mirror of https://github.com/coqui-ai/TTS.git
Add training and inference columns
This commit is contained in:
parent
774c4c1743
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@ -0,0 +1,163 @@
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import os
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from trainer import Trainer, TrainerArgs
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from TTS.config.shared_configs import BaseDatasetConfig
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from TTS.tts.datasets import load_tts_samples
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from TTS.tts.layers.xtts.trainer.gpt_trainer import GPTArgs, GPTTrainer, GPTTrainerConfig, XttsAudioConfig
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from TTS.utils.manage import ModelManager
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def train_gpt(language, num_epochs, batch_size, train_csv, eval_csv, output_path):
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# Logging parameters
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RUN_NAME = "GPT_XTTSv2.1_FT"
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PROJECT_NAME = "XTTS_trainer"
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DASHBOARD_LOGGER = "tensorboard"
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LOGGER_URI = None
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# Set here the path that the checkpoints will be saved. Default: ./run/training/
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OUT_PATH = os.path.join(output_path, "run", "training")
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# Training Parameters
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OPTIMIZER_WD_ONLY_ON_WEIGHTS = True # for multi-gpu training please make it False
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START_WITH_EVAL = True # if True it will star with evaluation
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BATCH_SIZE = batch_size # set here the batch size
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GRAD_ACUMM_STEPS = 1 # set here the grad accumulation steps
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# Note: we recommend that BATCH_SIZE * GRAD_ACUMM_STEPS need to be at least 252 for more efficient training. You can increase/decrease BATCH_SIZE but then set GRAD_ACUMM_STEPS accordingly.
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# Define here the dataset that you want to use for the fine-tuning on.
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config_dataset = BaseDatasetConfig(
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formatter="coqui",
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dataset_name="ft_dataset",
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path=os.path.dirname(train_csv),
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meta_file_train=train_csv,
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meta_file_val=eval_csv,
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language=language,
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)
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# Add here the configs of the datasets
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DATASETS_CONFIG_LIST = [config_dataset]
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# Define the path where XTTS v2.0.1 files will be downloaded
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CHECKPOINTS_OUT_PATH = os.path.join(OUT_PATH, "XTTS_v2.0_original_model_files/")
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os.makedirs(CHECKPOINTS_OUT_PATH, exist_ok=True)
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# DVAE files
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DVAE_CHECKPOINT_LINK = "https://coqui.gateway.scarf.sh/hf-coqui/XTTS-v2/main/dvae.pth"
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MEL_NORM_LINK = "https://coqui.gateway.scarf.sh/hf-coqui/XTTS-v2/main/mel_stats.pth"
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# Set the path to the downloaded files
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DVAE_CHECKPOINT = os.path.join(CHECKPOINTS_OUT_PATH, os.path.basename(DVAE_CHECKPOINT_LINK))
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MEL_NORM_FILE = os.path.join(CHECKPOINTS_OUT_PATH, os.path.basename(MEL_NORM_LINK))
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# download DVAE files if needed
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if not os.path.isfile(DVAE_CHECKPOINT) or not os.path.isfile(MEL_NORM_FILE):
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print(" > Downloading DVAE files!")
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ModelManager._download_model_files([MEL_NORM_LINK, DVAE_CHECKPOINT_LINK], CHECKPOINTS_OUT_PATH, progress_bar=True)
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# Download XTTS v2.0 checkpoint if needed
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TOKENIZER_FILE_LINK = "https://coqui.gateway.scarf.sh/hf-coqui/XTTS-v2/main/vocab.json"
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XTTS_CHECKPOINT_LINK = "https://coqui.gateway.scarf.sh/hf-coqui/XTTS-v2/main/model.pth"
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XTTS_CONFIG_LINK = "https://coqui.gateway.scarf.sh/hf-coqui/XTTS-v2/main/config.json"
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# XTTS transfer learning parameters: You we need to provide the paths of XTTS model checkpoint that you want to do the fine tuning.
