提交 b2464597 编写于 作者: K Kexin Zhao

fix conflict

......@@ -9,7 +9,7 @@ Parakeet aims to provide a flexible, efficient and state-of-the-art text-to-spee
In particular, it features the latest [WaveFlow] (https://arxiv.org/abs/1912.01219) model proposed by Baidu Research.
- WaveFlow can synthesize 22.05 kHz high-fidelity speech around 40x faster than real-time on a Nvidia V100 GPU without engineered inference kernels, which is faster than [WaveGlow] (https://github.com/NVIDIA/waveglow) and serveral orders of magnitude faster than WaveNet.
- WaveFlow is a small-footprint flow-based model for raw audio. It has only 5.9M parameters, which is 15x smalller than WaveGlow (87.9M) and comparable to WaveNet (4.6M).
- WaveFlow is directly trained with maximum likelihood without probability density distillation and auxiliary losses as used in Parallel WaveNet and ClariNet, which simplifies the training pipeline and reduces the cost of development.
- WaveFlow is directly trained with maximum likelihood without probability density distillation and auxiliary losses as used in Parallel WaveNet and ClariNet, which simplifies the training pipeline and reduces the cost of development.
### Setup
......@@ -45,8 +45,10 @@ nltk.download("cmudict")
- [Deep Voice 3: Scaling Text-to-Speech with Convolutional Sequence Learning](https://arxiv.org/abs/1710.07654)
- [Neural Speech Synthesis with Transformer Network](https://arxiv.org/abs/1809.08895)
- [FastSpeech: Fast, Robust and Controllable Text to Speech](https://arxiv.org/abs/1905.09263).
- [FastSpeech: Fast, Robust and Controllable Text to Speech](https://arxiv.org/abs/1905.09263)
- [WaveFlow: A Compact Flow-based Model for Raw Audio](https://arxiv.org/abs/1912.01219)
- [WaveNet: A Generative Model for Raw Audio](https://arxiv.org/abs/1609.03499)
- [ClariNet: Parallel Wave Generation in End-to-End Text-to-Speech](https://arxiv.org/abs/1807.07281)
## Examples
......@@ -54,6 +56,8 @@ nltk.download("cmudict")
- [Train a TransformerTTS model with ljspeech dataset](./examples/transformer_tts)
- [Train a FastSpeech model with ljspeech dataset](./examples/fastspeech)
- [Train a WaveFlow model with ljspeech dataset](./examples/waveflow)
- [Train a WaveNet model with ljspeech dataset](./examples/wavenet)
- [Train a Clarinet model with ljspeech dataset](./examples/clarinet)
## Copyright and License
......
data:
batch_size: 4
batch_size: 8
train_clip_seconds: 0.5
sample_rate: 22050
hop_length: 256
......
# Copyright (c) 2020 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import os
import sys
import argparse
import ruamel.yaml
import random
from tqdm import tqdm
import pickle
import numpy as np
from tensorboardX import SummaryWriter
import paddle.fluid.dygraph as dg
from paddle import fluid
from parakeet.models.wavenet import WaveNet, UpsampleNet
from parakeet.models.clarinet import STFT, Clarinet, ParallelWaveNet
from parakeet.data import TransformDataset, SliceDataset, RandomSampler, SequentialSampler, DataCargo
from parakeet.utils.layer_tools import summary, freeze
from utils import valid_model, eval_model, save_checkpoint, load_checkpoint, load_model
sys.path.append("../wavenet")
from data import LJSpeechMetaData, Transform, DataCollector
if __name__ == "__main__":
parser = argparse.ArgumentParser(
description="synthesize audio files from mel spectrogram in the validation set."
