss'].append({'epoch': epoch, 'it': self.it, 'value': self.info[k], 'type': k})
for k in keys['precisions']:
if k not in self.info:
continue
self.statistics['loss'].append({'epoch': epoch, 'it': self.it, 'value': self.info[k], 'type': k})
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for k in keys['losses']:
if k not in self.info:
continue
prefix = ""
if "mode" in self.info and self.info["mode"] == "validation":
prefix = f'{self.info["name"] if "name" in self.info else "val"}_'
self.statistics['loss'].append({'epoch': epoch, 'it': self.it, 'value': self.info[k], 'type': f'{prefix}{k}' })
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self.losses.append( self.statistics['loss'][-1] )
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for k in keys['grad_norms']:
if k not in self.info:
continue
self.statistics['grad_norm'].append({'epoch': epoch, 'it': self.it, 'value': self.info[k], 'type': k})
return data
def get_status(self):
message = None
self.metrics['rate'] = []
if self.epoch_rate:
self.metrics['rate'].append(self.epoch_rate)
if self.it_rate and self.epoch_rate[:-7] != self.it_rate[:-4]:
self.metrics['rate'].append(self.it_rate)
self.metrics['rate'] = ", ".join(self.metrics['rate'])
eta_hhmmss = self.eta_hhmmss if self.eta_hhmmss else "?"
self.metrics['loss'] = []
if 'lr' in self.info:
self.metrics['loss'].append(f'LR: {"{:.3e}".format(self.info["lr"])}')
if len(self.losses) > 0:
self.metrics['loss'].append(f'Loss: {"{:.3f}".format(self.losses[-1]["value"])}')
if False and len(self.losses) >= 2:
deriv = 0
accum_length = len(self.losses)//2 # i *guess* this is fine when you think about it
loss_value = self.losses[-1]["value"]
for i in range(accum_length):
d1_loss = self.losses[accum_length-i-1]["value"]
d2_loss = self.losses[accum_length-i-2]["value"]
dloss = (d2_loss - d1_loss)
d1_step = self.losses[accum_length-i-1]["it"]
d2_step = self.losses[accum_length-i-2]["it"]
dstep = (d2_step - d1_step)
if dstep == 0:
continue
inst_deriv = dloss / dstep
deriv += inst_deriv
deriv = deriv / accum_length
print("Deriv: ", deriv)
if deriv != 0: # dloss < 0:
next_milestone = None
for milestone in self.loss_milestones:
if loss_value > milestone:
next_milestone = milestone
break
print(f"Loss value: {loss_value} | Next milestone: {next_milestone} | Distance: {loss_value - next_milestone}")
if next_milestone:
# tfw can do simple calculus but not basic algebra in my head
est_its = (next_milestone - loss_value) / deriv * 100
print(f"Estimated: {est_its}")
if est_its >= 0:
self.metrics['loss'].append(f'Est. milestone {next_milestone} in: {int(est_its)}its')
else:
est_loss = inst_deriv * (self.its - self.it) + loss_value
if est_loss >= 0:
self.metrics['loss'].append(f'Est. final loss: {"{:.3f}".format(est_loss)}')
self.metrics['loss'] = ", ".join(self.metrics['loss'])
message = f"[{self.metrics['step']}] [{self.metrics['rate']}] [ETA: {eta_hhmmss}] [{self.metrics['loss']}]"
if self.nan_detected:
message = f"[!NaN DETECTED! {self.nan_detected}] {message}"
return message
def load_statistics(self, update=False):
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if not os.path.isdir(self.training_dir):
return
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if args.tts_backend == "tortoise":
logs = sorted([f'{self.training_dir}/finetune/{d}' for d in os.listdir(f'{self.training_dir}/finetune/') if d[-4:] == ".log" ])
else:
log_dir = "logs"
logs = sorted([f'{self.training_dir}/{log_dir}/{d}/log.txt' for d in os.listdir(f'{self.training_dir}/{log_dir}/') ])
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if update:
logs = [logs[-1]]
infos = {}
highest_step = self.last_info_check_at
if not update:
self.statistics['loss'] = []
self.statistics['lr'] = []
self.statistics['grad_norm'] = []
self.it_rates = 0
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unq = {}
averager = None
prev_state = 0
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for log in logs:
with open(log, 'r', encoding="utf-8") as f:
lines = f.readlines()
for line in lines:
line = line.strip()
if not line:
continue
if line[-1] == ".":
line = line[:-1]
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if line.find('Training Metrics:') >= 0:
split = line.split("Training Metrics:")[-1]
data = json.loads(split)
name = "train"
mode = "training"
prev_state = 0
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elif line.find('Validation Metrics:') >= 0:
data = json.loads(line.split("Validation Metrics:")[-1])
if "it" not in data:
data['it'] = it
if "epoch" not in data:
data['epoch'] = epoch
# name = data['name'] if 'name' in data else "val"
mode = "validation"
if prev_state == 0:
name = "subtrain"
else:
name = "val"
prev_state += 1
else:
continue
if "it" not in data:
continue
it = data['it']
epoch = data['epoch']
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if args.tts_backend == "vall-e":
if not averager or averager['key'] != f'{it}_{name}' or averager['mode'] != mode:
averager = {
'key': f'{it}_{name}',
'name': name,
'mode': mode,
"metrics": {}
}
for k in data:
if data[k] is None:
continue
averager['metrics'][k] = [ data[k] ]
else:
for k in data:
if data[k] is None:
continue
if k not in averager['metrics']:
averager['metrics'][k] = [ data[k] ]
else:
averager['metrics'][k].append( data[k] )
unq[f'{it}_{mode}_{name}'] = averager
else:
unq[f'{it}_{mode}_{name}'] = data
if update and it <= self.last_info_check_at:
continue
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blacklist = [ "batch", "eval" ]
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for it in unq:
if args.tts_backend == "vall-e":
stats = unq[it]
data = {k: sum(v) / len(v) for k, v in stats['metrics'].items() if k not in blacklist }
#data = {k: min(v) for k, v in stats['metrics'].items() if k not in blacklist }
#data = {k: max(v) for k, v in stats['metrics'].items() if k not in blacklist }
data['name'] = stats['name']
data['mode'] = stats['mode']
data['steps'] = len(stats['metrics']['it'])
else:
data = unq[it]
self.parse_metrics(data)
self.last_info_check_at = highest_step
def cleanup_old(self, keep=2):
if keep <= 0:
return
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if args.tts_backend == "vall-e":
return
if not os.path.isdir(f'{self.training_dir}/finetune/'):
return
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models = sorted([ int(d[:-8]) for d in os.listdir(f'{self.training_dir}/finetune/models/') if d[-8:] == "_gpt.pth" ])
states = sorted([ int(d[:-6]) for d in os.listdir(f'{self.training_dir}/finetune/training_state/') if d[-6:] == ".state" ])
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remove_models = models[:-keep]
remove_states = states[:-keep]
for d in remove_models:
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path = f'{self.training_dir}/finetune/models/{d}_gpt.pth'
print("Removing", path)
os.remove(path)
for d in remove_states:
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path = f'{self.training_dir}/finetune/training_state/{d}.state'
print("Removing", path)
os.remove(path)
def parse(self, line, verbose=False, keep_x_past_checkpoints=0, buffer_size=8, progress=None ):
self.buffer.append(f'{line}')
data = None
percent = 0
message = None
should_return = False
MESSAGE_START = 'Start training from epoch'
MESSAGE_FINSIHED = 'Finished training'
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MESSAGE_SAVING = 'Saving models and training states.'
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MESSAGE_METRICS_TRAINING = 'Training Metrics:'
MESSAGE_METRICS_VALIDATION = 'Validation Metrics:'
if line.find(MESSAGE_FINSIHED) >= 0:
self.killed = True
# rip out iteration info
elif not self.training_started:
if line.find(MESSAGE_START) >= 0:
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self.training_started = True # could just leverage the above variable, but this is python, and there's no point in these aggressive microoptimizations
match = re.findall(r'epoch: ([\d,]+)', line)
if match and len(match) > 0:
self.epoch = int(match[0].replace(",", ""))
match = re.findall(r'iter: ([\d,]+)', line)
if match and len(match) > 0:
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self.it = int(match[0].replace(",", ""))
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self.checkpoints = int((self.its - self.it) / self.save_rate)
self.load_statistics()
should_return = True
else:
if line.find(MESSAGE_SAVING) >= 0:
self.checkpoint += 1
message = f"[{self.checkpoint}/{self.checkpoints}] Saving checkpoint..."
