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PaddleSpeech/tests/deepspeech2_model_test.py

106 lines
3.4 KiB

E2E/Streaming Transformer/Conformer ASR (#578) * add cmvn and label smoothing loss layer * add layer for transformer * add glu and conformer conv * add torch compatiable hack, mask funcs * not hack size since it exists * add test; attention * add attention, common utils, hack paddle * add audio utils * conformer batch padding mask bug fix #223 * fix typo, python infer fix rnn mem opt name error and batchnorm1d, will be available at 2.0.2 * fix ci * fix ci * add encoder * refactor egs * add decoder * refactor ctc, add ctc align, refactor ckpt, add warmup lr scheduler, cmvn utils * refactor docs * add fix * fix readme * fix bugs, refactor collator, add pad_sequence, fix ckpt bugs * fix docstring * refactor data feed order * add u2 model * refactor cmvn, test * add utils * add u2 config * fix bugs * fix bugs * fix autograd maybe has problem when using inplace operation * refactor data, build vocab; add format data * fix text featurizer * refactor build vocab * add fbank, refactor feature of speech * refactor audio feat * refactor data preprare * refactor data * model init from config * add u2 bins * flake8 * can train * fix bugs, add coverage, add scripts * test can run * fix data * speed perturb with sox * add spec aug * fix for train * fix train logitc * fix logger * log valid loss, time dataset process * using np for speed perturb, remove some debug log of grad clip * fix logger * fix build vocab * fix logger name * using module logger as default * fix * fix install * reorder imports * fix board logger * fix logger * kaldi fbank and mfcc * fix cmvn and print prarams * fix add_eos_sos and cmvn * fix cmvn compute * fix logger and cmvn * fix subsampling, label smoothing loss, remove useless * add notebook test * fix log * fix tb logger * multi gpu valid * fix log * fix log * fix config * fix compute cmvn, need paddle 2.1 * add cmvn notebook * fix layer tools * fix compute cmvn * add rtf * fix decoding * fix layer tools * fix log, add avg script * more avg and test info * fix dataset pickle problem; using 2.1 paddle; num_workers can > 0; ckpt save in exp dir;fix setup.sh; * add vimrc * refactor tiny script, add transformer and stream conf * spm demo; librisppech scripts and confs * fix log * add librispeech scripts * refactor data pipe; fix conf; fix u2 default params * fix bugs * refactor aishell scripts * fix test * fix cmvn * fix s0 scripts * fix ds2 scripts and bugs * fix dev & test dataset filter * fix dataset filter * filter dev * fix ckpt path * filter test, since librispeech will cause OOM, but all test wer will be worse, since mismatch train with test * add comment * add syllable doc * fix ds2 configs * add doc * add pypinyin tools * fix decoder using blank_id=0 * mmseg with pybind11 * format code
4 years ago
# Copyright (c) 2021 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 unittest
import numpy as np
import paddle
from deepspeech.models.deepspeech2 import DeepSpeech2Model
class TestDeepSpeech2Model(unittest.TestCase):
def setUp(self):
paddle.set_device('cpu')
self.batch_size = 2
self.feat_dim = 161
max_len = 64
# (B, T, D)
audio = np.random.randn(self.batch_size, max_len, self.feat_dim)
audio_len = np.random.randint(max_len, size=self.batch_size)
audio_len[-1] = max_len
# (B, U)
text = np.array([[1, 2], [1, 2]])
text_len = np.array([2] * self.batch_size)
self.audio = paddle.to_tensor(audio, dtype='float32')
self.audio_len = paddle.to_tensor(audio_len, dtype='int64')
self.text = paddle.to_tensor(text, dtype='int32')
self.text_len = paddle.to_tensor(text_len, dtype='int64')
def test_ds2_1(self):
model = DeepSpeech2Model(
feat_size=self.feat_dim,
dict_size=10,
num_conv_layers=2,
num_rnn_layers=3,
rnn_size=1024,
use_gru=False,
share_rnn_weights=False, )
loss = model(self.audio, self.audio_len, self.text, self.text_len)
self.assertEqual(loss.numel(), 1)
def test_ds2_2(self):
model = DeepSpeech2Model(
feat_size=self.feat_dim,
dict_size=10,
num_conv_layers=2,
num_rnn_layers=3,
rnn_size=1024,
use_gru=True,
share_rnn_weights=False, )
loss = model(self.audio, self.audio_len, self.text, self.text_len)
self.assertEqual(loss.numel(), 1)
def test_ds2_3(self):
model = DeepSpeech2Model(
feat_size=self.feat_dim,
dict_size=10,
num_conv_layers=2,
num_rnn_layers=3,
rnn_size=1024,
use_gru=False,
share_rnn_weights=True, )
loss = model(self.audio, self.audio_len, self.text, self.text_len)
self.assertEqual(loss.numel(), 1)
def test_ds2_4(self):
model = DeepSpeech2Model(
feat_size=self.feat_dim,
dict_size=10,
num_conv_layers=2,
num_rnn_layers=3,
rnn_size=1024,
use_gru=True,
share_rnn_weights=True, )
loss = model(self.audio, self.audio_len, self.text, self.text_len)
self.assertEqual(loss.numel(), 1)
def test_ds2_5(self):
model = DeepSpeech2Model(
feat_size=self.feat_dim,
dict_size=10,
num_conv_layers=2,
num_rnn_layers=3,
rnn_size=1024,
use_gru=False,
share_rnn_weights=False, )
loss = model(self.audio, self.audio_len, self.text, self.text_len)
self.assertEqual(loss.numel(), 1)
if __name__ == '__main__':
unittest.main()