voice-xtts2 / TTS /tts /utils /synthesis.py
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changes in flenema
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import pkg_resources
installed = {pkg.key for pkg in pkg_resources.working_set} #pylint: disable=not-an-iterable
if 'tensorflow' in installed or 'tensorflow-gpu' in installed:
import tensorflow as tf
import torch
import numpy as np
from .text import text_to_sequence, phoneme_to_sequence
def text_to_seqvec(text, CONFIG):
text_cleaner = [CONFIG.text_cleaner]
# text ot phonemes to sequence vector
if CONFIG.use_phonemes:
seq = np.asarray(
phoneme_to_sequence(text, text_cleaner, CONFIG.phoneme_language,
CONFIG.enable_eos_bos_chars,
tp=CONFIG.characters if 'characters' in CONFIG.keys() else None,
add_blank=CONFIG['add_blank'] if 'add_blank' in CONFIG.keys() else False),
dtype=np.int32)
else:
seq = np.asarray(
text_to_sequence(text, text_cleaner, tp=CONFIG.characters if 'characters' in CONFIG.keys() else None,
add_blank=CONFIG['add_blank'] if 'add_blank' in CONFIG.keys() else False), dtype=np.int32)
return seq
def numpy_to_torch(np_array, dtype, cuda=False):
if np_array is None:
return None
tensor = torch.as_tensor(np_array, dtype=dtype)
if cuda:
return tensor.cuda()
return tensor
def numpy_to_tf(np_array, dtype):
if np_array is None:
return None
tensor = tf.convert_to_tensor(np_array, dtype=dtype)
return tensor
def compute_style_mel(style_wav, ap, cuda=False):
style_mel = torch.FloatTensor(ap.melspectrogram(
ap.load_wav(style_wav, sr=ap.sample_rate))).unsqueeze(0)
if cuda:
return style_mel.cuda()
return style_mel
def run_model_torch(model, inputs, CONFIG, truncated, speaker_id=None, style_mel=None, speaker_embeddings=None):
if 'tacotron' in CONFIG.model.lower():
if CONFIG.use_gst:
decoder_output, postnet_output, alignments, stop_tokens = model.inference(
inputs, style_mel=style_mel, speaker_ids=speaker_id, speaker_embeddings=speaker_embeddings)
else:
if truncated:
decoder_output, postnet_output, alignments, stop_tokens = model.inference_truncated(
inputs, speaker_ids=speaker_id, speaker_embeddings=speaker_embeddings)
else:
decoder_output, postnet_output, alignments, stop_tokens = model.inference(
inputs, speaker_ids=speaker_id, speaker_embeddings=speaker_embeddings)
elif 'glow' in CONFIG.model.lower():
inputs_lengths = torch.tensor(inputs.shape[1:2]).to(inputs.device) # pylint: disable=not-callable
if hasattr(model, 'module'):
# distributed model
postnet_output, _, _, _, alignments, _, _ = model.module.inference(inputs, inputs_lengths, g=speaker_id if speaker_id is not None else speaker_embeddings)
else:
postnet_output, _, _, _, alignments, _, _ = model.inference(inputs, inputs_lengths, g=speaker_id if speaker_id is not None else speaker_embeddings)
postnet_output = postnet_output.permute(0, 2, 1)
# these only belong to tacotron models.
decoder_output = None
stop_tokens = None
elif 'speedy_speech' in CONFIG.model.lower():
inputs_lengths = torch.tensor(inputs.shape[1:2]).to(inputs.device) # pylint: disable=not-callable
if hasattr(model, 'module'):
# distributed model
postnet_output, alignments= model.module.inference(inputs, inputs_lengths, g=speaker_id if speaker_id is not None else speaker_embeddings)
else:
postnet_output, alignments= model.inference(inputs, inputs_lengths, g=speaker_id if speaker_id is not None else speaker_embeddings)
postnet_output = postnet_output.permute(0, 2, 1)