percent = self.checkpoint / self.checkpoints
self.cleanup_old(keep=keep_x_past_checkpoints)
elif line.find(MESSAGE_METRICS_TRAINING) >= 0:
data = json.loads(line.split(MESSAGE_METRICS_TRAINING)[-1])
data['mode'] = "training"
elif line.find(MESSAGE_METRICS_VALIDATION) >= 0:
data = json.loads(line.split(MESSAGE_METRICS_VALIDATION)[-1])
data['mode'] = "validation"
if data is not None:
if ': nan' in line and not self.nan_detected:
self.nan_detected = self.it
self.parse_metrics( data )
message = self.get_status()
if message:
percent = self.it / float(self.its) # self.epoch / float(self.epochs)
if progress is not None:
progress(percent, message)
self.buffer.append(f'[{"{:.3f}".format(percent*100)}%] {message}')
should_return = True
if verbose and not self.training_started:
should_return = True
self.buffer = self.buffer[-buffer_size:]
result = None
if should_return:
result = "".join(self.buffer) if not self.training_started else message
return (
result,
percent,
message,
)
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try:
import altair as alt
alt.data_transformers.enable('default', max_rows=None)
except Exception as e:
print(e)
pass
def run_training(config_path, verbose=False, keep_x_past_checkpoints=0, progress=gr.Progress(track_tqdm=True)):
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global training_state
if training_state and training_state.process:
return "Training already in progress"
# ensure we have the dvae.pth
if args.tts_backend == "tortoise":
get_model_path('dvae.pth')
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# I don't know if this is still necessary, as it was bitching at me for not doing this, despite it being in a separate process
torch.multiprocessing.freeze_support()
unload_tts()
unload_whisper()
unload_voicefixer()
training_state = TrainingState(config_path=config_path, keep_x_past_checkpoints=keep_x_past_checkpoints)
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for line in iter(training_state.process.stdout.readline, ""):
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if training_state is None or training_state.killed:
return
result, percent, message = training_state.parse( line=line, verbose=verbose, keep_x_past_checkpoints=keep_x_past_checkpoints, progress=progress )
print(f"[Training] [{datetime.now().isoformat()}] {line[:-1]}")
if result:
yield result
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if progress is not None and message:
progress(percent, message)
if training_state:
training_state.process.stdout.close()
return_code = training_state.process.wait()
training_state = None
def update_training_dataplot(x_min=None, x_max=None, y_min=None, y_max=None, config_path=None):
global training_state
losses = None
lrs = None
grad_norms = None
x_lim = [ x_min, x_max ]
y_lim = [ y_min, y_max ]
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if not training_state:
if config_path:
training_state = TrainingState(config_path=config_path, start=False)
training_state.load_statistics()
message = training_state.get_status()
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if training_state:
if not x_lim[-1]:
x_lim[-1] = training_state.epochs
if not y_lim[-1]:
y_lim = None
if len(training_state.statistics['loss']) > 0:
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losses = gr.LinePlot.update(
value = pd.DataFrame(training_state.statistics['loss']),
x_lim=x_lim, y_lim=y_lim,
x="epoch", y="value", # x="it",
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title="Loss Metrics", color="type", tooltip=['epoch', 'it', 'value', 'type'],
width=500, height=350
)
if len(training_state.statistics['lr']) > 0:
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lrs = gr.LinePlot.update(
value = pd.DataFrame(training_state.statistics['lr']),
x_lim=x_lim,
x="epoch", y="value", # x="it",
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title="Learning Rate", color="type", tooltip=['epoch', 'it', 'value', 'type'],
width=500, height=350
)
if len(training_state.statistics['grad_norm']) > 0:
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grad_norms = gr.LinePlot.update(
value = pd.DataFrame(training_state.statistics['grad_norm']),
x_lim=x_lim,
x="epoch", y="value", # x="it",
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title="Gradient Normals", color="type", tooltip=['epoch', 'it', 'value', 'type'],
width=500, height=350
)
if config_path:
del training_state
training_state = None
return (losses, lrs, grad_norms)
def reconnect_training(verbose=False, progress=gr.Progress(track_tqdm=True)):
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global training_state
if not training_state or not training_state.process:
return "Training not in progress"
for line in iter(training_state.process.stdout.readline, ""):
result, percent, message = training_state.parse( line=line, verbose=verbose, progress=progress )
print(f"[Training] [{datetime.now().isoformat()}] {line[:-1]}")
if result:
yield result
if progress is not None and message:
progress(percent, message)
def stop_training():
global training_state
if training_state is None:
return "No training in progress"
print("Killing training process...")
training_state.killed = True
children = []
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if args.tts_backend == "tortoise":
# wrapped in a try/catch in case for some reason this fails outside of Linux
try:
children = [p.info for p in psutil.process_iter(attrs=['pid', 'name', 'cmdline']) if './src/train.py' in p.info['cmdline']]
except Exception as e:
pass
training_state.process.stdout.close()
training_state.process.terminate()
training_state.process.kill()
elif args.tts_backend == "vall-e":
print(training_state.process.communicate(input='quit')[0])
return_code = training_state.process.wait()
for p in children:
os.kill( p['pid'], signal.SIGKILL )
training_state = None
print("Killed training process.")
return f"Training cancelled: {return_code}"
def get_halfp_model_path():
autoregressive_model_path = get_model_path('autoregressive.pth')
return autoregressive_model_path.replace(".pth", "_half.pth")
def convert_to_halfp():
autoregressive_model_path = get_model_path('autoregressive.pth')
print(f'Converting model to half precision: {autoregressive_model_path}')
model = torch.load(autoregressive_model_path)
for k in model:
model[k] = model[k].half()
outfile = get_halfp_model_path()
torch.save(model, outfile)
print(f'Converted model to half precision: {outfile}')
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# collapses short segments into the previous segment
def whisper_sanitize( results ):
sanitized = json.loads(json.dumps(results))
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sanitized['segments'] = []
for segment in results['segments']:
length = segment['end'] - segment['start']
if length >= MIN_TRAINING_DURATION or len(sanitized['segments']) == 0:
sanitized['segments'].append(segment)
continue
last_segment = sanitized['segments'][-1]
# segment already asimilitated it, somehow
if last_segment['end'] >= segment['end']:
continue
"""
# segment already asimilitated it, somehow
if last_segment['text'].endswith(segment['text']):
continue
"""
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last_segment['text'] += segment['text']
last_segment['end'] = segment['end']
for i in range(len(sanitized['segments'])):
sanitized['segments'][i]['id'] = i
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return sanitized
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def whisper_transcribe( file, language=None ):
# shouldn't happen, but it's for safety
global whisper_model
global whisper_align_model
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if not whisper_model:
load_whisper_model(language=language)
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if args.whisper_backend == "openai/whisper":
if not language:
language = None
return whisper_model.transcribe(file, language=language)
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if args.whisper_backend == "lightmare/whispercpp":
res = whisper_model.transcribe(file)
segments = whisper_model.extract_text_and_timestamps( res )
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result = {
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'text': [],
'segments': []
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}
for segment in segments:
reparsed = {
'start': segment[0] / 100.0,
'end': segment[1] / 100.0,
'text': segment[2],
'id': len(result['segments'])
}
result['text'].append( segment[2] )
result['segments'].append(reparsed)
result['text'] = " ".join(result['text'])
return result
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if args.whisper_backend == "m-bain/whisperx":
import whisperx
device = "cuda" if get_device_name() == "cuda" else "cpu"
result = whisper_model.transcribe(file, batch_size=args.whisper_batchsize)
align_model, metadata = whisper_align_model
result_aligned = whisperx.align(result["segments"], align_model, metadata, file, device, return_char_alignments=False)
result['segments'] = result_aligned['segments']
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result['text'] = []
for segment in result['segments']:
segment['id'] = len(result['text'])
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result['text'].append(segment['text'].strip())
result['text'] = " ".join(result['text'])
return result
def validate_waveform( waveform, sample_rate, min_only=False ):
if not torch.any(waveform < 0):
return "Waveform is empty"
num_channels, num_frames = waveform.shape
duration = num_frames / sample_rate
if duration < MIN_TRAINING_DURATION:
return "Duration too short ({:.3f}s < {:.3f}s)".format(duration, MIN_TRAINING_DURATION)
if not min_only:
if duration > MAX_TRAINING_DURATION:
return "Duration too long ({:.3f}s < {:.3f}s)".format(MAX_TRAINING_DURATION, duration)
return
def transcribe_dataset( voice, language=None, skip_existings=False, progress=None ):
unload_tts()
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global whisper_model
if whisper_model is None:
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load_whisper_model(language=language)
results = {}
files = get_voice(voice, load_latents=False)
indir = f'./training/{voice}/'
infile = f'{indir}/whisper.json'
quantize_in_memory = args.tts_backend == "vall-e"
os.makedirs(f'{indir}/audio/', exist_ok=True)
TARGET_SAMPLE_RATE = 22050
if args.tts_backend != "tortoise":
TARGET_SAMPLE_RATE = 24000
if tts:
TARGET_SAMPLE_RATE = tts.input_sample_rate
if os.path.exists(infile):
results = json.load(open(infile, 'r', encoding="utf-8"))
for file in tqdm(files, desc="Iterating through voice files"):
basename = os.path.basename(file)
if basename in results and skip_existings:
print(f"Skipping already parsed file: {basename}")
continue
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try:
result = whisper_transcribe(file, language=language)
except Exception as e:
print("Failed to transcribe:", file, e)
continue
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results[basename] = result
if not quantize_in_memory:
waveform, sample_rate = torchaudio.load(file)
# resample to the input rate, since it'll get resampled for training anyways
# this should also "help" increase throughput a bit when filling the dataloaders
waveform, sample_rate = resample(waveform, sample_rate, TARGET_SAMPLE_RATE)
if waveform.shape[0] == 2:
waveform = waveform[:1]
try:
kwargs = {}
if basename[-4:] == ".wav":
kwargs['encoding'] = "PCM_S"
kwargs['bits_per_sample'] = 16
torchaudio.save(f"{indir}/audio/{basename}", waveform, sample_rate, **kwargs)
except Exception as e:
print(e)
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with open(infile, 'w', encoding="utf-8") as f:
f.write(json.dumps(results, indent='\t'))
do_gc()
modified = False
for basename in results:
try:
sanitized = whisper_sanitize(results[basename])
if len(sanitized['segments']) > 0 and len(sanitized['segments']) != len(results[basename]['segments']):
results[basename] = sanitized
modified = True
print("Segments sanizited: ", basename)
except Exception as e:
print("Failed to sanitize:", basename, e)
pass
if modified:
os.rename(infile, infile.replace(".json", ".unsanitized.json"))
with open(infile, 'w', encoding="utf-8") as f:
f.write(json.dumps(results, indent='\t'))
return f"Processed dataset to: {indir}"
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def slice_waveform( waveform, sample_rate, start, end, trim ):
start = int(start * sample_rate)
end = int(end * sample_rate)
if start < 0:
start = 0
if end >= waveform.shape[-1]:
end = waveform.shape[-1] - 1
sliced = waveform[:, start:end]
error = validate_waveform( sliced, sample_rate, min_only=True )
if trim and not error:
sliced = torchaudio.functional.vad( sliced, sample_rate )
return sliced, error
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def slice_dataset( voice, trim_silence=True, start_offset=0, end_offset=0, results=None, progress=gr.Progress() ):
indir = f'./training/{voice}/'
infile = f'{indir}/whisper.json'
messages = []
if not os.path.exists(infile):
message = f"Missing dataset: {infile}"
print(message)
return message
if results is None:
results = json.load(open(infile, 'r', encoding="utf-8"))
TARGET_SAMPLE_RATE = 22050
if args.tts_backend != "tortoise":
TARGET_SAMPLE_RATE = 24000
if tts:
TARGET_SAMPLE_RATE = tts.input_sample_rate
files = 0
segments = 0
for filename in results:
path = f'./voices/{voice}/{filename}'
extension = os.path.splitext(filename)[-1][1:]
out_extension = extension # "wav"
if not os.path.exists(path):
path = f'./training/{voice}/{filename}'
if not os.path.exists(path):
message = f"Missing source audio: {filename}"
print(message)
messages.append(message)
continue
files += 1
result = results[filename]
waveform, sample_rate = torchaudio.load(path)
num_channels, num_frames = waveform.shape
duration = num_frames / sample_rate
for segment in result['segments']:
file = filename.replace(f".{extension}", f"_{pad(segment['id'], 4)}.{out_extension}")
sliced, error = slice_waveform( waveform, sample_rate, segment['start'] + start_offset, segment['end'] + end_offset, trim_silence )
if error:
message = f"{error}, skipping... {file}"
print(message)
messages.append(message)
continue
sliced, _ = resample( sliced, sample_rate, TARGET_SAMPLE_RATE )
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if waveform.shape[0] == 2:
waveform = waveform[:1]
kwargs = {}
if file[-4:] == ".wav":
kwargs['encoding'] = "PCM_S"
kwargs['bits_per_sample'] = 16
torchaudio.save(f"{indir}/audio/{file}", sliced, TARGET_SAMPLE_RATE, **kwargs)
segments +=1
messages.append(f"Sliced segments: {files} => {segments}.")
return "\n".join(messages)
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# takes an LJSpeech-dataset-formatted .txt file and phonemize it
def phonemize_txt_file( path ):
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with open(path, 'r', encoding='utf-8') as f:
lines = f.readlines()
reparsed = []
with open(path.replace(".txt", ".phn.txt"), 'a', encoding='utf-8') as f:
for line in tqdm(lines, desc='Phonemizing...'):
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split = line.split("|")
audio = split[0]
text = split[2]
phonemes = phonemizer( text )
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reparsed.append(f'{audio}|{phonemes}')
f.write(f'\n{audio}|{phonemes}')
joined = "\n".join(reparsed)
with open(path.replace(".txt", ".phn.txt"), 'w', encoding='utf-8') as f:
f.write(joined)
return joined
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# takes an LJSpeech-dataset-formatted .txt (and phonemized .phn.txt from the above) and creates a JSON that should slot in as whisper.json
def create_dataset_json( path ):
with open(path, 'r', encoding='utf-8') as f:
lines = f.readlines()
phonemes = None
phn_path = path.replace(".txt", ".phn.txt")
if os.path.exists(phn_path):
with open(phn_path, 'r', encoding='utf-8') as f:
phonemes = f.readlines()
data = {}
for line in lines:
split = line.split("|")
audio = split[0]
text = split[1]
data[audio] = {
'text': text.strip()
}
for line in phonemes:
split = line.split("|")
audio = split[0]
text = split[1]
data[audio]['phonemes'] = text.strip()
with open(path.replace(".txt", ".json"), 'w', encoding='utf-8') as f:
f.write(json.dumps(data, indent="\t"))
cached_backends = {}
def phonemizer( text, language="en-us" ):
from phonemizer import phonemize
from phonemizer.backend import BACKENDS
def _get_backend( language="en-us", backend="espeak" ):
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key = f'{language}_{backend}'
if key in cached_backends:
return cached_backends[key]
if backend == 'espeak':
phonemizer = BACKENDS[backend]( language, preserve_punctuation=True, with_stress=True)
elif backend == 'espeak-mbrola':
phonemizer = BACKENDS[backend]( language )
else:
phonemizer = BACKENDS[backend]( language, preserve_punctuation=True )
cached_backends[key] = phonemizer
return phonemizer
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if language == "en":
language = "en-us"
backend = _get_backend(language=language, backend=args.phonemizer_backend)
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if backend is not None:
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tokens = backend.phonemize( [text], strip=True )
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else:
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tokens = phonemize( [text], language=language, strip=True, preserve_punctuation=True, with_stress=True )
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return tokens[0] if len(tokens) == 0 else tokens
tokenized = " ".join( tokens )
def should_phonemize():
if args.tts_backend == "vall-e":
return False
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should = args.tokenizer_json is not None and args.tokenizer_json[-8:] == "ipa.json"
if should:
try:
from phonemizer import phonemize
except Exception as e:
return False
return should
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def prepare_dataset( voice, use_segments=False, text_length=0, audio_length=0, progress=gr.Progress() ):
indir = f'./training/{voice}/'
infile = f'{indir}/whisper.json'
if not os.path.exists(infile):
message = f"Missing dataset: {infile}"
print(message)
return message
results = json.load(open(infile, 'r', encoding="utf-8"))
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errored = 0
messages = []
normalize = False # True
phonemize = should_phonemize()
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lines = { 'training': [], 'validation': [] }
segments = {}
quantize_in_memory = args.tts_backend == "vall-e"
if args.tts_backend != "tortoise":
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text_length = 0
audio_length = 0
start_offset = -0.1
end_offset = 0.1
trim_silence = False
TARGET_SAMPLE_RATE = 22050
if args.tts_backend != "tortoise":
TARGET_SAMPLE_RATE = 24000
if tts:
TARGET_SAMPLE_RATE = tts.input_sample_rate
for filename in tqdm(results, desc="Parsing results"):
use_segment = use_segments
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extension = os.path.splitext(filename)[-1][1:]
out_extension = extension # "wav"
result = results[filename]
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lang = result['language']
language = LANGUAGES[lang] if lang in LANGUAGES else lang
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normalizer = EnglishTextNormalizer() if language and language == "english" else BasicTextNormalizer()
# check if unsegmented text exceeds 200 characters
if not use_segment:
if len(result['text']) > MAX_TRAINING_CHAR_LENGTH:
message = f"Text length too long ({MAX_TRAINING_CHAR_LENGTH} < {len(result['text'])}), using segments: {filename}"
print(message)
messages.append(message)
use_segment = True
# check if unsegmented audio exceeds 11.6s
if not use_segment:
path = f'{indir}/audio/{filename}'
if not quantize_in_memory and not os.path.exists(path):
messages.append(f"Missing source audio: {filename}")
errored += 1
continue
duration = 0
for segment in result['segments']:
duration = max(duration, segment['end'])
if duration >= MAX_TRAINING_DURATION:
message = f"Audio too large, using segments: {filename}"
print(message)
messages.append(message)
use_segment = True
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# implicitly segment
if use_segment and not use_segments:
exists = True
for segment in result['segments']:
duration = segment['end'] - segment['start']
if duration <= MIN_TRAINING_DURATION or MAX_TRAINING_DURATION <= duration:
continue
path = f'{indir}/audio/' + filename.replace(f".{extension}", f"_{pad(segment['id'], 4)}.{out_extension}")
if os.path.exists(path):
continue
exists = False
break
if not quantize_in_memory and not exists:
tmp = {}
tmp[filename] = result
print(f"Audio not segmented, segmenting: {filename}")
message = slice_dataset( voice, results=tmp )
print(message)
messages = messages + message.split("\n")
waveform = None
if quantize_in_memory:
path = f'{indir}/audio/{filename}'
if not os.path.exists(path):
path = f'./voices/{voice}/{filename}'
if not os.path.exists(path):
message = f"Audio not found: {path}"
print(message)
messages.append(message)
#continue
else:
waveform = torchaudio.load(path)
waveform = resample(waveform[0], waveform[1], TARGET_SAMPLE_RATE)
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if not use_segment:
segments[filename] = {
'text': result['text'],
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'lang': lang,
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'language': language,
'normalizer': normalizer,
'phonemes': result['phonemes'] if 'phonemes' in result else None
}
if waveform:
segments[filename]['waveform'] = waveform
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else:
for segment in result['segments']:
duration = segment['end'] - segment['start']
if duration <= MIN_TRAINING_DURATION or MAX_TRAINING_DURATION <= duration:
continue
file = filename.replace(f".{extension}", f"_{pad(segment['id'], 4)}.{out_extension}")
segments[file] = {
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'text': segment['text'],
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'lang': lang,
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'language': language,
'normalizer': normalizer,
'phonemes': segment['phonemes'] if 'phonemes' in segment else None
}
if waveform:
sliced, error = slice_waveform( waveform[0], waveform[1], segment['start'] + start_offset, segment['end'] + end_offset, trim_silence )
if error:
message = f"{error}, skipping... {file}"
print(message)
messages.append(message)
segments[file]['error'] = error
#continue
else:
segments[file]['waveform'] = (sliced, waveform[1])
jobs = {
'quantize': [[], []],
'phonemize': [[], []],
}
for file in tqdm(segments, desc="Parsing segments"):
extension = os.path.splitext(file)[-1][1:]
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result = segments[file]
path = f'{indir}/audio/{file}'
text = result['text']
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lang = result['lang']
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language = result['language']
normalizer = result['normalizer']
phonemes = result['phonemes']
if phonemize and phonemes is None:
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phonemes = phonemizer( text, language=lang )
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normalized = normalizer(text) if normalize else text
if len(text) > MAX_TRAINING_CHAR_LENGTH:
message = f"Text length too long ({MAX_TRAINING_CHAR_LENGTH} < {len(text)}), skipping... {file}"
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print(message)
messages.append(message)
errored += 1
continue
# num_channels, num_frames = waveform.shape
#duration = num_frames / sample_rate
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culled = len(text) < text_length
#if not culled and audio_length > 0:
# culled = duration < audio_length
line = f'audio/{file}|{phonemes if phonemize and phonemes else text}'
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lines['training' if not culled else 'validation'].append(line)
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if culled or args.tts_backend != "vall-e":
continue
os.makedirs(f'{indir}/valle/', exist_ok=True)
#os.makedirs(f'./training/valle/data/{voice}/', exist_ok=True)
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phn_file = f'{indir}/valle/{file.replace(f".{extension}",".phn.txt")}'
#phn_file = f'./training/valle/data/{voice}/{file.replace(f".{extension}",".phn.txt")}'
if not os.path.exists(phn_file):
jobs['phonemize'][0].append(phn_file)
jobs['phonemize'][1].append(normalized)
"""
phonemized = valle_phonemize( normalized )
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open(f'{indir}/valle/{file.replace(".wav",".phn.txt")}', 'w', encoding='utf-8').write(" ".join(phonemized))
print("Phonemized:", file, normalized, text)
"""
qnt_file = f'{indir}/valle/{file.replace(f".{extension}",".qnt.pt")}'
#qnt_file = f'./training/valle/data/{voice}/{file.replace(f".{extension}",".qnt.pt")}'
if 'error' not in result:
if not quantize_in_memory and not os.path.exists(path):
message = f"Missing segment, skipping... {file}"
print(message)
messages.append(message)
errored += 1
continue
if not os.path.exists(qnt_file):
waveform = None
if 'waveform' in result:
waveform, sample_rate = result['waveform']
elif os.path.exists(path):
waveform, sample_rate = torchaudio.load(path)
error = validate_waveform( waveform, sample_rate )
if error:
message = f"{error}, skipping... {file}"
print(message)
messages.append(message)
errored += 1
continue
if waveform is not None:
jobs['quantize'][0].append(qnt_file)
jobs['quantize'][1].append((waveform, sample_rate))
"""
quantized = valle_quantize( waveform, sample_rate ).cpu()
torch.save(quantized, f'{indir}/valle/{file.replace(".wav",".qnt.pt")}')
print("Quantized:", file)
"""
for i in tqdm(range(len(jobs['quantize'][0])), desc="Quantizing"):
qnt_file = jobs['quantize'][0][i]
waveform, sample_rate = jobs['quantize'][1][i]
quantized = valle_quantize( waveform, sample_rate ).cpu()
torch.save(quantized, qnt_file)
#print("Quantized:", qnt_file)
for i in tqdm(range(len(jobs['phonemize'][0])), desc="Phonemizing"):
phn_file = jobs['phonemize'][0][i]
normalized = jobs['phonemize'][1][i]
try:
phonemized = valle_phonemize( normalized )
open(phn_file, 'w', encoding='utf-8').write(" ".join(phonemized))
#print("Phonemized:", phn_file)
except Exception as e:
message = f"Failed to phonemize: {phn_file}: {normalized}"
messages.append(message)
print(message)
training_joined = "\n".join(lines['training'])
validation_joined = "\n".join(lines['validation'])
with open(f'{indir}/train.txt', 'w', encoding="utf-8") as f:
f.write(training_joined)
with open(f'{indir}/validation.txt', 'w', encoding="utf-8") as f:
f.write(validation_joined)
messages.append(f"Prepared {len(lines['training'])} lines (validation: {len(lines['validation'])}, culled: {errored}).\n{training_joined}\n\n{validation_joined}")
return "\n".join(messages)
def calc_iterations( epochs, lines, batch_size ):
return int(math.ceil(epochs * math.ceil(lines / batch_size)))
def schedule_learning_rate( iterations, schedule=LEARNING_RATE_SCHEDULE ):
return [int(iterations * d) for d in schedule]
def optimize_training_settings( **kwargs ):
messages = []
settings = {}
settings.update(kwargs)
dataset_path = f"./training/{settings['voice']}/train.txt"
with open(dataset_path, 'r', encoding="utf-8") as f:
lines = len(f.readlines())
if lines == 0:
raise Exception("Empty dataset.")
if settings['batch_size'] > lines:
settings['batch_size'] = lines
messages.append(f"Batch size is larger than your dataset, clamping batch size to: {settings['batch_size']}")
"""
if lines % settings['batch_size'] != 0:
settings['batch_size'] = int(lines / settings['batch_size'])
if settings['batch_size'] == 0:
settings['batch_size'] = 1
messages.append(f"Batch size not neatly divisible by dataset size, adjusting batch size to: {settings['batch_size']}")
"""
if settings['gradient_accumulation_size'] == 0:
settings['gradient_accumulation_size'] = 1
if settings['batch_size'] / settings['gradient_accumulation_size'] < 2:
settings['gradient_accumulation_size'] = int(settings['batch_size'] / 2)
if settings['gradient_accumulation_size'] == 0:
settings['gradient_accumulation_size'] = 1
messages.append(f"Gradient accumulation size is too large for a given batch size, clamping gradient accumulation size to: {settings['gradient_accumulation_size']}")
elif settings['batch_size'] % settings['gradient_accumulation_size'] != 0:
settings['gradient_accumulation_size'] -= settings['batch_size'] % settings['gradient_accumulation_size']
if settings['gradient_accumulation_size'] == 0:
settings['gradient_accumulation_size'] = 1
messages.append(f"Batch size is not evenly divisible by the gradient accumulation size, adjusting gradient accumulation size to: {settings['gradient_accumulation_size']}")
if settings['batch_size'] % settings['gpus'] != 0:
settings['batch_size'] -= settings['batch_size'] % settings['gpus']
if settings['batch_size'] == 0:
settings['batch_size'] = 1
messages.append(f"Batch size not neatly divisible by GPU count, adjusting batch size to: {settings['batch_size']}")
def get_device_batch_size( vram ):
DEVICE_BATCH_SIZE_MAP = [
(70, 128), # based on an A100-80G, I can safely get a ratio of 4096:32 = 128
(32, 64), # based on my two 6800XTs, I can only really safely get a ratio of 128:2 = 64
(16, 8), # based on an A4000, I can do a ratio of 512:64 = 8:1
(8, 4), # interpolated
(6, 2), # based on my 2060, it only really lets me have a batch ratio of 2:1
]
for k, v in DEVICE_BATCH_SIZE_MAP:
if vram > (k-1):
return v
return 1
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if settings['gpus'] > get_device_count():
settings['gpus'] = get_device_count()
messages.append(f"GPU count exceeds defacto GPU count, clamping to: {settings['gpus']}")
if settings['gpus'] <= 1:
settings['gpus'] = 1
else:
messages.append(f"! EXPERIMENTAL ! Multi-GPU training is extremely particular, expect issues.")