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TOKENIZER_FILE = os.path.join(CHECKPOINTS_OUT_PATH, os.path.basename(TOKENIZER_FILE_LINK)) # vocab.json file
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XTTS_CHECKPOINT = os.path.join(CHECKPOINTS_OUT_PATH, os.path.basename(XTTS_CHECKPOINT_LINK)) # model.pth file
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XTTS_CONFIG_FILE = os.path.join(CHECKPOINTS_OUT_PATH, os.path.basename(XTTS_CONFIG_LINK)) # config.json file
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# download XTTS v2.0 files if needed
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if not os.path.isfile(TOKENIZER_FILE) or not os.path.isfile(XTTS_CHECKPOINT):
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print(" > Downloading XTTS v2.0 files!")
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ModelManager._download_model_files(
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[TOKENIZER_FILE_LINK, XTTS_CHECKPOINT_LINK, XTTS_CONFIG_LINK], CHECKPOINTS_OUT_PATH, progress_bar=True
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)
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# init args and config
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model_args = GPTArgs(
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max_conditioning_length=132300, # 6 secs
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min_conditioning_length=66150, # 3 secs
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debug_loading_failures=False,
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max_wav_length=255995, # ~11.6 seconds
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max_text_length=200,
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mel_norm_file=MEL_NORM_FILE,
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dvae_checkpoint=DVAE_CHECKPOINT,
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xtts_checkpoint=XTTS_CHECKPOINT, # checkpoint path of the model that you want to fine-tune
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tokenizer_file=TOKENIZER_FILE,
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gpt_num_audio_tokens=1026,
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gpt_start_audio_token=1024,
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gpt_stop_audio_token=1025,
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gpt_use_masking_gt_prompt_approach=True,
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gpt_use_perceiver_resampler=True,
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)
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# define audio config
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audio_config = XttsAudioConfig(sample_rate=22050, dvae_sample_rate=22050, output_sample_rate=24000)
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# training parameters config
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config = GPTTrainerConfig(
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epochs=num_epochs,
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output_path=OUT_PATH,
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model_args=model_args,
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run_name=RUN_NAME,
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project_name=PROJECT_NAME,
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run_description="""
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GPT XTTS training
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""",
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dashboard_logger=DASHBOARD_LOGGER,
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logger_uri=LOGGER_URI,
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audio=audio_config,
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batch_size=BATCH_SIZE,
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batch_group_size=48,
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eval_batch_size=BATCH_SIZE,
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num_loader_workers=8,
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eval_split_max_size=256,
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print_step=50,
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plot_step=100,
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log_model_step=100,
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save_step=1000,
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save_n_checkpoints=1,
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save_checkpoints=True,
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# target_loss="loss",
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print_eval=False,
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# Optimizer values like tortoise, pytorch implementation with modifications to not apply WD to non-weight parameters.
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optimizer="AdamW",
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optimizer_wd_only_on_weights=OPTIMIZER_WD_ONLY_ON_WEIGHTS,
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optimizer_params={"betas": [0.9, 0.96], "eps": 1e-8, "weight_decay": 1e-2},
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lr=5e-06, # learning rate
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lr_scheduler="MultiStepLR",
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# it was adjusted accordly for the new step scheme
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lr_scheduler_params={"milestones": [50000 * 18, 150000 * 18, 300000 * 18], "gamma": 0.5, "last_epoch": -1},
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test_sentences=[],
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)
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# init the model from config
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model = GPTTrainer.init_from_config(config)
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# load training samples
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train_samples, eval_samples = load_tts_samples(
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DATASETS_CONFIG_LIST,
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eval_split=True,
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eval_split_max_size=config.eval_split_max_size,
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eval_split_size=config.eval_split_size,
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)
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# init the trainer and 🚀
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trainer = Trainer(
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TrainerArgs(
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restore_path=None, # xtts checkpoint is restored via xtts_checkpoint key so no need of restore it using Trainer restore_path parameter
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skip_train_epoch=False,
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start_with_eval=START_WITH_EVAL,
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grad_accum_steps=GRAD_ACUMM_STEPS,
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),
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config,
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output_path=OUT_PATH,
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model=model,
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train_samples=train_samples,
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eval_samples=eval_samples,
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)
