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mirror of https://github.com/malarinv/tacotron2 synced 2026-03-08 09:42:34 +00:00

11 Commits

Author SHA1 Message Date
6d3679d760 relax scipy version 2021-07-02 23:21:45 +05:30
a851e80db2 1. added rpyc server optional package
2. updated path variable
2020-03-05 16:59:55 +05:30
cb0c8ddd06 1. make pyaudio as extra requirement
2. warn if pyaudio not installed if player_gen is used
2020-01-22 14:09:37 +05:30
42a85d177e added glow as module 2019-11-29 00:20:01 +05:30
5efb1e2758 1. fix synth_speech_fast interface
2. rename private methods
2019-11-28 18:11:47 +05:30
ea11c5199e tested gl/wavglow working 2019-11-28 18:00:52 +05:30
78eed2d295 bugfix for vocoder arg 2019-11-28 17:03:46 +05:30
009b87e716 update dependency version and add speed/sample rate and vocoder args 2019-11-28 16:57:36 +05:30
ac5ffcf6d5 enable gpu support if available 2019-11-27 22:53:41 +05:30
5a30069f0a update tempo and output sample rate 2019-10-09 17:25:51 +05:30
dcc9ab3625 add tts cli args 2019-10-09 16:23:21 +05:30
10 changed files with 282 additions and 98 deletions

View File

@@ -29,13 +29,14 @@ import torch
from torch.autograd import Variable
import torch.nn.functional as F
DEVICE = torch.device('cuda:0' if torch.cuda.is_available() else 'cpu')
@torch.jit.script
def fused_add_tanh_sigmoid_multiply(input_a, input_b, n_channels):
n_channels_int = n_channels[0]
in_act = input_a+input_b
t_act = torch.tanh(in_act[:, :n_channels_int, :])
s_act = torch.sigmoid(in_act[:, n_channels_int:, :])
t_act = torch.nn.functional.tanh(in_act[:, :n_channels_int, :])
s_act = torch.nn.functional.sigmoid(in_act[:, n_channels_int:, :])
acts = t_act * s_act
return acts
@@ -90,7 +91,7 @@ class Invertible1x1Conv(torch.nn.Module):
# Reverse computation
W_inverse = W.float().inverse()
W_inverse = Variable(W_inverse[..., None])
if z.type() == 'torch.HalfTensor':
if z.type() == 'torch.cuda.HalfTensor' or z.type() == 'torch.HalfTensor':
W_inverse = W_inverse.half()
self.W_inverse = W_inverse
z = F.conv1d(z, self.W_inverse, bias=None, stride=1, padding=0)
@@ -117,6 +118,7 @@ class WN(torch.nn.Module):
self.n_channels = n_channels
self.in_layers = torch.nn.ModuleList()
self.res_skip_layers = torch.nn.ModuleList()
self.cond_layers = torch.nn.ModuleList()
start = torch.nn.Conv1d(n_in_channels, n_channels, 1)
start = torch.nn.utils.weight_norm(start, name='weight')
@@ -129,9 +131,6 @@ class WN(torch.nn.Module):
end.bias.data.zero_()
self.end = end
cond_layer = torch.nn.Conv1d(n_mel_channels, 2*n_channels*n_layers, 1)
self.cond_layer = torch.nn.utils.weight_norm(cond_layer, name='weight')
for i in range(n_layers):
dilation = 2 ** i
padding = int((kernel_size*dilation - dilation)/2)
@@ -140,6 +139,9 @@ class WN(torch.nn.Module):
in_layer = torch.nn.utils.weight_norm(in_layer, name='weight')
self.in_layers.append(in_layer)
cond_layer = torch.nn.Conv1d(n_mel_channels, 2*n_channels, 1)
cond_layer = torch.nn.utils.weight_norm(cond_layer, name='weight')
self.cond_layers.append(cond_layer)
# last one is not necessary
if i < n_layers - 1:
@@ -153,25 +155,24 @@ class WN(torch.nn.Module):
def forward(self, forward_input):
audio, spect = forward_input
audio = self.start(audio)
output = torch.zeros_like(audio)
n_channels_tensor = torch.IntTensor([self.n_channels])
spect = self.cond_layer(spect)
for i in range(self.n_layers):
spect_offset = i*2*self.n_channels
acts = fused_add_tanh_sigmoid_multiply(
self.in_layers[i](audio),
spect[:,spect_offset:spect_offset+2*self.n_channels,:],
n_channels_tensor)
self.cond_layers[i](spect),
torch.IntTensor([self.n_channels]))
res_skip_acts = self.res_skip_layers[i](acts)
if i < self.n_layers - 1:
audio = audio + res_skip_acts[:,:self.n_channels,:]
output = output + res_skip_acts[:,self.n_channels:,:]
audio = res_skip_acts[:,:self.n_channels,:] + audio
skip_acts = res_skip_acts[:,self.n_channels:,:]
else:
output = output + res_skip_acts
skip_acts = res_skip_acts
if i == 0:
output = skip_acts
else:
output = skip_acts + output
return self.end(output)
@@ -257,6 +258,16 @@ class WaveGlow(torch.nn.Module):
spect = spect.unfold(2, self.n_group, self.n_group).permute(0, 2, 1, 3)
spect = spect.contiguous().view(spect.size(0), spect.size(1), -1).permute(0, 2, 1)
if torch.cuda.is_available():
if spect.type() == 'torch.cuda.HalfTensor':
audio = torch.cuda.HalfTensor(spect.size(0),
self.n_remaining_channels,
spect.size(2)).normal_()
else:
audio = torch.cuda.FloatTensor(spect.size(0),
self.n_remaining_channels,
spect.size(2)).normal_()
else:
if spect.type() == 'torch.HalfTensor':
audio = torch.HalfTensor(spect.size(0),
self.n_remaining_channels,
@@ -274,7 +285,6 @@ class WaveGlow(torch.nn.Module):
audio_1 = audio[:,n_half:,:]
output = self.WN[k]((audio_0, spect))
s = output[:, n_half:, :]
b = output[:, :n_half, :]
audio_1 = (audio_1 - b)/torch.exp(s)
@@ -283,6 +293,12 @@ class WaveGlow(torch.nn.Module):
audio = self.convinv[k](audio, reverse=True)
if k % self.n_early_every == 0 and k > 0:
if torch.cuda.is_available():
if spect.type() == 'torch.cuda.HalfTensor':
z = torch.cuda.HalfTensor(spect.size(0), self.n_early_size, spect.size(2)).normal_()
else:
z = torch.cuda.FloatTensor(spect.size(0), self.n_early_size, spect.size(2)).normal_()
else:
if spect.type() == 'torch.HalfTensor':
z = torch.HalfTensor(spect.size(0), self.n_early_size, spect.size(2)).normal_()
else:
@@ -298,7 +314,7 @@ class WaveGlow(torch.nn.Module):
for WN in waveglow.WN:
WN.start = torch.nn.utils.remove_weight_norm(WN.start)
WN.in_layers = remove(WN.in_layers)
WN.cond_layer = torch.nn.utils.remove_weight_norm(WN.cond_layer)
WN.cond_layers = remove(WN.cond_layers)
WN.res_skip_layers = remove(WN.res_skip_layers)
return waveglow