)
parser.add_argument("--config", type=str, help="path of the config file.")
parser.add_argument(
"--device", type=int, default=-1, help="device to use.")
parser.add_argument("--data", type=str, help="path of LJspeech dataset.")
parser.add_argument(
"checkpoint", type=str, help="checkpoint to load from.")
parser.add_argument(
"output", type=str, default="experiment", help="path to save student.")
args = parser.parse_args()
with open(args.config, 'rt') as f:
config = ruamel.yaml.safe_load(f)
ljspeech_meta = LJSpeechMetaData(args.data)
data_config = config["data"]
sample_rate = data_config["sample_rate"]
n_fft = data_config["n_fft"]
win_length = data_config["win_length"]
hop_length = data_config["hop_length"]
n_mels = data_config["n_mels"]
train_clip_seconds = data_config["train_clip_seconds"]
transform = Transform(sample_rate, n_fft, win_length, hop_length, n_mels)
ljspeech = TransformDataset(ljspeech_meta, transform)
valid_size = data_config["valid_size"]
ljspeech_valid = SliceDataset(ljspeech, 0, valid_size)
ljspeech_train = SliceDataset(ljspeech, valid_size, len(ljspeech))
teacher_config = config["teacher"]
n_loop = teacher_config["n_loop"]
n_layer = teacher_config["n_layer"]
filter_size = teacher_config["filter_size"]
context_size = 1 + n_layer * sum([filter_size**i for i in range(n_loop)])
print("context size is {} samples".format(context_size))
train_batch_fn = DataCollector(context_size, sample_rate, hop_length,
train_clip_seconds)
valid_batch_fn = DataCollector(
context_size, sample_rate, hop_length, train_clip_seconds, valid=True)
batch_size = data_config["batch_size"]
train_cargo = DataCargo(
ljspeech_train,
train_batch_fn,
batch_size,
sampler=RandomSampler(ljspeech_train))
# only batch=1 for validation is enabled
valid_cargo = DataCargo(
ljspeech_valid,
valid_batch_fn,
batch_size=1,
sampler=SequentialSampler(ljspeech_valid))
if args.device == -1:
place = fluid.CPUPlace()
else:
place = fluid.CUDAPlace(args.device)
with dg.guard(place):
# conditioner(upsampling net)
conditioner_config = config["conditioner"]
upsampling_factors = conditioner_config["upsampling_factors"]
upsample_net = UpsampleNet(upscale_factors=upsampling_factors)
freeze(upsample_net)
residual_channels = teacher_config["residual_channels"]
loss_type = teacher_config["loss_type"]
output_dim = teacher_config["output_dim"]
log_scale_min = teacher_config["log_scale_min"]
assert loss_type == "mog" and output_dim == 3, \
"the teacher wavenet should be a wavenet with single gaussian output"
teacher = WaveNet(n_loop, n_layer, residual_channels, output_dim,
n_mels, filter_size, loss_type, log_scale_min)
# load & freeze upsample_net & teacher
freeze(teacher)
student_config = config["student"]
n_loops = student_config["n_loops"]
n_layers = student_config["n_layers"]
student_residual_channels = student_config["residual_channels"]
student_filter_size = student_config["filter_size"]
student_log_scale_min = student_config["log_scale_min"]
student = ParallelWaveNet(n_loops, n_layers, student_residual_channels,
n_mels, student_filter_size)
stft_config = config["stft"]
stft = STFT(
n_fft=stft_config["n_fft"],
hop_length=stft_config["hop_length"],
win_length=stft_config["win_length"])
lmd = config["loss"]["lmd"]
model = Clarinet(upsample_net, teacher, student, stft,
student_log_scale_min, lmd)
summary(model)
load_model(model, args.checkpoint)
# loader
train_loader = fluid.io.DataLoader.from_generator(
capacity=10, return_list=True)
train_loader.set_batch_generator(train_cargo, place)
valid_loader = fluid.io.DataLoader.from_generator(
capacity=10, return_list=True)
valid_loader.set_batch_generator(valid_cargo, place)
if not os.path.exists(args.output):
os.makedirs(args.output)
eval_model(model, valid_loader, args.output, sample_rate)
# Copyright (c) 2020 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import os
import sys
import argparse
import ruamel.yaml
import random
from tqdm import tqdm
import pickle
import numpy as np
from tensorboardX import SummaryWriter
import paddle.fluid.dygraph as dg
from paddle import fluid
from parakeet.models.wavenet import WaveNet, UpsampleNet
from parakeet.models.clarinet import STFT, Clarinet, ParallelWaveNet
from parakeet.data import TransformDataset, SliceDataset, RandomSampler, SequentialSampler, DataCargo
from parakeet.utils.layer_tools import summary, freeze
from utils import make_output_tree, valid_model, save_checkpoint, load_checkpoint, load_wavenet
sys.path.append("../wavenet")
from data import LJSpeechMetaData, Transform, DataCollector
if __name__ == "__main__":
parser = argparse.ArgumentParser(
description="train a clarinet model with LJspeech and a trained wavenet model."