# assuming you have equal GPUs
vram = get_device_vram() * settings['gpus']
batch_ratio = int(settings['batch_size'] / settings['gradient_accumulation_size'])
batch_cap = get_device_batch_size(vram)
if batch_ratio > batch_cap:
settings['gradient_accumulation_size'] = int(settings['batch_size'] / batch_cap)
messages.append(f"Batch ratio ({batch_ratio}) is expected to exceed your VRAM capacity ({'{:.3f}'.format(vram)}GB, suggested {batch_cap} batch size cap), adjusting gradient accumulation size to: {settings['gradient_accumulation_size']}")
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iterations = calc_iterations(epochs=settings['epochs'], lines=lines, batch_size=settings['batch_size'])
if settings['epochs'] < settings['save_rate']:
settings['save_rate'] = settings['epochs']
messages.append(f"Save rate is too small for the given iteration step, clamping save rate to: {settings['save_rate']}")
if settings['epochs'] < settings['validation_rate']:
settings['validation_rate'] = settings['epochs']
messages.append(f"Validation rate is too small for the given iteration step, clamping validation rate to: {settings['validation_rate']}")
if settings['resume_state'] and not os.path.exists(settings['resume_state']):
settings['resume_state'] = None
messages.append("Resume path specified, but does not exist. Disabling...")
if settings['bitsandbytes']:
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messages.append("! EXPERIMENTAL ! BitsAndBytes requested.")
if settings['half_p']:
if settings['bitsandbytes']:
settings['half_p'] = False
messages.append("Half Precision requested, but BitsAndBytes is also requested. Due to redundancies, disabling half precision...")
else:
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messages.append("! EXPERIMENTAL ! Half Precision requested.")
if not os.path.exists(get_halfp_model_path()):
convert_to_halfp()
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steps = int(iterations / settings['epochs'])
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messages.append(f"For {settings['epochs']} epochs with {lines} lines in batches of {settings['batch_size']}, iterating for {iterations} steps ({steps}) steps per epoch)")
return settings, messages
def save_training_settings( **kwargs ):
messages = []
settings = {}
settings.update(kwargs)
outjson = f'./training/{settings["voice"]}/train.json'
with open(outjson, 'w', encoding="utf-8") as f:
f.write(json.dumps(settings, indent='\t') )
settings['dataset_path'] = f"./training/{settings['voice']}/train.txt"
settings['validation_path'] = f"./training/{settings['voice']}/validation.txt"
with open(settings['dataset_path'], 'r', encoding="utf-8") as f:
lines = len(f.readlines())
settings['iterations'] = calc_iterations(epochs=settings['epochs'], lines=lines, batch_size=settings['batch_size'])
if not settings['source_model'] or settings['source_model'] == "auto":
settings['source_model'] = f"./models/tortoise/autoregressive{'_half' if settings['half_p'] else ''}.pth"
if settings['half_p']:
if not os.path.exists(get_halfp_model_path()):
convert_to_halfp()
messages.append(f"For {settings['epochs']} epochs with {lines} lines, iterating for {settings['iterations']} steps")
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iterations_per_epoch = settings['iterations'] / settings['epochs']
settings['save_rate'] = int(settings['save_rate'] * iterations_per_epoch)
settings['validation_rate'] = int(settings['validation_rate'] * iterations_per_epoch)
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iterations_per_epoch = int(iterations_per_epoch)
if settings['save_rate'] < 1:
settings['save_rate'] = 1
"""
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if settings['validation_rate'] < 1:
settings['validation_rate'] = 1
"""
"""
if settings['iterations'] % settings['save_rate'] != 0:
adjustment = int(settings['iterations'] / settings['save_rate']) * settings['save_rate']
messages.append(f"Iteration rate is not evenly divisible by save rate, adjusting: {settings['iterations']} => {adjustment}")
settings['iterations'] = adjustment
"""
settings['validation_batch_size'] = int(settings['batch_size'] / settings['gradient_accumulation_size'])
if not os.path.exists(settings['validation_path']):
settings['validation_enabled'] = False
messages.append("Validation not found, disabling validation...")
elif settings['validation_batch_size'] == 0:
settings['validation_enabled'] = False
messages.append("Validation batch size == 0, disabling validation...")
else:
with open(settings['validation_path'], 'r', encoding="utf-8") as f:
validation_lines = len(f.readlines())
if validation_lines < settings['validation_batch_size']:
settings['validation_batch_size'] = validation_lines
messages.append(f"Batch size exceeds validation dataset size, clamping validation batch size to {validation_lines}")
settings['tokenizer_json'] = args.tokenizer_json if args.tokenizer_json else get_tokenizer_jsons()[0]
if settings['gpus'] > get_device_count():
settings['gpus'] = get_device_count()
# what an utter mistake this was
settings['optimizer'] = 'adamw' # if settings['gpus'] == 1 else 'adamw_zero'
if 'learning_rate_scheme' not in settings or settings['learning_rate_scheme'] not in LEARNING_RATE_SCHEMES:
settings['learning_rate_scheme'] = "Multistep"
settings['learning_rate_scheme'] = LEARNING_RATE_SCHEMES[settings['learning_rate_scheme']]
learning_rate_schema = [f"default_lr_scheme: {settings['learning_rate_scheme']}"]
if settings['learning_rate_scheme'] == "MultiStepLR":
if not settings['learning_rate_schedule']:
settings['learning_rate_schedule'] = LEARNING_RATE_SCHEDULE
elif isinstance(settings['learning_rate_schedule'],str):
settings['learning_rate_schedule'] = json.loads(settings['learning_rate_schedule'])
settings['learning_rate_schedule'] = schedule_learning_rate( iterations_per_epoch, settings['learning_rate_schedule'] )
learning_rate_schema.append(f" gen_lr_steps: {settings['learning_rate_schedule']}")
learning_rate_schema.append(f" lr_gamma: 0.5")
elif settings['learning_rate_scheme'] == "CosineAnnealingLR_Restart":
epochs = settings['epochs']
restarts = settings['learning_rate_restarts']
restart_period = int(epochs / restarts)
if 'learning_rate_warmup' not in settings:
settings['learning_rate_warmup'] = 0
if 'learning_rate_min' not in settings:
settings['learning_rate_min'] = 1e-08
if 'learning_rate_period' not in settings:
settings['learning_rate_period'] = [ iterations_per_epoch * restart_period for x in range(epochs) ]
settings['learning_rate_restarts'] = [ iterations_per_epoch * (x+1) * restart_period for x in range(restarts) ] # [52, 104, 156, 208]
if 'learning_rate_restart_weights' not in settings:
settings['learning_rate_restart_weights'] = [ ( restarts - x - 1 ) / restarts for x in range(restarts) ] # [.75, .5, .25, .125]
settings['learning_rate_restart_weights'][-1] = settings['learning_rate_restart_weights'][-2] * 0.5
learning_rate_schema.append(f" T_period: {settings['learning_rate_period']}")
learning_rate_schema.append(f" warmup: {settings['learning_rate_warmup']}")
learning_rate_schema.append(f" eta_min: !!float {settings['learning_rate_min']}")
learning_rate_schema.append(f" restarts: {settings['learning_rate_restarts']}")
learning_rate_schema.append(f" restart_weights: {settings['learning_rate_restart_weights']}")
settings['learning_rate_scheme'] = "\n".join(learning_rate_schema)
if settings['resume_state']:
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settings['source_model'] = f"# pretrain_model_gpt: '{settings['source_model']}'"
settings['resume_state'] = f"resume_state: '{settings['resume_state']}'"
else:
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settings['source_model'] = f"pretrain_model_gpt: '{settings['source_model']}'"
settings['resume_state'] = f"# resume_state: '{settings['resume_state']}'"
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def use_template(template, out):
with open(template, 'r', encoding="utf-8") as f:
yaml = f.read()
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# i could just load and edit the YAML directly, but this is easier, as I don't need to bother with path traversals
for k in settings:
if settings[k] is None:
continue
yaml = yaml.replace(f"${{{k}}}", str(settings[k]))
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with open(out, 'w', encoding="utf-8") as f:
f.write(yaml)
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if args.tts_backend == "tortoise":
use_template(f'./models/.template.dlas.yaml', f'./training/{settings["voice"]}/train.yaml')
elif args.tts_backend == "vall-e":
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settings['model_name'] = "[ 'ar-quarter', 'nar-quarter' ]"
use_template(f'./models/.template.valle.yaml', f'./training/{settings["voice"]}/config.yaml')
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messages.append(f"Saved training output")
return settings, messages
def import_voices(files, saveAs=None, progress=None):
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global args
if not isinstance(files, list):
files = [files]
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for file in tqdm(files, desc="Importing voice files"):
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}")
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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, sample_rate = torchaudio.load(filename)
if args.voice_fixer:
if not voicefixer:
load_voicefixer()
waveform, sample_rate = resample(waveform, sample_rate, 44100)
torchaudio.save(path, waveform, sample_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, sample_rate)