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@@ -12,15 +12,22 @@ with open("HISTORY.rst") as history_file:
requirements = [
"klepto==0.1.6",
"numpy==1.16.4",
"numpy~=1.16.4",
"inflect==0.2.5",
"librosa==0.6.0",
"scipy==1.3.0",
"scipy~=1.3",
"Unidecode==1.0.22",
"torch==1.1.0",
"PyAudio==0.2.11"
"torch~=1.1.0",
]
extra_requirements = {
"playback": ["PyAudio==0.2.11"],
"server": [
"google-cloud-texttospeech==1.0.1",
"rpyc==4.1.4",
],
}
setup_requirements = ["pytest-runner"]
test_requirements = ["pytest"]
@@ -44,6 +51,7 @@ setup(
],
description="Taco2 TTS package.",
install_requires=requirements,
extras_require=extra_requirements,
long_description=readme + "\n\n" + history,
include_package_data=True,
keywords="tacotron2 tts",
@@ -53,7 +61,12 @@ setup(
test_suite="tests",
tests_require=test_requirements,
url="https://github.com/malarinv/tacotron2",
version="0.2.0",
version="0.3.0",
zip_safe=False,
entry_points={"console_scripts": ("tts_debug = taco2.tts:main",)},
entry_points={
"console_scripts": (
"tts_debug = taco2.tts:main",
"tts_rpyc_server = taco2.server.__main__:main",
)
},
)