)
parser.add_argument("--config", type=str, help="path of the config file.")
parser.add_argument(
"--device", type=int, default=-1, help="device to use.")
parser.add_argument(
"--output",
type=str,
default="experiment",
help="path to save student.")
parser.add_argument("--data", type=str, help="path of LJspeech dataset.")
parser.add_argument("--resume", type=str, help="checkpoint to load from.")
parser.add_argument(
"--wavenet", type=str, help="wavenet checkpoint to use.")
args = parser.parse_args()
with open(args.config, 'rt') as f:
config = ruamel.yaml.safe_load(f)
ljspeech_meta = LJSpeechMetaData(args.data)
data_config = config["data"]
sample_rate = data_config["sample_rate"]
n_fft = data_config["n_fft"]
win_length = data_config["win_length"]
hop_length = data_config["hop_length"]
n_mels = data_config["n_mels"]
train_clip_seconds = data_config["train_clip_seconds"]
transform = Transform(sample_rate, n_fft, win_length, hop_length, n_mels)
ljspeech = TransformDataset(ljspeech_meta, transform)
valid_size = data_config["valid_size"]
ljspeech_valid = SliceDataset(ljspeech, 0, valid_size)
ljspeech_train = SliceDataset(ljspeech, valid_size, len(ljspeech))
teacher_config = config["teacher"]
n_loop = teacher_config["n_loop"]
n_layer = teacher_config["n_layer"]
filter_size = teacher_config["filter_size"]
context_size = 1 + n_layer * sum([filter_size**i for i in range(n_loop)])
print("context size is {} samples".format(context_size))
train_batch_fn = DataCollector(context_size, sample_rate, hop_length,
train_clip_seconds)
valid_batch_fn = DataCollector(
context_size, sample_rate, hop_length, train_clip_seconds, valid=True)
batch_size = data_config["batch_size"]
train_cargo = DataCargo(
ljspeech_train,
train_batch_fn,
batch_size,
sampler=RandomSampler(ljspeech_train))
# only batch=1 for validation is enabled
valid_cargo = DataCargo(
ljspeech_valid,
valid_batch_fn,
batch_size=1,
sampler=SequentialSampler(ljspeech_valid))
make_output_tree(args.output)
if args.device == -1:
place = fluid.CPUPlace()
else:
place = fluid.CUDAPlace(args.device)
with dg.guard(place):
# conditioner(upsampling net)
conditioner_config = config["conditioner"]
upsampling_factors = conditioner_config["upsampling_factors"]
upsample_net = UpsampleNet(upscale_factors=upsampling_factors)
freeze(upsample_net)
residual_channels = teacher_config["residual_channels"]
loss_type = teacher_config["loss_type"]
output_dim = teacher_config["output_dim"]
log_scale_min = teacher_config["log_scale_min"]
assert loss_type == "mog" and output_dim == 3, \
"the teacher wavenet should be a wavenet with single gaussian output"
teacher = WaveNet(n_loop, n_layer, residual_channels, output_dim,
n_mels, filter_size, loss_type, log_scale_min)
freeze(teacher)
student_config = config["student"]
n_loops = student_config["n_loops"]
n_layers = student_config["n_layers"]
student_residual_channels = student_config["residual_channels"]
student_filter_size = student_config["filter_size"]
student_log_scale_min = student_config["log_scale_min"]
student = ParallelWaveNet(n_loops, n_layers, student_residual_channels,
n_mels, student_filter_size)
stft_config = config["stft"]
stft = STFT(
n_fft=stft_config["n_fft"],
hop_length=stft_config["hop_length"],
win_length=stft_config["win_length"])
lmd = config["loss"]["lmd"]
model = Clarinet(upsample_net, teacher, student, stft,
student_log_scale_min, lmd)
summary(model)
# optim
train_config = config["train"]
learning_rate = train_config["learning_rate"]
anneal_rate = train_config["anneal_rate"]
anneal_interval = train_config["anneal_interval"]
lr_scheduler = dg.ExponentialDecay(
learning_rate, anneal_interval, anneal_rate, staircase=True)
optim = fluid.optimizer.Adam(
lr_scheduler, parameter_list=model.parameters())
gradiant_max_norm = train_config["gradient_max_norm"]
clipper = fluid.dygraph_grad_clip.GradClipByGlobalNorm(
gradiant_max_norm)
assert args.wavenet or args.resume, "you should load from a trained wavenet or resume training; training without a trained wavenet is not recommended."