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trainer.fit()
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return XTTS_CONFIG_FILE, XTTS_CHECKPOINT, TOKENIZER_FILE, trainer.output_path, train_samples[0]["audio_file"]
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@ -10,15 +10,50 @@ import os
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import torch
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import torch
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import torchaudio
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import torchaudio
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from TTS.demos.xtts_ft_demo.utils.formatter import format_audio_list, list_audios
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from TTS.demos.xtts_ft_demo.utils.formatter import format_audio_list, list_audios
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from TTS.demos.xtts_ft_demo.utils.gpt_train import train_gpt
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from TTS.tts.configs.xtts_config import XttsConfig
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from TTS.tts.models.xtts import Xtts
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import logging
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PORT = 5003
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PORT = 5003
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def load_model(xtts_checkpoint, xtts_config, xtts_vocab):
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config = XttsConfig()
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config.load_json(xtts_config)
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model = Xtts.init_from_config(config)
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print("Loading XTTS model! ")
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model.load_checkpoint(config, checkpoint_path=xtts_checkpoint, vocab_path=xtts_vocab, use_deepspeed=False)
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if torch.cuda.is_available():
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model.cuda()
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return model
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def run_tts(lang, tts_text, xtts_checkpoint, xtts_config, xtts_vocab, speaker_audio_file, state_vars):
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# ToDo: add the load in other function to fast inference
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model = load_model(xtts_checkpoint, xtts_config, xtts_vocab)
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gpt_cond_latent, speaker_embedding = model.get_conditioning_latents(audio_path=speaker_audio_file, gpt_cond_len=model.config.gpt_cond_len, max_ref_length=model.config.max_ref_len, sound_norm_refs=model.config.sound_norm_refs)
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speaker_embedding
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out = model.inference(
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text=tts_text,
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language=lang,
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gpt_cond_latent=gpt_cond_latent,
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speaker_embedding=speaker_embedding,
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temperature=model.config.temperature, # Add custom parameters here
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length_penalty=model.config.length_penalty,
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repetition_penalty=model.config.repetition_penalty,
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top_k=model.config.top_k,
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top_p=model.config.top_p,
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)
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with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as fp:
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out["wav"] = torch.tensor(out["wav"]).unsqueeze(0)
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out_path = fp.name
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torchaudio.save(out_path, out["wav"], 24000)
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return out_path, speaker_audio_file
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def run_tts(lang, tts_text, state_vars, temperature, rms_norm_output=False):
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return None
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# define a logger to redirect
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# define a logger to redirect
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class Logger:
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class Logger:
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@ -43,6 +78,16 @@ sys.stdout = Logger()
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sys.stderr = sys.stdout
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sys.stderr = sys.stdout
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# logging.basicConfig(stream=sys.stdout, level=logging.INFO)
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import logging
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logging.basicConfig(
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level=logging.INFO,
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format="%(asctime)s [%(levelname)s] %(message)s",
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handlers=[
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logging.StreamHandler(sys.stdout)
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]
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)
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def read_logs():
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def read_logs():
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sys.stdout.flush()
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sys.stdout.flush()
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with open(sys.stdout.log_file, "r") as f:
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with open(sys.stdout.log_file, "r") as f:
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@ -82,8 +127,8 @@ with gr.Blocks() as demo:
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"ja"
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"ja"
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],
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],
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)
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)
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voice_ready = gr.Label(
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progress_data = gr.Label(
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label="Progress."
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label="Progress:"
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)
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)
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logs = gr.Textbox(
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logs = gr.Textbox(
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label="Logs:",
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label="Logs:",
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@ -94,23 +139,78 @@ with gr.Blocks() as demo:
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prompt_compute_btn = gr.Button(value="Step 1 - Create dataset.")
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prompt_compute_btn = gr.Button(value="Step 1 - Create dataset.")