# these only belong to tacotron models.
decoder_output = None
stop_tokens = None
return decoder_output, postnet_output, alignments, stop_tokens
def run_model_tf(model, inputs, CONFIG, truncated, speaker_id=None, style_mel=None):
if CONFIG.use_gst and style_mel is not None:
raise NotImplementedError(' [!] GST inference not implemented for TF')
if truncated:
raise NotImplementedError(' [!] Truncated inference not implemented for TF')
if speaker_id is not None:
raise NotImplementedError(' [!] Multi-Speaker not implemented for TF')
# TODO: handle multispeaker case
decoder_output, postnet_output, alignments, stop_tokens = model(
inputs, training=False)
return decoder_output, postnet_output, alignments, stop_tokens
def run_model_tflite(model, inputs, CONFIG, truncated, speaker_id=None, style_mel=None):
if CONFIG.use_gst and style_mel is not None:
raise NotImplementedError(' [!] GST inference not implemented for TfLite')
if truncated:
raise NotImplementedError(' [!] Truncated inference not implemented for TfLite')
if speaker_id is not None:
raise NotImplementedError(' [!] Multi-Speaker not implemented for TfLite')
# get input and output details
input_details = model.get_input_details()
output_details = model.get_output_details()
# reshape input tensor for the new input shape
model.resize_tensor_input(input_details[0]['index'], inputs.shape)
model.allocate_tensors()
detail = input_details[0]
# input_shape = detail['shape']
model.set_tensor(detail['index'], inputs)
# run the model
model.invoke()
# collect outputs
decoder_output = model.get_tensor(output_details[0]['index'])
postnet_output = model.get_tensor(output_details[1]['index'])
# tflite model only returns feature frames
return decoder_output, postnet_output, None, None
def parse_outputs_torch(postnet_output, decoder_output, alignments, stop_tokens):
postnet_output = postnet_output[0].data.cpu().numpy()
decoder_output = None if decoder_output is None else decoder_output[0].data.cpu().numpy()
alignment = alignments[0].cpu().data.numpy()
stop_tokens = None if stop_tokens is None else stop_tokens[0].cpu().numpy()
return postnet_output, decoder_output, alignment, stop_tokens
def parse_outputs_tf(postnet_output, decoder_output, alignments, stop_tokens):
postnet_output = postnet_output[0].numpy()
decoder_output = decoder_output[0].numpy()
alignment = alignments[0].numpy()
stop_tokens = stop_tokens[0].numpy()
return postnet_output, decoder_output, alignment, stop_tokens
def parse_outputs_tflite(postnet_output, decoder_output):
postnet_output = postnet_output[0]
decoder_output = decoder_output[0]
return postnet_output, decoder_output
def trim_silence(wav, ap):
return wav[:ap.find_endpoint(wav)]
def inv_spectrogram(postnet_output, ap, CONFIG):
if CONFIG.model.lower() in ["tacotron"]:
wav = ap.inv_spectrogram(postnet_output.T)
else:
wav = ap.inv_melspectrogram(postnet_output.T)
return wav
def id_to_torch(speaker_id, cuda=False):
if speaker_id is not None:
speaker_id = np.asarray(speaker_id)
# TODO: test this for tacotron models
speaker_id = torch.from_numpy(speaker_id)
if cuda:
return speaker_id.cuda()
return speaker_id
def embedding_to_torch(speaker_embedding, cuda=False):
if speaker_embedding is not None:
speaker_embedding = np.asarray(speaker_embedding)
speaker_embedding = torch.from_numpy(speaker_embedding).unsqueeze(0).type(torch.FloatTensor)
if cuda:
return speaker_embedding.cuda()
return speaker_embedding
# TODO: perform GL with pytorch for batching
def apply_griffin_lim(inputs, input_lens, CONFIG, ap):
'''Apply griffin-lim to each sample iterating throught the first dimension.
Args:
inputs (Tensor or np.Array): Features to be converted by GL. First dimension is the batch size.
input_lens (Tensor or np.Array): 1D array of sample lengths.
CONFIG (Dict): TTS config.
ap (AudioProcessor): TTS audio processor.