View File

@@ -9,6 +9,11 @@ class Denoiser(torch.nn.Module):
def __init__(self, waveglow, filter_length=1024, n_overlap=4,
win_length=1024, mode='zeros', n_mel_channels=80,):
super(Denoiser, self).__init__()
if torch.cuda.is_available():
self.stft = STFT(filter_length=filter_length,
hop_length=int(filter_length/n_overlap),
win_length=win_length).cuda()
else:
self.stft = STFT(filter_length=filter_length,
hop_length=int(filter_length/n_overlap),
win_length=win_length).cpu()
@@ -32,6 +37,9 @@ class Denoiser(torch.nn.Module):
self.register_buffer('bias_spec', bias_spec[:, :, 0][:, :, None])
def forward(self, audio, strength=0.1):
if torch.cuda.is_available():
audio_spec, audio_angles = self.stft.transform(audio.cuda().float())
else:
audio_spec, audio_angles = self.stft.transform(audio.cpu().float())
audio_spec_denoised = audio_spec - self.bias_spec * strength
audio_spec_denoised = torch.clamp(audio_spec_denoised, 0.0)

View File

@@ -35,13 +35,13 @@ class HParams(object):
# Audio Parameters #
################################
max_wav_value = 32768.0
sampling_rate = 16000
sampling_rate = 22050
filter_length = 1024
hop_length = 256
win_length = 1024
n_mel_channels: int = 40
n_mel_channels: int = 80
mel_fmin: float = 0.0
mel_fmax: float = 4000.0
mel_fmax: float = 8000.0
################################
# Model Parameters #
################################

0
taco2/server/__init__.py Normal file
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48
taco2/server/__main__.py Normal file
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@@ -0,0 +1,48 @@
import os
import logging
import rpyc
from rpyc.utils.server import ThreadedServer
from .backend import TTSSynthesizer
tts_backend = os.environ.get("TTS_BACKEND", "taco2")
tts_synthesizer = TTSSynthesizer(backend=tts_backend)
class TTSService(rpyc.Service):
def on_connect(self, conn):
# code that runs when a connection is created
# (to init the service, if needed)
pass
def on_disconnect(self, conn):
# code that runs after the connection has already closed
# (to finalize the service, if needed)
pass
def exposed_synth_speech(self, utterance: str): # this is an exposed method
speech_audio = tts_synthesizer.synth_speech(utterance)
return speech_audio
def exposed_synth_speech_cb(
self, utterance: str, respond
): # this is an exposed method
speech_audio = tts_synthesizer.synth_speech(utterance)
respond(speech_audio)
def main():
logging.basicConfig(
level=logging.INFO,
format="%(asctime)s - %(name)s - %(levelname)s - %(message)s",
)
port = int(os.environ.get("TTS_RPYC_PORT", "7754"))
logging.info("starting tts server...")
t = ThreadedServer(TTSService, port=port)
t.start()
if __name__ == "__main__":
main()

45
taco2/server/backend.py Normal file
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@@ -0,0 +1,45 @@
import os
from google.cloud import texttospeech
from ..tts import TTSModel
tts_model_weights = os.environ.get(
"TTS_MODELS", "models/tacotron2_statedict.pt,models/waveglow_256channels.pt"
)
tts_creds = os.environ.get(
"GOOGLE_APPLICATION_CREDENTIALS", "/code/config/gre2e/keys/gre2e_gcp.json"
)
taco2, wav_glow = tts_model_weights.split(",", 1)
class TTSSynthesizer(object):
"""docstring for TTSSynthesizer."""
def __init__(self, backend="taco2"):
super(TTSSynthesizer, self).__init__()
if backend == "taco2":
tts_model = TTSModel(f"{taco2}", f"{wav_glow}") # Loads the models
self.synth_speech = tts_model.synth_speech
elif backend == "gcp":
client = texttospeech.TextToSpeechClient()
# Build the voice request, select the language code ("en-US") and the ssml
# voice gender ("neutral")
voice = texttospeech.types.VoiceSelectionParams(language_code="en-US")
# Select the type of audio file you want returned
audio_config = texttospeech.types.AudioConfig(
audio_encoding=texttospeech.enums.AudioEncoding.LINEAR16
)
# Perform the text-to-speech request on the text input with the selected
# voice parameters and audio file type
def gcp_synthesize(speech_text):
synthesis_input = texttospeech.types.SynthesisInput(text=speech_text)
response = client.synthesize_speech(
synthesis_input, voice, audio_config
)
return response.audio_content
self.synth_speech = gcp_synthesize