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print(f"Imported voice to {path}")
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def relative_paths( dirs ):
return [ './' + os.path.relpath( d ).replace("\\", "/") for d in dirs ]
def get_voice( name, dir=get_voice_dir(), load_latents=True, extensions=["wav", "mp3", "flac"] ):
subj = f'{dir}/{name}/'
if not os.path.isdir(subj):
return
files = os.listdir(subj)
if load_latents:
extensions.append("pth")
voice = []
for file in files:
ext = os.path.splitext(file)[-1][1:]
if ext not in extensions:
continue
voice.append(f'{subj}/{file}')
return sorted( voice )
def get_voice_list(dir=get_voice_dir(), append_defaults=False, extensions=["wav", "mp3", "flac", "pth"]):
defaults = [ "random", "microphone" ]
os.makedirs(dir, exist_ok=True)
#res = sorted([d for d in os.listdir(dir) if d not in defaults and os.path.isdir(os.path.join(dir, d)) and len(os.listdir(os.path.join(dir, d))) > 0 ])
res = []
for name in os.listdir(dir):
if name in defaults:
continue
if not os.path.isdir(f'{dir}/{name}'):
continue
if len(os.listdir(os.path.join(dir, name))) == 0:
continue
files = get_voice( name, dir=dir, extensions=extensions )
if len(files) > 0:
res.append(name)
else:
for subdir in os.listdir(f'{dir}/{name}'):
if not os.path.isdir(f'{dir}/{name}/{subdir}'):
continue
files = get_voice( f'{name}/{subdir}', dir=dir, extensions=extensions )
if len(files) == 0:
continue
res.append(f'{name}/{subdir}')
res = sorted(res)
if append_defaults:
res = res + defaults
return res
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def get_valle_models(dir="./training/"):
return [ f'{dir}/{d}/config.yaml' for d in os.listdir(dir) if os.path.exists(f'{dir}/{d}/config.yaml') ]
def get_autoregressive_models(dir="./models/finetunes/", prefixed=False, auto=False):
os.makedirs(dir, exist_ok=True)
base = [get_model_path('autoregressive.pth')]
halfp = get_halfp_model_path()
if os.path.exists(halfp):
base.append(halfp)
additionals = sorted([f'{dir}/{d}' for d in os.listdir(dir) if d[-4:] == ".pth" ])
found = []
for training in os.listdir(f'./training/'):
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if not os.path.isdir(f'./training/{training}/') or not os.path.isdir(f'./training/{training}/finetune/') or not os.path.isdir(f'./training/{training}/finetune/models/'):
continue
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models = sorted([ int(d[:-8]) for d in os.listdir(f'./training/{training}/finetune/models/') if d[-8:] == "_gpt.pth" ])
found = found + [ f'./training/{training}/finetune/models/{d}_gpt.pth' for d in models ]
res = base + additionals + found
if prefixed:
for i in range(len(res)):
path = res[i]
hash = hash_file(path)
shorthash = hash[:8]
res[i] = f'[{shorthash}] {path}'
paths = relative_paths(res)
if auto:
paths = ["auto"] + paths
return paths
def get_diffusion_models(dir="./models/finetunes/", prefixed=False):
return relative_paths([ get_model_path('diffusion_decoder.pth') ])
def get_tokenizer_jsons( dir="./models/tokenizers/" ):
additionals = sorted([ f'{dir}/{d}' for d in os.listdir(dir) if d[-5:] == ".json" ]) if os.path.isdir(dir) else []
return relative_paths([ "./modules/tortoise-tts/tortoise/data/tokenizer.json" ] + additionals)
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def tokenize_text( text, config=None, stringed=True, skip_specials=False ):
from tortoise.utils.tokenizer import VoiceBpeTokenizer
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if not config:
config = args.tokenizer_json if args.tokenizer_json else get_tokenizer_jsons()[0]
if not tts:
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tokenizer = VoiceBpeTokenizer(config)
else:
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tokenizer = tts.tokenizer
encoded = tokenizer.encode(text)
decoded = tokenizer.tokenizer.decode(encoded, skip_special_tokens=skip_specials).split(" ")
if stringed:
return "\n".join([ str(encoded), str(decoded) ])
return decoded
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 "train.txt" in os.listdir(os.path.join(dir, d)) ])
def get_training_list(dir="./training/"):
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if args.tts_backend == "tortoise":
return sorted([f'./training/{d}/train.yaml' for d in os.listdir(dir) if os.path.isdir(os.path.join(dir, d)) and "train.yaml" in os.listdir(os.path.join(dir, d)) ])
else:
return sorted([f'./training/{d}/config.yaml' for d in os.listdir(dir) if os.path.isdir(os.path.join(dir, d)) and "config.yaml" in os.listdir(os.path.join(dir, d)) ])
def pad(num, zeroes):
return str(num).zfill(zeroes+1)
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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( dir = None ):
if dir is None:
check_for_updates("./.git/")
check_for_updates("./.git/modules/dlas/")
check_for_updates("./.git/modules/tortoise-tts/")
return
git_dir = dir
if not os.path.isfile(f'{git_dir}/FETCH_HEAD'):
print(f"Cannot check for updates for {dir}: not from a git repo")
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return False
with open(f'{git_dir}/FETCH_HEAD', 'r', encoding="utf-8") as f:
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head = f.read()
match = re.findall(r"^([a-f0-9]+).+?https:\/\/(.+?)\/(.+?)\/(.+?)\n", head)
if match is None or len(match) == 0:
print(f"Cannot check for updates for {dir}: cannot parse FETCH_HEAD")
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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(f"Cannot check for updates for {dir}: cannot fetch from remote")
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return False
remote = res[0]["commit"]["id"]
if remote != local:
print(f"New version found for {dir}: {local[:8]} => {remote[:8]}")
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return True
return False
def notify_progress(message, progress=None, verbose=True):
if verbose:
print(message)
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if progress is None:
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tqdm.write(message)
else:
progress(0, desc=message)
def get_args():
global args
return args
def setup_args(cli=False):
global args
default_arguments = {
'share': False,
'listen': None,
'check-for-updates': False,
'models-from-local-only': False,
'low-vram': False,
'sample-batch-size': None,
'unsqueeze-sample-batches': False,
'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,
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'force-cpu-for-conditioning-latents': False,
'defer-tts-load': False,
'device-override': None,
'prune-nonfinal-outputs': True,
'concurrency-count': 2,
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'autocalculate-voice-chunk-duration-size': 10,
'output-sample-rate': 44100,
'output-volume': 1,
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'results-folder': "./results/",
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'hf-token': None,
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'tts-backend': TTSES[0],
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'autoregressive-model': None,
'diffusion-model': None,
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'vocoder-model': VOCODERS[-1],
'tokenizer-json': None,
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'phonemizer-backend': 'espeak',
'valle-model': None,
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'whisper-backend': 'openai/whisper',
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'whisper-model': "base",
'whisper-batchsize': 1,
'training-default-halfp': False,
'training-default-bnb': True,
'websocket-listen-address': "127.0.0.1",
'websocket-listen-port': 8069,
'websocket-enabled': False
}
if os.path.isfile('./config/exec.json'):
with open(f'./config/exec.json', 'r', encoding="utf-8") as f:
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try:
overrides = json.load(f)
for k in overrides:
default_arguments[k] = overrides[k]
except Exception as e:
print(e)
pass
parser = argparse.ArgumentParser(allow_abbrev=not cli)
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("--prune-nonfinal-outputs", default=default_arguments['prune-nonfinal-outputs'], action='store_true', help="Deletes non-final output files on completing a generation")
parser.add_argument("--device-override", default=default_arguments['device-override'], help="A device string to override pass through Torch")
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("--unsqueeze-sample-batches", default=default_arguments['unsqueeze-sample-batches'], action='store_true', help="Unsqueezes sample batches to process one by one after sampling")
parser.add_argument("--concurrency-count", type=int, default=default_arguments['concurrency-count'], help="How many Gradio events to process at once")
parser.add_argument("--autocalculate-voice-chunk-duration-size", type=float, default=default_arguments['autocalculate-voice-chunk-duration-size'], help="Number of seconds to suggest voice chunk size for (for example, 100 seconds of audio at 10 seconds per chunk will suggest 10 chunks)")
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")
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parser.add_argument("--results-folder", type=str, default=default_arguments['results-folder'], help="Sets output directory")
parser.add_argument("--hf-token", type=str, default=default_arguments['hf-token'], help="HuggingFace Token")
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parser.add_argument("--tts-backend", default=default_arguments['tts-backend'], help="Specifies which TTS backend to use.")
