update dependency version and add speed/sample rate and vocoder args

master
Malar 2019-11-28 16:57:36 +05:30
parent ac5ffcf6d5
commit 009b87e716
3 changed files with 37 additions and 20 deletions

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@ -12,12 +12,12 @@ with open("HISTORY.rst") as history_file:
requirements = [ requirements = [
"klepto==0.1.6", "klepto==0.1.6",
"numpy==1.16.4", "numpy~=1.16.4",
"inflect==0.2.5", "inflect==0.2.5",
"librosa==0.6.0", "librosa==0.6.0",
"scipy==1.3.0", "scipy~=1.3.0",
"Unidecode==1.0.22", "Unidecode==1.0.22",
"torch==1.1.0", "torch~=1.1.0",
"PyAudio==0.2.11" "PyAudio==0.2.11"
] ]
@ -53,7 +53,7 @@ setup(
test_suite="tests", test_suite="tests",
tests_require=test_requirements, tests_require=test_requirements,
url="https://github.com/malarinv/tacotron2", url="https://github.com/malarinv/tacotron2",
version="0.2.0", version="0.3.0",
zip_safe=False, zip_safe=False,
entry_points={"console_scripts": ("tts_debug = taco2.tts:main",)}, entry_points={"console_scripts": ("tts_debug = taco2.tts:main",)},
) )

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@ -35,7 +35,7 @@ class HParams(object):
# Audio Parameters # # Audio Parameters #
################################ ################################
max_wav_value = 32768.0 max_wav_value = 32768.0
sampling_rate = 16000 sampling_rate = 22050
filter_length = 1024 filter_length = 1024
hop_length = 256 hop_length = 256
win_length = 1024 win_length = 1024

View File

@ -15,9 +15,9 @@ from .text import text_to_sequence
from .denoiser import Denoiser from .denoiser import Denoiser
from .audio_processing import griffin_lim, postprocess_audio from .audio_processing import griffin_lim, postprocess_audio
TTS_SAMPLE_RATE = 22050
OUTPUT_SAMPLE_RATE = 22050 OUTPUT_SAMPLE_RATE = 22050
# OUTPUT_SAMPLE_RATE = 16000 GL_ITERS = 30
VOCODER_MODEL = "wavglow"
# config from # config from
# https://github.com/NVIDIA/waveglow/blob/master/config.json # https://github.com/NVIDIA/waveglow/blob/master/config.json
@ -37,7 +37,7 @@ class TTSModel(object):
def __init__(self, tacotron2_path, waveglow_path, **kwargs): def __init__(self, tacotron2_path, waveglow_path, **kwargs):
super(TTSModel, self).__init__() super(TTSModel, self).__init__()
hparams = HParams(**kwargs) hparams = HParams(**kwargs)
hparams.sampling_rate = TTS_SAMPLE_RATE self.hparams = hparams
self.model = Tacotron2(hparams) self.model = Tacotron2(hparams)
if torch.cuda.is_available(): if torch.cuda.is_available():
self.model.load_state_dict(torch.load(tacotron2_path)["state_dict"]) self.model.load_state_dict(torch.load(tacotron2_path)["state_dict"])
@ -78,7 +78,7 @@ class TTSModel(object):
) )
else: else:
self.synth_speech = klepto.safe.inf_cache(cache=self.k_cache)( self.synth_speech = klepto.safe.inf_cache(cache=self.k_cache)(
self.synth_speech_gl self.synth_speech_fast
) )
self.taco_stft = TacotronSTFT( self.taco_stft = TacotronSTFT(
hparams.filter_length, hparams.filter_length,
@ -101,23 +101,40 @@ class TTSModel(object):
) )
return mel_outputs_postnet return mel_outputs_postnet
def synth_speech_array(self, text): def synth_speech_array(self, text, vocoder):
mel_outputs_postnet = self.generate_mel_postnet(text) mel_outputs_postnet = self.generate_mel_postnet(text)
if method == "wavglow":
with torch.no_grad(): with torch.no_grad():
audio_t = self.waveglow.infer(mel_outputs_postnet, sigma=0.666) audio_t = self.waveglow.infer(mel_outputs_postnet, sigma=0.666)
audio_t = self.denoiser(audio_t, 0.1)[0] audio_t = self.denoiser(audio_t, 0.1)[0]
audio = audio_t[0].data.cpu().numpy() audio = audio_t[0].data.cpu().numpy()
elif method == "gl":
mel_decompress = self.taco_stft.spectral_de_normalize(mel_outputs_postnet)
mel_decompress = mel_decompress.transpose(1, 2).data.cpu()
spec_from_mel_scaling = 1000
spec_from_mel = torch.mm(mel_decompress[0], self.taco_stft.mel_basis)
spec_from_mel = spec_from_mel.transpose(0, 1).unsqueeze(0)
spec_from_mel = spec_from_mel * spec_from_mel_scaling
audio = griffin_lim(
torch.autograd.Variable(spec_from_mel[:, :, :-1]),
self.taco_stft.stft_fn,
60,
)
audio = audio.squeeze()
audio = audio.cpu().numpy()
else:
raise ValueError("vocoder arg should be one of [wavglow|gl]")
return audio return audio
def synth_speech(self, text): def synth_speech(self, text, speed: 1.0, sample_rate=OUTPUT_SAMPLE_RATE):
audio = self.synth_speech_array(text) audio = self.synth_speech_array(text, VOCODER_MODEL)
return postprocess_audio( return postprocess_audio(
audio, src_rate=TTS_SAMPLE_RATE, dst_rate=OUTPUT_SAMPLE_RATE audio, src_rate=self.hparams.sample_rate, dst_rate=sample_rate, tempo=speed
) )
def synth_speech_gl(self, text, griffin_iters=60): def synth_speech_fast(self, text, speed: 1.0, sample_rate=OUTPUT_SAMPLE_RATE):
mel_outputs_postnet = self.generate_mel_postnet(text) mel_outputs_postnet = self.generate_mel_postnet(text)
mel_decompress = self.taco_stft.spectral_de_normalize(mel_outputs_postnet) mel_decompress = self.taco_stft.spectral_de_normalize(mel_outputs_postnet)
@ -129,13 +146,13 @@ class TTSModel(object):
audio = griffin_lim( audio = griffin_lim(
torch.autograd.Variable(spec_from_mel[:, :, :-1]), torch.autograd.Variable(spec_from_mel[:, :, :-1]),
self.taco_stft.stft_fn, self.taco_stft.stft_fn,
griffin_iters, GL_ITERS,
) )
audio = audio.squeeze() audio = audio.squeeze()
audio = audio.cpu().numpy() audio = audio.cpu().numpy()
return postprocess_audio( return postprocess_audio(
audio, tempo=0.6, src_rate=TTS_SAMPLE_RATE, dst_rate=OUTPUT_SAMPLE_RATE audio, tempo=speed, src_rate=self.hparams.sample_rate, dst_rate=sample_rate,
) )