import os import random from typing import Dict, List, Tuple, Union import torch import torch.distributed as dist from coqpit import Coqpit from torch import nn from torch.utils.data import DataLoader from torch.utils.data.distributed import DistributedSampler from TTS.model import BaseModel from TTS.tts.datasets.dataset import TTSDataset from TTS.tts.utils.languages import LanguageManager, get_language_weighted_sampler from TTS.tts.utils.speakers import SpeakerManager, get_speaker_manager, get_speaker_weighted_sampler from TTS.tts.utils.synthesis import synthesis from TTS.tts.utils.visual import plot_alignment, plot_spectrogram # pylint: skip-file class BaseTTS(BaseModel): """Base `tts` class. Every new `tts` model must inherit this. It defines common `tts` specific functions on top of `Model` implementation. Notes on input/output tensor shapes: Any input or output tensor of the model must be shaped as - 3D tensors `batch x time x channels` - 2D tensors `batch x channels` - 1D tensors `batch x 1` """ def __init__( self, config: Coqpit, ap: "AudioProcessor", tokenizer: "TTSTokenizer", speaker_manager: SpeakerManager = None ): super().__init__(config) self.config = config self.ap = ap self.tokenizer = tokenizer self.speaker_manager = speaker_manager self._set_model_args(config) def _set_model_args(self, config: Coqpit): """Setup model args based on the config type (`ModelConfig` or `ModelArgs`). `ModelArgs` has all the fields reuqired to initialize the model architecture. `ModelConfig` has all the fields required for training, inference and containes `ModelArgs`. If the config is for training with a name like "*Config", then the model args are embeded in the config.model_args If the config is for the model with a name like "*Args", then we assign the directly. """ # don't use isintance not to import recursively if "Config" in config.__class__.__name__: config_num_chars = ( self.config.model_args.num_chars if hasattr(self.config, "model_args") else self.config.num_chars ) num_chars = config_num_chars if self.tokenizer is None else self.tokenizer.characters.num_chars if "characters" in config: self.config.num_chars = num_chars if hasattr(self.config, "model_args"): config.model_args.num_chars = num_chars self.args = self.config.model_args else: self.config = config self.args = config.model_args elif "Args" in config.__class__.__name__: self.args = config else: raise ValueError("config must be either a *Config or *Args") def init_multispeaker(self, config: Coqpit, data: List = None): """Initialize a speaker embedding layer if needen and define expected embedding channel size for defining `in_channels` size of the connected layers. This implementation yields 3 possible outcomes: 1. If `config.use_speaker_embedding` and `config.use_d_vector_file are False, do nothing. 2. If `config.use_d_vector_file` is True, set expected embedding channel size to `config.d_vector_dim` or 512. 3. If `config.use_speaker_embedding`, initialize a speaker embedding layer with channel size of `config.d_vector_dim` or 512. You can override this function for new models. Args: config (Coqpit): Model configuration. """ # set number of speakers if self.speaker_manager is not None: self.num_speakers = self.speaker_manager.num_speakers elif hasattr(config, "num_speakers"): self.num_speakers = config.num_speakers # set ultimate speaker embedding size if config.use_speaker_embedding or config.use_d_vector_file: self.embedded_speaker_dim = ( config.d_vector_dim if "d_vector_dim" in config and config.d_vector_dim is not None else 512 ) # init speaker embedding layer if config.use_speaker_embedding and not config.use_d_vector_file: print(" > Init speaker_embedding layer.") self.speaker_embedding = nn.Embedding(self.num_speakers, self.embedded_speaker_dim) self.speaker_embedding.weight.data.normal_(0, 0.3) def get_aux_input(self, **kwargs) -> Dict: """Prepare and return `aux_input` used by `forward()`""" return {"speaker_id": None, "style_wav": None, "d_vector": None, "language_id": None} def get_aux_input_from_test_setences(self, sentence_info): if hasattr(self.config, "model_args"): config = self.config.model_args else: config = self.config # extract speaker and language info text, speaker_name, style_wav, language_name = None, None, None, None if isinstance(sentence_info, list): if len(sentence_info) == 1: text = sentence_info[0] elif len(sentence_info) == 2: text, speaker_name = sentence_info elif len(sentence_info) == 3: text, speaker_name, style_wav = sentence_info elif len(sentence_info) == 4: text, speaker_name, style_wav, language_name = sentence_info else: text = sentence_info # get speaker id/d_vector speaker_id, d_vector, language_id = None, None, None if hasattr(self, "speaker_manager"): if config.use_d_vector_file: if speaker_name is None: d_vector = self.speaker_manager.get_random_d_vector() else: d_vector = self.speaker_manager.get_d_vector_by_speaker(speaker_name) elif config.use_speaker_embedding: if speaker_name is None: speaker_id = self.speaker_manager.get_random_speaker_id() else: speaker_id = self.speaker_manager.speaker_ids[speaker_name] # get language id if hasattr(self, "language_manager") and config.use_language_embedding and language_name is not None: language_id = self.language_manager.language_id_mapping[language_name] return { "text": text, "speaker_id": speaker_id, "style_wav": style_wav, "d_vector": d_vector, "language_id": language_id, } def format_batch(self, batch: Dict) -> Dict: """Generic batch formatting for `TTSDataset`. You must override this if you use a custom dataset. Args: batch (Dict): [description] Returns: Dict: [description] """ # setup input batch text_input = batch["token_id"] text_lengths = batch["token_id_lengths"] speaker_names = batch["speaker_names"] linear_input = batch["linear"] mel_input = batch["mel"] mel_lengths = batch["mel_lengths"] stop_targets = batch["stop_targets"] item_idx = batch["item_idxs"] d_vectors = batch["d_vectors"] speaker_ids = batch["speaker_ids"] attn_mask = batch["attns"] waveform = batch["waveform"] pitch = batch["pitch"] language_ids = batch["language_ids"] max_text_length = torch.max(text_lengths.float()) max_spec_length = torch.max(mel_lengths.float()) # compute durations from attention masks durations = None if attn_mask is not None: durations = torch.zeros(attn_mask.shape[0], attn_mask.shape[2]) for idx, am in enumerate(attn_mask): # compute raw durations c_idxs = am[:, : text_lengths[idx], : mel_lengths[idx]].max(1)[1] # c_idxs, counts = torch.unique_consecutive(c_idxs, return_counts=True) c_idxs, counts = torch.unique(c_idxs, return_counts=True) dur = torch.ones([text_lengths[idx]]).to(counts.dtype) dur[c_idxs] = counts # smooth the durations and set any 0 duration to 1 # by cutting off from the largest duration indeces. extra_frames = dur.sum() - mel_lengths[idx] largest_idxs = torch.argsort(-dur)[:extra_frames] dur[largest_idxs] -= 1 assert ( dur.sum() == mel_lengths[idx] ), f" [!] total duration {dur.sum()} vs spectrogram length {mel_lengths[idx]}" durations[idx, : text_lengths[idx]] = dur # set stop targets wrt reduction factor stop_targets = stop_targets.view(text_input.shape[0], stop_targets.size(1) // self.config.r, -1) stop_targets = (stop_targets.sum(2) > 0.0).unsqueeze(2).float().squeeze(2) stop_target_lengths = torch.divide(mel_lengths, self.config.r).ceil_() return { "text_input": text_input, "text_lengths": text_lengths, "speaker_names": speaker_names, "mel_input": mel_input, "mel_lengths": mel_lengths, "linear_input": linear_input, "stop_targets": stop_targets, "stop_target_lengths": stop_target_lengths, "attn_mask": attn_mask, "durations": durations, "speaker_ids": speaker_ids, "d_vectors": d_vectors, "max_text_length": float(max_text_length), "max_spec_length": float(max_spec_length), "item_idx": item_idx, "waveform": waveform, "pitch": pitch, "language_ids": language_ids, } def get_data_loader( self, config: Coqpit, assets: Dict, is_eval: bool, samples: Union[List[Dict], List[List]], verbose: bool, num_gpus: int, rank: int = None, ) -> "DataLoader": if is_eval and not config.run_eval: loader = None else: # setup multi-speaker attributes if hasattr(self, "speaker_manager") and self.speaker_manager is not None: if hasattr(config, "model_args"): speaker_id_mapping = ( self.speaker_manager.speaker_ids if config.model_args.use_speaker_embedding else None ) d_vector_mapping = self.speaker_manager.d_vectors if config.model_args.use_d_vector_file else None config.use_d_vector_file = config.model_args.use_d_vector_file else: speaker_id_mapping = self.speaker_manager.speaker_ids if config.use_speaker_embedding else None d_vector_mapping = self.speaker_manager.d_vectors if config.use_d_vector_file else None else: speaker_id_mapping = None d_vector_mapping = None # setup multi-lingual attributes if hasattr(self, "language_manager"): language_id_mapping = ( self.language_manager.language_id_mapping if self.args.use_language_embedding else None ) else: language_id_mapping = None # init dataloader dataset = TTSDataset( outputs_per_step=config.r if "r" in config else 1, compute_linear_spec=config.model.lower() == "tacotron" or config.compute_linear_spec, compute_f0=config.get("compute_f0", False), f0_cache_path=config.get("f0_cache_path", None), samples=samples, ap=self.ap, return_wav=config.return_wav if "return_wav" in config else