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parser.add_argument("--autoregressive-model", default=default_arguments['autoregressive-model'], help="Specifies which autoregressive model to use for sampling.")
parser.add_argument("--diffusion-model", default=default_arguments['diffusion-model'], help="Specifies which diffusion model to use for sampling.")
parser.add_argument("--vocoder-model", default=default_arguments['vocoder-model'], action='store_true', help="Specifies with vocoder to use")
parser.add_argument("--tokenizer-json", default=default_arguments['tokenizer-json'], help="Specifies which tokenizer json to use for tokenizing.")
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parser.add_argument("--phonemizer-backend", default=default_arguments['phonemizer-backend'], help="Specifies which phonemizer backend to use.")
parser.add_argument("--valle-model", default=default_arguments['valle-model'], help="Specifies which VALL-E model to use for sampling.")
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parser.add_argument("--whisper-backend", default=default_arguments['whisper-backend'], action='store_true', help="Picks which whisper backend to use (openai/whisper, lightmare/whispercpp)")
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parser.add_argument("--whisper-model", default=default_arguments['whisper-model'], help="Specifies which whisper model to use for transcription.")
parser.add_argument("--whisper-batchsize", type=int, default=default_arguments['whisper-batchsize'], help="Specifies batch size for WhisperX")
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parser.add_argument("--training-default-halfp", action='store_true', default=default_arguments['training-default-halfp'], help="Training default: halfp")
parser.add_argument("--training-default-bnb", action='store_true', default=default_arguments['training-default-bnb'], help="Training default: bnb")
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parser.add_argument("--websocket-listen-port", type=int, default=default_arguments['websocket-listen-port'], help="Websocket server listen port, default: 8069")
parser.add_argument("--websocket-listen-address", default=default_arguments['websocket-listen-address'], help="Websocket server listen address, default: 127.0.0.1")
parser.add_argument("--websocket-enabled", action='store_true', default=default_arguments['websocket-enabled'], help="Websocket API server enabled, default: false")
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if cli:
args, unknown = parser.parse_known_args()
else:
args = parser.parse_args()
args.embed_output_metadata = not args.no_embed_output_metadata
if not args.device_override:
set_device_name(args.device_override)
if args.sample_batch_size == 0 and get_device_batch_size() == 1:
print("!WARNING! Automatically deduced sample batch size returned 1.")
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)
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if args.listen_port == 0:
args.listen_port = None
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return args
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def get_default_settings( hypenated=True ):
settings = {
'listen': None if not 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,
'prune-nonfinal-outputs': args.prune_nonfinal_outputs,
'device-override': args.device_override,
'sample-batch-size': args.sample_batch_size,
'unsqueeze-sample-batches': args.unsqueeze_sample_batches,
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'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,
'autocalculate-voice-chunk-duration-size': args.autocalculate_voice_chunk_duration_size,
'output-volume': args.output_volume,
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'results-folder': args.results_folder,
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'hf-token': args.hf_token,
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'tts-backend': args.tts_backend,
'autoregressive-model': args.autoregressive_model,
'diffusion-model': args.diffusion_model,
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'vocoder-model': args.vocoder_model,
'tokenizer-json': args.tokenizer_json,
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'phonemizer-backend': args.phonemizer_backend,
'valle-model': args.valle_model,
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'whisper-backend': args.whisper_backend,
'whisper-model': args.whisper_model,
'whisper-batchsize': args.whisper_batchsize,
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'training-default-halfp': args.training_default_halfp,
'training-default-bnb': args.training_default_bnb,
}
res = {}
for k in settings:
res[k.replace("-", "_") if not hypenated else k] = settings[k]
return res
def update_args( **kwargs ):
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global args
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settings = get_default_settings(hypenated=False)
settings.update(kwargs)
args.listen = settings['listen']
args.share = settings['share']
args.check_for_updates = settings['check_for_updates']
args.models_from_local_only = settings['models_from_local_only']
args.low_vram = settings['low_vram']
args.force_cpu_for_conditioning_latents = settings['force_cpu_for_conditioning_latents']
args.defer_tts_load = settings['defer_tts_load']
args.prune_nonfinal_outputs = settings['prune_nonfinal_outputs']
args.device_override = settings['device_override']
args.sample_batch_size = settings['sample_batch_size']
args.unsqueeze_sample_batches = settings['unsqueeze_sample_batches']
args.embed_output_metadata = settings['embed_output_metadata']
args.latents_lean_and_mean = settings['latents_lean_and_mean']
args.voice_fixer = settings['voice_fixer']
args.voice_fixer_use_cuda = settings['voice_fixer_use_cuda']
args.concurrency_count = settings['concurrency_count']
args.output_sample_rate = 44000
args.autocalculate_voice_chunk_duration_size = settings['autocalculate_voice_chunk_duration_size']
args.output_volume = settings['output_volume']
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args.results_folder = settings['results_folder']
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args.hf_token = settings['hf_token']
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args.tts_backend = settings['tts_backend']
args.autoregressive_model = settings['autoregressive_model']
args.diffusion_model = settings['diffusion_model']
args.vocoder_model = settings['vocoder_model']
args.tokenizer_json = settings['tokenizer_json']
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args.phonemizer_backend = settings['phonemizer_backend']
args.valle_model = settings['valle_model']
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args.whisper_backend = settings['whisper_backend']
args.whisper_model = settings['whisper_model']
args.whisper_batchsize = settings['whisper_batchsize']
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args.training_default_halfp = settings['training_default_halfp']
args.training_default_bnb = settings['training_default_bnb']
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save_args_settings()
def save_args_settings():
global args
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settings = get_default_settings()
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os.makedirs('./config/', exist_ok=True)
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with open(f'./config/exec.json', 'w', encoding="utf-8") as f:
f.write(json.dumps(settings, indent='\t') )
# super kludgy )`;
def import_generate_settings(file = None):
if not file:
file = "./config/generate.json"
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res = {
'text': None,
'delimiter': None,
'emotion': None,
'prompt': None,
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'voice': "random",
'mic_audio': None,
'voice_latents_chunks': None,
'candidates': None,
'seed': None,
'num_autoregressive_samples': 16,
'diffusion_iterations': 30,
'temperature': 0.8,
'diffusion_sampler': "DDIM",
'breathing_room': 8 ,
'cvvp_weight': 0.0,
'top_p': 0.8,
'diffusion_temperature': 1.0,
'length_penalty': 1.0,
'repetition_penalty': 2.0,
'cond_free_k': 2.0,
'experimentals': None,
}
settings, _ = read_generate_settings(file, read_latents=False)
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if settings is not None:
res.update(settings)
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return res
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def reset_generate_settings():
with open(f'./config/generate.json', 'w', encoding="utf-8") as f:
f.write(json.dumps({}, indent='\t') )
return import_generate_settings()
def read_generate_settings(file, read_latents=True):
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j = None
latents = None
if isinstance(file, list) and len(file) == 1:
file = file[0]
try:
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)
except Exception as e:
pass
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if j is not None:
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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"])
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return (
j,
latents,
)
def version_check_tts( min_version ):
global tts
if not tts:
raise Exception("TTS is not initialized")
if not hasattr(tts, 'version'):
return False
if min_version[0] > tts.version[0]:
return True
if min_version[1] > tts.version[1]:
return True
if min_version[2] >= tts.version[2]:
return True
return False
def load_tts( restart=False,
# TorToiSe configs
autoregressive_model=None, diffusion_model=None, vocoder_model=None, tokenizer_json=None,
# VALL-E configs
valle_model=None,
):
global args
global tts
if restart:
unload_tts()
tts_loading = True
if args.tts_backend == "tortoise":
if autoregressive_model:
args.autoregressive_model = autoregressive_model
else:
autoregressive_model = args.autoregressive_model
if autoregressive_model == "auto":
autoregressive_model = deduce_autoregressive_model()
if diffusion_model:
args.diffusion_model = diffusion_model
else:
diffusion_model = args.diffusion_model
if vocoder_model:
args.vocoder_model = vocoder_model
else:
vocoder_model = args.vocoder_model
if tokenizer_json:
args.tokenizer_json = tokenizer_json
else:
tokenizer_json = args.tokenizer_json
if get_device_name() == "cpu":
print("!!!! WARNING !!!! No GPU available in PyTorch. You may need to reinstall PyTorch.")