View File

@@ -40,6 +40,7 @@ from scipy.signal import get_window
from librosa.util import pad_center, tiny
from .audio_processing import window_sumsquare
DEVICE = torch.device('cuda:0' if torch.cuda.is_available() else 'cpu')
class STFT(torch.nn.Module):
"""
@@ -83,8 +84,8 @@ class STFT(torch.nn.Module):
forward_basis *= fft_window
inverse_basis *= fft_window
self.register_buffer("forward_basis", forward_basis.float())
self.register_buffer("inverse_basis", inverse_basis.float())
self.register_buffer("forward_basis", forward_basis.float().to(DEVICE))
self.register_buffer("inverse_basis", inverse_basis.float().to(DEVICE))
def transform(self, input_data):
num_batches = input_data.size(0)
@@ -120,10 +121,10 @@ class STFT(torch.nn.Module):
return magnitude, phase
def inverse(self, magnitude, phase):
phase = phase.to(DEVICE)
recombine_magnitude_phase = torch.cat(
[magnitude * torch.cos(phase), magnitude * torch.sin(phase)], dim=1
)
inverse_transform = F.conv_transpose1d(
recombine_magnitude_phase,
Variable(self.inverse_basis, requires_grad=False),
@@ -143,13 +144,10 @@ class STFT(torch.nn.Module):
# remove modulation effects
approx_nonzero_indices = torch.from_numpy(
np.where(window_sum > tiny(window_sum))[0]
)
).to(DEVICE)
window_sum = torch.autograd.Variable(
torch.from_numpy(window_sum), requires_grad=False
)
# window_sum = window_sum.cuda() if magnitude.is_cuda else
# window_sum
# initially not commented out
).to(DEVICE)
inverse_transform[:, :, approx_nonzero_indices] /= window_sum[
approx_nonzero_indices
]