False, batch_group_size=0 if is_eval else config.batch_group_size * config.batch_size, min_text_len=config.min_text_len, max_text_len=config.max_text_len, min_audio_len=config.min_audio_len, max_audio_len=config.max_audio_len, phoneme_cache_path=config.phoneme_cache_path, precompute_num_workers=config.precompute_num_workers, use_noise_augment=False if is_eval else config.use_noise_augment, verbose=verbose, speaker_id_mapping=speaker_id_mapping, d_vector_mapping=d_vector_mapping if config.use_d_vector_file else None, tokenizer=self.tokenizer, language_id_mapping=language_id_mapping, ) # wait all the DDP process to be ready if num_gpus > 1: dist.barrier() # sort input sequences from short to long dataset.preprocess_samples() # sampler for DDP sampler = DistributedSampler(dataset) if num_gpus > 1 else None # Weighted samplers # TODO: make this DDP amenable assert not ( num_gpus > 1 and getattr(config, "use_language_weighted_sampler", False) ), "language_weighted_sampler is not supported with DistributedSampler" assert not ( num_gpus > 1 and getattr(config, "use_speaker_weighted_sampler", False) ), "speaker_weighted_sampler is not supported with DistributedSampler" if sampler is None: if getattr(config, "use_language_weighted_sampler", False): print(" > Using Language weighted sampler") sampler = get_language_weighted_sampler(dataset.samples) elif getattr(config, "use_speaker_weighted_sampler", False): print(" > Using Language weighted sampler") sampler = get_speaker_weighted_sampler(dataset.samples) loader = DataLoader( dataset, batch_size=config.eval_batch_size if is_eval else config.batch_size, shuffle=False, collate_fn=dataset.collate_fn, drop_last=False, sampler=sampler, num_workers=config.num_eval_loader_workers if is_eval else config.num_loader_workers, pin_memory=False, ) return loader def _get_test_aux_input( self, ) -> Dict: d_vector = None if self.config.use_d_vector_file: d_vector = [self.speaker_manager.d_vectors[name]["embedding"] for name in self.speaker_manager.d_vectors] d_vector = (random.sample(sorted(d_vector), 1),) aux_inputs = { "speaker_id": None if not self.config.use_speaker_embedding else random.sample(sorted(self.speaker_manager.speaker_ids.values()), 1), "d_vector": d_vector, "style_wav": None, # TODO: handle GST style input } return aux_inputs def test_run(self, assets: Dict) -> Tuple[Dict, Dict]: """Generic test run for `tts` models used by `Trainer`. You can override this for a different behaviour. Args: assets (dict): A dict of training assets. For `tts` models, it must include `{'audio_processor': ap}`. Returns: Tuple[Dict, Dict]: Test figures and audios to be projected to Tensorboard. """ print(" | > Synthesizing test sentences.") test_audios = {} test_figures = {} test_sentences = self.config.test_sentences aux_inputs = self._get_test_aux_input() for idx, sen in enumerate(test_sentences): outputs_dict = synthesis( self, sen, self.config, "cuda" in str(next(self.parameters()).device), speaker_id=aux_inputs["speaker_id"], d_vector=aux_inputs["d_vector"], style_wav=aux_inputs["style_wav"], use_griffin_lim=True, do_trim_silence=False, ) test_audios["{}-audio".format(idx)] = outputs_dict["wav"] test_figures["{}-prediction".format(idx)] = plot_spectrogram( outputs_dict["outputs"]["model_outputs"], self.ap, output_fig=False ) test_figures["{}-alignment".format(idx)] = plot_alignment( outputs_dict["outputs"]["alignments"], output_fig=False ) return test_figures, test_audios def on_init_start(self, trainer): """Save the speaker.json and language_ids.json at the beginning of the training. Also update both paths.""" if self.speaker_manager is not None: output_path = os.path.join(trainer.output_path, "speakers.json") self.speaker_manager.save_speaker_ids_to_file(output_path) trainer.config.speakers_file = output_path # some models don't have `model_args` set if hasattr(trainer.config, "model_args"): trainer.config.model_args.speakers_file = output_path trainer.config.save_json(os.path.join(trainer.output_path, "config.json")) print(f" > `speakers.json` is saved to {output_path}.") print(" > `speakers_file` is updated in the config.json.") if hasattr(self, "language_manager") and self.language_manager is not None: output_path = os.path.join(trainer.output_path, "language_ids.json") self.language_manager.save_language_ids_to_file(output_path) trainer.config.language_ids_file = output_path if hasattr(trainer.config, "model_args"): trainer.config.model_args.language_ids_file = output_path trainer.config.save_json(os.path.join(trainer.output_path, "config.json")) print(f" > `language_ids.json` is saved to {output_path}.") print(" > `language_ids_file` is updated in the config.json.")