if args.wavenet:
load_wavenet(model, args.wavenet)
if args.resume:
load_checkpoint(model, optim, args.resume)
# loader
train_loader = fluid.io.DataLoader.from_generator(
capacity=10, return_list=True)
train_loader.set_batch_generator(train_cargo, place)
valid_loader = fluid.io.DataLoader.from_generator(
capacity=10, return_list=True)
valid_loader.set_batch_generator(valid_cargo, place)
# train
max_iterations = train_config["max_iterations"]
checkpoint_interval = train_config["checkpoint_interval"]
eval_interval = train_config["eval_interval"]
checkpoint_dir = os.path.join(args.output, "checkpoints")
state_dir = os.path.join(args.output, "states")
log_dir = os.path.join(args.output, "log")
writer = SummaryWriter(log_dir)
# training loop
global_step = 1
global_epoch = 1
while global_step < max_iterations:
epoch_loss = 0.
for j, batch in tqdm(enumerate(train_loader), desc="[train]"):
audios, mels, audio_starts = batch
model.train()
loss_dict = model(
audios, mels, audio_starts, clip_kl=global_step > 500)
writer.add_scalar("learning_rate",
optim._learning_rate.step().numpy()[0],
global_step)
for k, v in loss_dict.items():
writer.add_scalar("loss/{}".format(k),
v.numpy()[0], global_step)
l = loss_dict["loss"]
step_loss = l.numpy()[0]
print("[train] loss: {:<8.6f}".format(step_loss))
epoch_loss += step_loss
l.backward()
optim.minimize(l, grad_clip=clipper)
optim.clear_gradients()
if global_step % eval_interval == 0:
# evaluate on valid dataset
valid_model(model, valid_loader, state_dir, global_step,
sample_rate)
if global_step % checkpoint_interval == 0:
save_checkpoint(model, optim, checkpoint_dir, global_step)
global_step += 1
# epoch loss
average_loss = epoch_loss / j
writer.add_scalar("average_loss", average_loss, global_epoch)
global_epoch += 1
# Copyright (c) 2020 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import os
import soundfile as sf
from tensorboardX import SummaryWriter
from collections import OrderedDict
from paddle import fluid
import paddle.fluid.dygraph as dg
def make_output_tree(output_dir):
checkpoint_dir = os.path.join(output_dir, "checkpoints")
if not os.path.exists(checkpoint_dir):
os.makedirs(checkpoint_dir)
state_dir = os.path.join(output_dir, "states")
if not os.path.exists(state_dir):
os.makedirs(state_dir)
def valid_model(model, valid_loader, output_dir, global_step, sample_rate):
model.eval()
for i, batch in enumerate(valid_loader):
# print("sentence {}".format(i))
path = os.path.join(output_dir,
"step_{}_sentence_{}.wav".format(global_step, i))
audio_clips, mel_specs, audio_starts = batch
wav_var = model.synthesis(mel_specs)
wav_np = wav_var.numpy()[0]
sf.write(path, wav_np, samplerate=sample_rate)
print("generated {}".format(path))
def eval_model(model, valid_loader, output_dir, sample_rate):
model.eval()
for i, batch in enumerate(valid_loader):
# print("sentence {}".format(i))
path = os.path.join(output_dir, "sentence_{}.wav".format(i))
audio_clips, mel_specs, audio_starts = batch
wav_var = model.synthesis(mel_specs)
