mirror of https://github.com/coqui-ai/TTS.git
289 lines
9.0 KiB
Python
289 lines
9.0 KiB
Python
from typing import Dict
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import numpy as np
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import torch
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from torch import nn
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def numpy_to_torch(np_array, dtype, cuda=False):
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if np_array is None:
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return None
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tensor = torch.as_tensor(np_array, dtype=dtype)
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if cuda:
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return tensor.cuda()
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return tensor
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def compute_style_mel(style_wav, ap, cuda=False):
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style_mel = torch.FloatTensor(ap.melspectrogram(ap.load_wav(style_wav, sr=ap.sample_rate))).unsqueeze(0)
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if cuda:
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return style_mel.cuda()
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return style_mel
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def run_model_torch(
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model: nn.Module,
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inputs: torch.Tensor,
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speaker_id: int = None,
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style_mel: torch.Tensor = None,
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d_vector: torch.Tensor = None,
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language_id: torch.Tensor = None,
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) -> Dict:
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"""Run a torch model for inference. It does not support batch inference.
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Args:
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model (nn.Module): The model to run inference.
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inputs (torch.Tensor): Input tensor with character ids.
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speaker_id (int, optional): Input speaker ids for multi-speaker models. Defaults to None.
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style_mel (torch.Tensor, optional): Spectrograms used for voice styling . Defaults to None.
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d_vector (torch.Tensor, optional): d-vector for multi-speaker models . Defaults to None.
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Returns:
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Dict: model outputs.
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"""
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input_lengths = torch.tensor(inputs.shape[1:2]).to(inputs.device)
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if hasattr(model, "module"):
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_func = model.module.inference
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else:
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_func = model.inference
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outputs = _func(
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inputs,
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aux_input={
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"x_lengths": input_lengths,
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"speaker_ids": speaker_id,
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"d_vectors": d_vector,
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"style_mel": style_mel,
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"language_ids": language_id,
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},
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)
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return outputs
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def trim_silence(wav, ap):
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return wav[: ap.find_endpoint(wav)]
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def inv_spectrogram(postnet_output, ap, CONFIG):
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if CONFIG.model.lower() in ["tacotron"]:
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wav = ap.inv_spectrogram(postnet_output.T)
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else:
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wav = ap.inv_melspectrogram(postnet_output.T)
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return wav
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def id_to_torch(aux_id, cuda=False):
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if aux_id is not None:
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aux_id = np.asarray(aux_id)
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aux_id = torch.from_numpy(aux_id)
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if cuda:
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return aux_id.cuda()
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return aux_id
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def embedding_to_torch(d_vector, cuda=False):
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if d_vector is not None:
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d_vector = np.asarray(d_vector)
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d_vector = torch.from_numpy(d_vector).type(torch.FloatTensor)
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d_vector = d_vector.squeeze().unsqueeze(0)
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if cuda:
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return d_vector.cuda()
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return d_vector
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# TODO: perform GL with pytorch for batching
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def apply_griffin_lim(inputs, input_lens, CONFIG, ap):
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"""Apply griffin-lim to each sample iterating throught the first dimension.
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Args:
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inputs (Tensor or np.Array): Features to be converted by GL. First dimension is the batch size.
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input_lens (Tensor or np.Array): 1D array of sample lengths.
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CONFIG (Dict): TTS config.
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ap (AudioProcessor): TTS audio processor.
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"""
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wavs = []
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for idx, spec in enumerate(inputs):
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wav_len = (input_lens[idx] * ap.hop_length) - ap.hop_length # inverse librosa padding
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wav = inv_spectrogram(spec, ap, CONFIG)
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# assert len(wav) == wav_len, f" [!] wav lenght: {len(wav)} vs expected: {wav_len}"
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wavs.append(wav[:wav_len])
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return wavs
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def synthesis(
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model,
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text,
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CONFIG,
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use_cuda,
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speaker_id=None,
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style_wav=None,
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use_griffin_lim=False,
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do_trim_silence=False,
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d_vector=None,
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language_id=None,
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):
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"""Synthesize voice for the given text using Griffin-Lim vocoder or just compute output features to be passed to
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the vocoder model.
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Args:
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model (TTS.tts.models):
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The TTS model to synthesize audio with.
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text (str):
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The input text to convert to speech.
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CONFIG (Coqpit):
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Model configuration.
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use_cuda (bool):
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Enable/disable CUDA.
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speaker_id (int):
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Speaker ID passed to the speaker embedding layer in multi-speaker model. Defaults to None.
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style_wav (str | Dict[str, float]):
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Path or tensor to/of a waveform used for computing the style embedding. Defaults to None.
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enable_eos_bos_chars (bool):
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enable special chars for end of sentence and start of sentence. Defaults to False.
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do_trim_silence (bool):
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trim silence after synthesis. Defaults to False.