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@@ -3,8 +3,10 @@
import numpy as np
import torch
import pyaudio
import klepto
import argparse
import warnings
from pathlib import Path
from .model import Tacotron2
from glow import WaveGlow
from .hparams import HParams
@@ -13,8 +15,9 @@ from .text import text_to_sequence
from .denoiser import Denoiser
from .audio_processing import griffin_lim, postprocess_audio
TTS_SAMPLE_RATE = 22050
OUTPUT_SAMPLE_RATE = 16000
OUTPUT_SAMPLE_RATE = 22050
GL_ITERS = 30
VOCODER_WAVEGLOW, VOCODER_GL = "wavglow", "gl"
# config from
# https://github.com/NVIDIA/waveglow/blob/master/config.json
@@ -34,22 +37,31 @@ class TTSModel(object):
def __init__(self, tacotron2_path, waveglow_path, **kwargs):
super(TTSModel, self).__init__()
hparams = HParams(**kwargs)
hparams.sampling_rate = TTS_SAMPLE_RATE
self.hparams = hparams
self.model = Tacotron2(hparams)
if torch.cuda.is_available():
self.model.load_state_dict(torch.load(tacotron2_path)["state_dict"])
self.model.cuda().eval()
else:
self.model.load_state_dict(
torch.load(tacotron2_path, map_location="cpu")["state_dict"]
)
self.model.eval()
self.k_cache = klepto.archives.file_archive(cached=False)
if waveglow_path:
if torch.cuda.is_available():
wave_params = torch.load(waveglow_path)
else:
wave_params = torch.load(waveglow_path, map_location="cpu")
try:
self.waveglow = WaveGlow(**WAVEGLOW_CONFIG)
self.waveglow.load_state_dict(wave_params)
self.waveglow.eval()
except:
self.waveglow = wave_params["model"]
self.waveglow = self.waveglow.remove_weightnorm(self.waveglow)
if torch.cuda.is_available():
self.waveglow.cuda().eval()
else:
self.waveglow.eval()
# workaround from
# https://github.com/NVIDIA/waveglow/issues/127
@@ -57,16 +69,16 @@ class TTSModel(object):
if "Conv" in str(type(m)):
setattr(m, "padding_mode", "zeros")
for k in self.waveglow.convinv:
k.float()
k.float().half()
self.denoiser = Denoiser(
self.waveglow, n_mel_channels=hparams.n_mel_channels
)
self.synth_speech = klepto.safe.inf_cache(cache=self.k_cache)(
self.synth_speech
self._synth_speech
)
else:
self.synth_speech = klepto.safe.inf_cache(cache=self.k_cache)(
self.synth_speech_gl
self._synth_speech_fast
)
self.taco_stft = TacotronSTFT(
hparams.filter_length,
@@ -77,8 +89,11 @@ class TTSModel(object):
mel_fmax=4000,
)
def generate_mel_postnet(self, text):
def _generate_mel_postnet(self, text):
sequence = np.array(text_to_sequence(text, ["english_cleaners"]))[None, :]
if torch.cuda.is_available():
sequence = torch.autograd.Variable(torch.from_numpy(sequence)).cuda().long()
else:
sequence = torch.autograd.Variable(torch.from_numpy(sequence)).long()
with torch.no_grad():
mel_outputs, mel_outputs_postnet, _, alignments = self.model.inference(
@@ -86,41 +101,66 @@ class TTSModel(object):
)
return mel_outputs_postnet
def synth_speech(self, text):
mel_outputs_postnet = self.generate_mel_postnet(text)
def synth_speech_array(self, text, vocoder):
mel_outputs_postnet = self._generate_mel_postnet(text)
if vocoder == VOCODER_WAVEGLOW:
with torch.no_grad():
audio_t = self.waveglow.infer(mel_outputs_postnet, sigma=0.666)
audio_t = self.denoiser(audio_t, 0.1)[0]
audio = audio_t[0].data.cpu().numpy()
return postprocess_audio(
audio, src_rate=TTS_SAMPLE_RATE, dst_rate=OUTPUT_SAMPLE_RATE
)
def synth_speech_gl(self, text, griffin_iters=60):
mel_outputs_postnet = self.generate_mel_postnet(text)
audio = audio_t[0].data
elif vocoder == VOCODER_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
spec_from_mel = (
spec_from_mel.cuda() if torch.cuda.is_available() else spec_from_mel
)
audio = griffin_lim(
torch.autograd.Variable(spec_from_mel[:, :, :-1]),
self.taco_stft.stft_fn,
griffin_iters,
GL_ITERS,
)
audio = audio.squeeze()
else:
raise ValueError("vocoder arg should be one of [wavglow|gl]")
audio = audio.cpu().numpy()
return audio
def _synth_speech(
self, text, speed: float = 1.0, sample_rate: int = OUTPUT_SAMPLE_RATE
):
audio = self.synth_speech_array(text, VOCODER_WAVEGLOW)
return postprocess_audio(
audio, src_rate=TTS_SAMPLE_RATE, dst_rate=OUTPUT_SAMPLE_RATE
audio,
src_rate=self.hparams.sampling_rate,
dst_rate=sample_rate,
tempo=speed,
)
def _synth_speech_fast(
self, text, speed: float = 1.0, sample_rate: int = OUTPUT_SAMPLE_RATE
):
audio = self.synth_speech_array(text, VOCODER_GL)
return postprocess_audio(
audio,
tempo=speed,
src_rate=self.hparams.sampling_rate,
dst_rate=sample_rate,
)
def player_gen():
try:
import pyaudio
except ModuleNotFoundError:
warnings.warn("module 'pyaudio' is not installed requried for playback")
return
audio_interface = pyaudio.PyAudio()
_audio_stream = audio_interface.open(
format=pyaudio.paInt16, channels=1, rate=OUTPUT_SAMPLE_RATE, output=True
@@ -133,10 +173,7 @@ def player_gen():
return play_device
def repl():
tts_model = TTSModel(
"/Users/malar/Work/tacotron2_r4_83000.pt", "/Users/malar/Work/waveglow_484000"
)
def repl(tts_model):
player = player_gen()
def loop():
@@ -148,7 +185,26 @@ def repl():
def main():
interactive_loop = repl()
parser = argparse.ArgumentParser(
formatter_class=argparse.ArgumentDefaultsHelpFormatter
)
parser.add_argument(
"-t",
"--tacotron2_path",
type=Path,
default="./tacotron.pt",
help="Path to a tacotron2 model",
)
parser.add_argument(
"-w",
"--waveglow_path",
type=Path,
default="./waveglow_256channels.pt",
help="Path to a waveglow model",
)
args = parser.parse_args()
tts_model = TTSModel(**vars(args))
interactive_loop = repl(tts_model)
while True:
interactive_loop()

View File

@@ -27,6 +27,6 @@ def load_filepaths_and_text(filename, split="|"):
def to_gpu(x):
x = x.contiguous()
# if torch.cuda.is_available(): #initially not commented out
# x = x.cuda(non_blocking=True) # initially not commented out
if torch.cuda.is_available(): #initially not commented out
x = x.cuda(non_blocking=True) # initially not commented out
return torch.autograd.Variable(x)