wav_np = wav_var.numpy()[0]
sf.write(path, wav_np, samplerate=sample_rate)
print("generated {}".format(path))
def save_checkpoint(model, optim, checkpoint_dir, global_step):
path = os.path.join(checkpoint_dir, "step_{}".format(global_step))
dg.save_dygraph(model.state_dict(), path)
print("saving model to {}".format(path + ".pdparams"))
if optim:
dg.save_dygraph(optim.state_dict(), path)
print("saving optimizer to {}".format(path + ".pdopt"))
def load_model(model, path):
model_dict, _ = dg.load_dygraph(path)
model.state_dict(model_dict)
print("loaded model from {}.pdparams".format(path))
def load_checkpoint(model, optim, path):
model_dict, optim_dict = dg.load_dygraph(path)
model.state_dict(model_dict)
print("loaded model from {}.pdparams".format(path))
if optim_dict:
optim.set_dict(optim_dict)
print("loaded optimizer from {}.pdparams".format(path))
def load_wavenet(model, path):
wavenet_dict, _ = dg.load_dygraph(path)
encoder_dict = OrderedDict()
teacher_dict = OrderedDict()
for k, v in wavenet_dict.items():
if k.startswith("encoder."):
encoder_dict[k.split('.', 1)[1]] = v
else:
# k starts with "decoder."
teacher_dict[k.split('.', 1)[1]] = v
model.encoder.set_dict(encoder_dict)
model.teacher.set_dict(teacher_dict)
print("loaded the encoder part and teacher part from wavenet model.")
......@@ -23,7 +23,7 @@ The model consists of an encoder, a decoder and a converter (and a speaker embed
```text
├── data.py data_processing
├── ljspeech.yaml (example) configuration file
├── configs/ (example) configuration files
├── sentences.txt sample sentences
├── synthesis.py script to synthesize waveform from text
├── train.py script to train a model
......@@ -72,7 +72,7 @@ optional arguments:
Example script:
```bash
python train.py --config=./ljspeech.yaml --data=./LJSpeech-1.1/ --output=experiment --device=0
python train.py --config=configs/ljspeech.yaml --data=./LJSpeech-1.1/ --output=experiment --device=0
```
You can monitor training log via tensorboard, using the script below.
......@@ -110,5 +110,5 @@ optional arguments:
Example script:
```bash
python synthesis.py --config=./ljspeech.yaml --device=0 experiment/checkpoints/model_step_005000000 sentences.txt generated
python synthesis.py --config=configs/ljspeech.yaml --device=0 experiment/checkpoints/model_step_005000000 sentences.txt generated
```
data:
batch_size: 4
batch_size: 16
train_clip_seconds: 0.5
sample_rate: 22050
hop_length: 256
......@@ -30,7 +30,7 @@ train:
snap_interval: 10000
eval_interval: 10000
max_iterations: 200000
max_iterations: 2000000
data:
batch_size: 4
batch_size: 16
train_clip_seconds: 0.5
sample_rate: 22050
hop_length: 256
......@@ -30,7 +30,7 @@ train:
snap_interval: 10000
eval_interval: 10000
max_iterations: 200000
max_iterations: 2000000
data:
batch_size: 4
batch_size: 16
train_clip_seconds: 0.5
sample_rate: 22050
hop_length: 256
......@@ -30,7 +30,7 @@ train:
snap_interval: 10000
eval_interval: 10000
max_iterations: 200000
max_iterations: 2000000
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