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d_vector (torch.Tensor):
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d-vector for multi-speaker models in share :math:`[1, D]`. Defaults to None.
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language_id (int):
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Language ID passed to the language embedding layer in multi-langual model. Defaults to None.
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"""
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# GST processing
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style_mel = None
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if CONFIG.has("gst") and CONFIG.gst and style_wav is not None:
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if isinstance(style_wav, dict):
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style_mel = style_wav
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else:
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style_mel = compute_style_mel(style_wav, model.ap, cuda=use_cuda)
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# convert text to sequence of token IDs
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text_inputs = np.asarray(
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model.tokenizer.text_to_ids(text, language=language_id),
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dtype=np.int32,
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)
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# pass tensors to backend
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if speaker_id is not None:
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speaker_id = id_to_torch(speaker_id, cuda=use_cuda)
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if d_vector is not None:
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d_vector = embedding_to_torch(d_vector, cuda=use_cuda)
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if language_id is not None:
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language_id = id_to_torch(language_id, cuda=use_cuda)
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if not isinstance(style_mel, dict):
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style_mel = numpy_to_torch(style_mel, torch.float, cuda=use_cuda)
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text_inputs = numpy_to_torch(text_inputs, torch.long, cuda=use_cuda)
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text_inputs = text_inputs.unsqueeze(0)
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# synthesize voice
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outputs = run_model_torch(model, text_inputs, speaker_id, style_mel, d_vector=d_vector, language_id=language_id)
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model_outputs = outputs["model_outputs"]
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model_outputs = model_outputs[0].data.cpu().numpy()
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alignments = outputs["alignments"]
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# convert outputs to numpy
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# plot results
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wav = None
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model_outputs = model_outputs.squeeze()
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if model_outputs.ndim == 2: # [T, C_spec]
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if use_griffin_lim:
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wav = inv_spectrogram(model_outputs, model.ap, CONFIG)
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# trim silence
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if do_trim_silence:
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wav = trim_silence(wav, model.ap)
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else: # [T,]
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wav = model_outputs
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return_dict = {
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"wav": wav,
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"alignments": alignments,
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"text_inputs": text_inputs,
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"outputs": outputs,
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}
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return return_dict
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def transfer_voice(
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model,
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CONFIG,
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use_cuda,
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reference_wav,
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speaker_id=None,
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d_vector=None,
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reference_speaker_id=None,
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reference_d_vector=None,
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do_trim_silence=False,
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use_griffin_lim=False,
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):
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"""Synthesize voice for the given text using Griffin-Lim vocoder or just compute output features to be passed to
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the vocoder model.
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Args:
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model (TTS.tts.models):
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The TTS model to synthesize audio with.
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CONFIG (Coqpit):
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Model configuration.
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use_cuda (bool):
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Enable/disable CUDA.
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reference_wav (str):
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Path of reference_wav to be used to voice conversion.
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speaker_id (int):
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Speaker ID passed to the speaker embedding layer in multi-speaker model. Defaults to None.
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d_vector (torch.Tensor):
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d-vector for multi-speaker models in share :math:`[1, D]`. Defaults to None.
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reference_speaker_id (int):
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Reference Speaker ID passed to the speaker embedding layer in multi-speaker model. Defaults to None.
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reference_d_vector (torch.Tensor):
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Reference d-vector for multi-speaker models in share :math:`[1, D]`. Defaults to None.
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enable_eos_bos_chars (bool):
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enable special chars for end of sentence and start of sentence. Defaults to False.
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do_trim_silence (bool):
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trim silence after synthesis. Defaults to False.
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"""
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# pass tensors to backend
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if speaker_id is not None:
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speaker_id = id_to_torch(speaker_id, cuda=use_cuda)
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if d_vector is not None:
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d_vector = embedding_to_torch(d_vector, cuda=use_cuda)
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if reference_d_vector is not None:
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reference_d_vector = embedding_to_torch(reference_d_vector, cuda=use_cuda)
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# load reference_wav audio
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reference_wav = embedding_to_torch(model.ap.load_wav(reference_wav, sr=model.ap.sample_rate), cuda=use_cuda)
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if hasattr(model, "module"):
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_func = model.module.inference_voice_conversion
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else:
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_func = model.inference_voice_conversion
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model_outputs = _func(reference_wav, speaker_id, d_vector, reference_speaker_id, reference_d_vector)
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# convert outputs to numpy
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# plot results
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wav = None
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model_outputs = model_outputs.squeeze()
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if model_outputs.ndim == 2: # [T, C_spec]
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if use_griffin_lim:
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wav = inv_spectrogram(model_outputs, model.ap, CONFIG)
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# trim silence
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if do_trim_silence:
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wav = trim_silence(wav, model.ap)
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else: # [T,]
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wav = model_outputs
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return wav
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