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with gr.Column() as col2:
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with gr.Column() as col2:
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num_epochs = gr.Slider(
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label="num_epochs",
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minimum=1,
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maximum=100,
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step=1,
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value=2,# 15
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)
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batch_size = gr.Slider(
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label="batch_size",
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minimum=2,
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maximum=512,
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step=1,
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value=15,
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)
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progress_train = gr.Label(
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label="Progress:"
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)
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logs_tts_train = gr.Textbox(
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label="Logs:",
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interactive=False,
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)
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demo.load(read_logs, None, logs_tts_train, every=1)
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train_btn = gr.Button(value="Step 2 - Run the training")
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with gr.Column() as col3:
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xtts_checkpoint = gr.Textbox(
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label="XTTS checkpoint path:",
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value="",
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)
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xtts_config = gr.Textbox(
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label="XTTS config path:",
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value="",
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)
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xtts_vocab = gr.Textbox(
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label="XTTS config path:",
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value="",
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)
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speaker_reference_audio = gr.Textbox(
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label="Speaker reference audio:",
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value="",
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)
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tts_language = gr.Dropdown(
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label="Language",
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value="en",
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choices=[
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"en",
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"es",
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"fr",
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"de",
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"it",
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"pt",
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"pl",
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"tr",
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"ru",
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"nl",
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"cs",
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"ar",
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"zh",
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"hu",
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"ko",
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"ja",
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]
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)
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tts_text = gr.Textbox(
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tts_text = gr.Textbox(
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label="Input Text.",
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label="Input Text.",
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value="This model sounds really good and above all, it's reasonably fast.",
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value="This model sounds really good and above all, it's reasonably fast.",
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)
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)
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temperature = gr.Slider(
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tts_btn = gr.Button(value="Step 3 - Inference XTTS model")
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label="temperature", minimum=0.00001, maximum=1.0, step=0.05, value=0.75
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)
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tts_output_audio = gr.Audio(label="Generated Audio.")
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rms_norm_output = gr.Checkbox(
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reference_audio = gr.Audio(label="Reference audio used.")
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label="RMS norm output.", value=True, interactive=True