'''
wavs = []
for idx, spec in enumerate(inputs):
wav_len = (input_lens[idx] * ap.hop_length) - ap.hop_length # inverse librosa padding
wav = inv_spectrogram(spec, ap, CONFIG)
# assert len(wav) == wav_len, f" [!] wav lenght: {len(wav)} vs expected: {wav_len}"
wavs.append(wav[:wav_len])
return wavs
def synthesis(model,
text,
CONFIG,
use_cuda,
ap,
speaker_id=None,
style_wav=None,
truncated=False,
enable_eos_bos_chars=False, #pylint: disable=unused-argument
use_griffin_lim=False,
do_trim_silence=False,
speaker_embedding=None,
backend='torch'):
"""Synthesize voice for the given text.
Args:
model (TTS.tts.models): model to synthesize.
text (str): target text
CONFIG (dict): config dictionary to be loaded from config.json.
use_cuda (bool): enable cuda.
ap (TTS.tts.utils.audio.AudioProcessor): audio processor to process
model outputs.
speaker_id (int): id of speaker
style_wav (str): Uses for style embedding of GST.
truncated (bool): keep model states after inference. It can be used
for continuous inference at long texts.
enable_eos_bos_chars (bool): enable special chars for end of sentence and start of sentence.
do_trim_silence (bool): trim silence after synthesis.
backend (str): tf or torch
"""
# GST processing
style_mel = None
if 'use_gst' in CONFIG.keys() and CONFIG.use_gst and style_wav is not None:
if isinstance(style_wav, dict):
style_mel = style_wav
else:
style_mel = compute_style_mel(style_wav, ap, cuda=use_cuda)
# preprocess the given text
inputs = text_to_seqvec(text, CONFIG)
# pass tensors to backend
if backend == 'torch':
if speaker_id is not None:
speaker_id = id_to_torch(speaker_id, cuda=use_cuda)
if speaker_embedding is not None:
speaker_embedding = embedding_to_torch(speaker_embedding, cuda=use_cuda)
if not isinstance(style_mel, dict):
style_mel = numpy_to_torch(style_mel, torch.float, cuda=use_cuda)
inputs = numpy_to_torch(inputs, torch.long, cuda=use_cuda)
inputs = inputs.unsqueeze(0)
elif backend == 'tf':
# TODO: handle speaker id for tf model
style_mel = numpy_to_tf(style_mel, tf.float32)
inputs = numpy_to_tf(inputs, tf.int32)
inputs = tf.expand_dims(inputs, 0)
elif backend == 'tflite':
style_mel = numpy_to_tf(style_mel, tf.float32)
inputs = numpy_to_tf(inputs, tf.int32)
inputs = tf.expand_dims(inputs, 0)
# synthesize voice
if backend == 'torch':
decoder_output, postnet_output, alignments, stop_tokens = run_model_torch(
model, inputs, CONFIG, truncated, speaker_id, style_mel, speaker_embeddings=speaker_embedding)
postnet_output, decoder_output, alignment, stop_tokens = parse_outputs_torch(
postnet_output, decoder_output, alignments, stop_tokens)
elif backend == 'tf':
decoder_output, postnet_output, alignments, stop_tokens = run_model_tf(
model, inputs, CONFIG, truncated, speaker_id, style_mel)
postnet_output, decoder_output, alignment, stop_tokens = parse_outputs_tf(
postnet_output, decoder_output, alignments, stop_tokens)
elif backend == 'tflite':
decoder_output, postnet_output, alignment, stop_tokens = run_model_tflite(
model, inputs, CONFIG, truncated, speaker_id, style_mel)
postnet_output, decoder_output = parse_outputs_tflite(
postnet_output, decoder_output)
# convert outputs to numpy
# plot results
wav = None
if use_griffin_lim:
wav = inv_spectrogram(postnet_output, ap, CONFIG)
# trim silence
if do_trim_silence:
wav = trim_silence(wav, ap)
return wav, alignment, decoder_output, postnet_output, stop_tokens, inputs