print(f"Loading TorToiSe... (AR: {autoregressive_model}, diffusion: {diffusion_model}, vocoder: {vocoder_model})")
tts = TorToise_TTS(minor_optimizations=not args.low_vram, autoregressive_model_path=autoregressive_model, diffusion_model_path=diffusion_model, vocoder_model=vocoder_model, tokenizer_json=tokenizer_json, unsqueeze_sample_batches=args.unsqueeze_sample_batches)
elif args.tts_backend == "vall-e":
if valle_model:
args.valle_model = valle_model
else:
valle_model = args.valle_model
print(f"Loading VALL-E... (Config: {valle_model})")
tts = VALLE_TTS(config=args.valle_model)
elif args.tts_backend == "bark":
print(f"Loading Bark...")
tts = Bark_TTS(small=args.low_vram)
print("Loaded TTS, ready for generation.")
tts_loading = False
return tts
def unload_tts():
global tts
if tts:
del tts
tts = None
print("Unloaded TTS")
do_gc()
def reload_tts():
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unload_tts()
load_tts()
def get_current_voice():
global current_voice
if current_voice:
return current_voice
settings, _ = read_generate_settings("./config/generate.json", read_latents=False)
if settings and "voice" in settings['voice']:
return settings["voice"]
return None
def deduce_autoregressive_model(voice=None):
if not voice:
voice = get_current_voice()
if voice:
if os.path.exists(f'./models/finetunes/{voice}.pth'):
return f'./models/finetunes/{voice}.pth'
dir = f'./training/{voice}/finetune/models/'
if os.path.isdir(dir):
counts = sorted([ int(d[:-8]) for d in os.listdir(dir) if d[-8:] == "_gpt.pth" ])
names = [ f'{dir}/{d}_gpt.pth' for d in counts ]
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if len(names) > 0:
return names[-1]
if args.autoregressive_model != "auto":
return args.autoregressive_model
return get_model_path('autoregressive.pth')
def update_autoregressive_model(autoregressive_model_path):
if args.tts_backend != "tortoise":
raise f"Unsupported backend: {args.tts_backend}"
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if autoregressive_model_path == "auto":
autoregressive_model_path = deduce_autoregressive_model()
else:
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match = re.findall(r'^\[[a-fA-F0-9]{8}\] (.+?)$', autoregressive_model_path)
if match:
autoregressive_model_path = match[0]
if not autoregressive_model_path or not os.path.exists(autoregressive_model_path):
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print(f"Invalid model: {autoregressive_model_path}")
return
args.autoregressive_model = autoregressive_model_path
save_args_settings()
print(f'Stored autoregressive model to settings: {autoregressive_model_path}')
global tts
if not tts:
if tts_loading:
raise Exception("TTS is still initializing...")
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return
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
if autoregressive_model_path == tts.autoregressive_model_path:
return
tts.load_autoregressive_model(autoregressive_model_path)
do_gc()
return autoregressive_model_path
def update_diffusion_model(diffusion_model_path):
if args.tts_backend != "tortoise":
raise f"Unsupported backend: {args.tts_backend}"
match = re.findall(r'^\[[a-fA-F0-9]{8}\] (.+?)$', diffusion_model_path)
if match:
diffusion_model_path = match[0]
if not diffusion_model_path or not os.path.exists(diffusion_model_path):
print(f"Invalid model: {diffusion_model_path}")
return
args.diffusion_model = diffusion_model_path
save_args_settings()
print(f'Stored diffusion model to settings: {diffusion_model_path}')
global tts
if not tts:
if tts_loading:
raise Exception("TTS is still initializing...")
return
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
if diffusion_model_path == "auto":
diffusion_model_path = deduce_diffusion_model()
if diffusion_model_path == tts.diffusion_model_path:
return
tts.load_diffusion_model(diffusion_model_path)
do_gc()
return diffusion_model_path
def update_vocoder_model(vocoder_model):
if args.tts_backend != "tortoise":
raise f"Unsupported backend: {args.tts_backend}"
args.vocoder_model = vocoder_model
save_args_settings()
print(f'Stored vocoder model to settings: {vocoder_model}')
global tts
if not tts:
if tts_loading:
raise Exception("TTS is still initializing...")
return
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
print(f"Loading model: {vocoder_model}")
tts.load_vocoder_model(vocoder_model)
print(f"Loaded model: {tts.vocoder_model}")
do_gc()
return vocoder_model
def update_tokenizer(tokenizer_json):
if args.tts_backend != "tortoise":
raise f"Unsupported backend: {args.tts_backend}"
args.tokenizer_json = tokenizer_json
save_args_settings()
print(f'Stored tokenizer to settings: {tokenizer_json}')
global tts
if not tts:
if tts_loading:
raise Exception("TTS is still initializing...")
return
if hasattr(tts, "loading") and tts.loading:
raise Exception("TTS is still initializing...")
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print(f"Loading tokenizer vocab: {tokenizer_json}")
tts.load_tokenizer_json(tokenizer_json)
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print(f"Loaded tokenizer vocab: {tts.tokenizer_json}")
do_gc()
return vocoder_model
def load_voicefixer(restart=False):
global voicefixer
if restart:
unload_voicefixer()
try:
print("Loading Voicefixer")
from voicefixer import VoiceFixer
voicefixer = VoiceFixer()
print("Loaded Voicefixer")
except Exception as e:
print(f"Error occurred while tring to initialize voicefixer: {e}")
if voicefixer:
del voicefixer
voicefixer = None
def unload_voicefixer():
global voicefixer
if voicefixer:
del voicefixer
voicefixer = None
print("Unloaded Voicefixer")
do_gc()
def load_whisper_model(language=None, model_name=None, progress=None):
global whisper_model
global whisper_align_model
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if args.whisper_backend not in WHISPER_BACKENDS:
raise Exception(f"unavailable backend: {args.whisper_backend}")
if not model_name:
model_name = args.whisper_model
else:
args.whisper_model = model_name
save_args_settings()
if language and f'{model_name}.{language}' in WHISPER_SPECIALIZED_MODELS:
model_name = f'{model_name}.{language}'
print(f"Loading specialized model for language: {language}")
notify_progress(f"Loading Whisper model: {model_name}", progress=progress)
if args.whisper_backend == "openai/whisper":
import whisper
try:
#is it possible for model to fit on vram but go oom later on while executing on data?
whisper_model = whisper.load_model(model_name)
except:
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print("Out of VRAM memory. falling back to loading Whisper on CPU.")
whisper_model = whisper.load_model(model_name, device="cpu")
elif args.whisper_backend == "lightmare/whispercpp":
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from whispercpp import Whisper
if not language:
language = 'auto'
b_lang = language.encode('ascii')
whisper_model = Whisper(model_name, models_dir='./models/', language=b_lang)
elif args.whisper_backend == "m-bain/whisperx":
import whisper, whisperx
device = "cuda" if get_device_name() == "cuda" else "cpu"
whisper_model = whisperx.load_model(model_name, device)
whisper_align_model = whisperx.load_align_model(model_name="WAV2VEC2_ASR_LARGE_LV60K_960H" if language=="en" else None, language_code=language, device=device)
print("Loaded Whisper model")
def unload_whisper():
global whisper_model
global whisper_align_model
if whisper_align_model:
del whisper_align_model
whisper_align_model = None
if whisper_model:
del whisper_model
whisper_model = None
print("Unloaded Whisper")
do_gc()
# shamelessly borrowed from Voldy's Web UI: https://github.com/AUTOMATIC1111/stable-diffusion-webui/blob/master/modules/extras.py#L74
def merge_models( primary_model_name, secondary_model_name, alpha, progress=gr.Progress() ):
key_blacklist = []
def weighted_sum(theta0, theta1, alpha):
return ((1 - alpha) * theta0) + (alpha * theta1)
def read_model( filename ):
print(f"Loading {filename}")
return torch.load(filename)
theta_func = weighted_sum
theta_0 = read_model(primary_model_name)
theta_1 = read_model(secondary_model_name)
for key in tqdm(theta_0.keys(), desc="Merging..."):
if key in key_blacklist:
print("Skipping ignored key:", key)
continue
a = theta_0[key]
b = theta_1[key]
if a.dtype != torch.float32 and a.dtype != torch.float16:
print("Skipping key:", key, a.dtype)
continue
if b.dtype != torch.float32 and b.dtype != torch.float16:
print("Skipping key:", key, b.dtype)
continue
theta_0[key] = theta_func(a, b, alpha)
del theta_1
primary_basename = os.path.splitext(os.path.basename(primary_model_name))[0]
secondary_basename = os.path.splitext(os.path.basename(secondary_model_name))[0]
suffix = "{:.3f}".format(alpha)
output_path = f'./models/finetunes/{primary_basename}_{secondary_basename}_{suffix}_merge.pth'
torch.save(theta_0, output_path)
message = f"Saved to {output_path}"
print(message)
return message