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)
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tts_btn = gr.Button(value="Step 2 - TTS")
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with gr.Column() as col3:
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tts_output_audio_no_enhanced = gr.Audio(label="HiFi-GAN.")
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tts_output_audio_no_enhanced_ft = gr.Audio(label="HiFi-GAN new.")
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reference_audio = gr.Audio(label="Reference Speech used.")
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def preprocess_dataset(audio_path, language, state_vars, progress=gr.Progress(track_tqdm=True)):
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def preprocess_dataset(audio_path, language, state_vars, progress=gr.Progress(track_tqdm=True)):
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# create a temp directory to save the dataset
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# create a temp directory to save the dataset
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@ -119,12 +219,12 @@ with gr.Blocks() as demo:
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# ToDo: raise an error
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# ToDo: raise an error
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pass
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pass
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else:
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else:
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train_meta, eval_meta = format_audio_list([audio_path], target_language=language, out_path=out_path, gradio_progress=progress)
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train_meta, eval_meta = format_audio_list([audio_path], target_language=language, out_path=out_path, gradio_progress=progress)
|
||||||
|
|
||||||
state_vars = {}
|
state_vars = {}
|
||||||
state_vars["train_csv"] = train_meta
|
state_vars["train_csv"] = train_meta
|
||||||
state_vars["eval_csv"] = eval_meta
|
state_vars["eval_csv"] = eval_meta
|
||||||
|
print(state_vars)
|
||||||
return "Dataset Processed!", state_vars
|
return "Dataset Processed!", state_vars
|
||||||
|
|
||||||
prompt_compute_btn.click(
|
prompt_compute_btn.click(
|
||||||
|
@ -135,23 +235,55 @@ with gr.Blocks() as demo:
|
||||||
state_vars,
|
state_vars,
|
||||||
],
|
],
|
||||||
outputs=[
|
outputs=[
|
||||||
voice_ready,
|
progress_data,
|
||||||
state_vars,
|
state_vars,
|
||||||
],
|
],
|
||||||
)
|
)
|
||||||
|
|
||||||
|
def train_model(language, num_epochs, batch_size, state_vars, output_path="./", progress=gr.Progress(track_tqdm=True)):
|
||||||
|
# state_vars = {'train_csv': '/tmp/tmprh4k_vou/metadata_train.csv', 'eval_csv': '/tmp/tmprh4k_vou/metadata_eval.csv'}
|
||||||
|
|
||||||
|
config_path, original_xtts_checkpoint, vocab_file, exp_path, speaker_wav = train_gpt(language, num_epochs, batch_size, state_vars["train_csv"], state_vars["eval_csv"], output_path=output_path)
|
||||||
|
# copy original files to avoid parameters changes issues
|
||||||
|
os.system(f"cp {config_path} {exp_path}")
|
||||||
|
os.system(f"cp {vocab_file} {exp_path}")
|
||||||
|
|
||||||
|
ft_xtts_checkpoint = os.path.join(exp_path, "best_model.pth")
|
||||||
|
state_vars["config_path"] = config_path
|
||||||
|
state_vars["original_xtts_checkpoint"] = original_xtts_checkpoint
|
||||||
|
state_vars["vocab_file"] = vocab_file
|
||||||
|
state_vars["ft_xtts_checkpoint"] = ft_xtts_checkpoint
|
||||||
|
state_vars["speaker_audio_file"] = speaker_wav
|
||||||
|
return "Model training done!", state_vars, config_path, vocab_file, ft_xtts_checkpoint, speaker_wav
|
||||||
|
|
||||||
|
|
||||||
|
train_btn.click(
|
||||||
|
fn=train_model,
|
||||||
|
inputs=[
|
||||||
|
lang,
|
||||||
|
num_epochs,
|
||||||
|
batch_size,
|
||||||
|
state_vars,
|
||||||
|
],
|
||||||
|
outputs=[progress_train, state_vars, xtts_config, xtts_vocab, xtts_checkpoint, speaker_reference_audio],
|
||||||
|
)
|
||||||
|
|
||||||
tts_btn.click(
|
tts_btn.click(
|
||||||
fn=run_tts,
|
fn=run_tts,
|
||||||
inputs=[
|
inputs=[
|
||||||
lang,
|
tts_language,
|
||||||
tts_text,
|
tts_text,
|
||||||
|
xtts_checkpoint,
|
||||||
|
xtts_config,
|
||||||
|
xtts_vocab,
|
||||||
|
speaker_reference_audio,
|
||||||
state_vars,
|
state_vars,
|
||||||
temperature,
|
|
||||||
rms_norm_output,
|
|
||||||
],
|
],
|
||||||
outputs=[tts_output_audio_no_enhanced, tts_output_audio_no_enhanced_ft],
|
outputs=[tts_output_audio, reference_audio],
|
||||||
)
|
)
|
||||||
|
|
||||||
|
|
||||||
|
|
||||||
if __name__ == "__main__":
|
if __name__ == "__main__":
|
||||||
demo.launch(
|
demo.launch(
|
||||||
share=True,
|
share=True,
|
||||||
|
|
|
@ -225,11 +225,11 @@ class GPTTrainer(BaseTTS):
|
||||||
|
|
||||||
@torch.no_grad()
|
@torch.no_grad()
|
||||||
def test_run(self, assets) -> Tuple[Dict, Dict]: # pylint: disable=W0613
|
def test_run(self, assets) -> Tuple[Dict, Dict]: # pylint: disable=W0613
|
||||||
|
test_audios = {}
|
||||||
if self.config.test_sentences:
|
if self.config.test_sentences:
|
||||||
# init gpt for inference mode
|
# init gpt for inference mode
|
||||||
self.xtts.gpt.init_gpt_for_inference(kv_cache=self.args.kv_cache, use_deepspeed=False)
|
self.xtts.gpt.init_gpt_for_inference(kv_cache=self.args.kv_cache, use_deepspeed=False)
|
||||||
self.xtts.gpt.eval()
|
self.xtts.gpt.eval()
|
||||||
test_audios = {}
|
|
||||||
print(" | > Synthesizing test sentences.")
|
print(" | > Synthesizing test sentences.")
|
||||||
for idx, s_info in enumerate(self.config.test_sentences):
|
for idx, s_info in enumerate(self.config.test_sentences):
|
||||||
wav = self.xtts.synthesize(
|
wav = self.xtts.synthesize(
|
||||||
|
|
Loading…
Reference in New Issue