msb-public
/
PaddleSpeech
mirror of https://github.com/PaddlePaddle/PaddleSpeech
1
0
Code Issues Projects Releases Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from '7fc9f5d459'
${ noResults }
PaddleSpeech/paddlespeech/s2t/exps/deepspeech2/model.py

638 lines
26 KiB
Raw Blame History

# 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.
"""Contains DeepSpeech2 and DeepSpeech2Online model."""
import os
import time
from collections import defaultdict
from contextlib import nullcontext
import jsonlines
import numpy as np
import paddle
from paddle import distributed as dist
from paddle import inference
from paddle.io import DataLoader
from paddlespeech.s2t.io.dataloader import BatchDataLoader
from paddlespeech.s2t.frontend.featurizer.text_featurizer import TextFeaturizer
from paddlespeech.s2t.io.dataset import ManifestDataset
from paddlespeech.s2t.models.ds2 import DeepSpeech2InferModel
from paddlespeech.s2t.models.ds2 import DeepSpeech2Model
from paddlespeech.s2t.training.gradclip import ClipGradByGlobalNormWithLog
from paddlespeech.s2t.training.reporter import report
from paddlespeech.s2t.training.timer import Timer
from paddlespeech.s2t.training.trainer import Trainer
from paddlespeech.s2t.utils import error_rate
from paddlespeech.s2t.utils import layer_tools
from paddlespeech.s2t.utils import mp_tools
from paddlespeech.s2t.utils.log import Log
from paddlespeech.s2t.utils.utility import UpdateConfig
logger = Log(__name__).getlog()
class DeepSpeech2Trainer(Trainer):
def __init__(self, config, args):
super().__init__(config, args)
def train_batch(self, batch_index, batch_data, msg):
batch_size = self.config.batch_size
accum_grad = self.config.accum_grad
start = time.time()
# forward
utt, audio, audio_len, text, text_len = batch_data
loss = self.model(audio, audio_len, text, text_len)
losses_np = {
'train_loss': float(loss),
}
# loss backward
if (batch_index + 1) % accum_grad != 0:
# Disable gradient synchronizations across DDP processes.
# Within this context, gradients will be accumulated on module
# variables, which will later be synchronized.
context = self.model.no_sync if (hasattr(self.model, "no_sync") and
self.parallel) else nullcontext
else:
# Used for single gpu training and DDP gradient synchronization
# processes.
context = nullcontext
with context():
loss.backward()
layer_tools.print_grads(self.model, print_func=None)
# optimizer step
if (batch_index + 1) % accum_grad == 0:
self.optimizer.step()
self.optimizer.clear_grad()
self.iteration += 1
iteration_time = time.time() - start
for k, v in losses_np.items():
report(k, v)
report("batch_size", batch_size)
report("accum", accum_grad)
report("step_cost", iteration_time)
if dist.get_rank() == 0 and self.visualizer:
for k, v in losses_np.items():
# `step -1` since we update `step` after optimizer.step().
self.visualizer.add_scalar("train/{}".format(k), v,
self.iteration - 1)
@paddle.no_grad()
def valid(self):
logger.info(f"Valid Total Examples: {len(self.valid_loader.dataset)}")
self.model.eval()
valid_losses = defaultdict(list)
num_seen_utts = 1
total_loss = 0.0
for i, batch in enumerate(self.valid_loader):
utt, audio, audio_len, text, text_len = batch
loss = self.model(audio, audio_len, text, text_len)
if paddle.isfinite(loss):
num_utts = batch[1].shape[0]
num_seen_utts += num_utts
total_loss += float(loss) * num_utts
valid_losses['val_loss'].append(float(loss))
if (i + 1) % self.config.log_interval == 0:
valid_dump = {k: np.mean(v) for k, v in valid_losses.items()}
valid_dump['val_history_loss'] = total_loss / num_seen_utts
# logging
msg = f"Valid: Rank: {dist.get_rank()}, "
msg += "epoch: {}, ".format(self.epoch)
msg += "step: {}, ".format(self.iteration)
msg += "batch : {}/{}, ".format(i + 1, len(self.valid_loader))
msg += ', '.join('{}: {:>.6f}'.format(k, v)
for k, v in valid_dump.items())
logger.info(msg)
logger.info('Rank {} Val info val_loss {}'.format(
dist.get_rank(), total_loss / num_seen_utts))
return total_loss, num_seen_utts
def setup_model(self):
config = self.config.clone()
with UpdateConfig(config):
if self.train:
config.input_dim = self.train_loader.feat_dim
config.output_dim = self.train_loader.vocab_size
else:
config.input_dim = self.test_loader.feat_dim
config.output_dim = self.test_loader.vocab_size
model = DeepSpeech2Model.from_config(config)
if self.parallel:
model = paddle.DataParallel(model)
logger.info(f"{model}")
layer_tools.print_params(model, logger.info)
self.model = model
logger.info("Setup model!")
if not self.train:
return
grad_clip = ClipGradByGlobalNormWithLog(config.global_grad_clip)
lr_scheduler = paddle.optimizer.lr.ExponentialDecay(
learning_rate=config.lr, gamma=config.lr_decay, verbose=True)
optimizer = paddle.optimizer.Adam(
learning_rate=lr_scheduler,
parameters=model.parameters(),
weight_decay=paddle.regularizer.L2Decay(config.weight_decay),
grad_clip=grad_clip)
self.optimizer = optimizer
self.lr_scheduler = lr_scheduler
logger.info("Setup optimizer/lr_scheduler!")
def setup_dataloader(self):
config = self.config.clone()
config.defrost()
if self.train:
# train/valid dataset, return token ids
self.train_loader = BatchDataLoader(
json_file=config.train_manifest,
train_mode=True,
sortagrad=config.sortagrad,
batch_size=config.batch_size,
maxlen_in=config.maxlen_in,
maxlen_out=config.maxlen_out,
minibatches=config.minibatches,
mini_batch_size=self.args.ngpu,
batch_count=config.batch_count,
batch_bins=config.batch_bins,
batch_frames_in=config.batch_frames_in,
batch_frames_out=config.batch_frames_out,
batch_frames_inout=config.batch_frames_inout,
preprocess_conf=config.preprocess_config,
n_iter_processes=config.num_workers,
subsampling_factor=1,
num_encs=1,
dist_sampler=config.get('dist_sampler', False),
shortest_first=False)
self.valid_loader = BatchDataLoader(
json_file=config.dev_manifest,
train_mode=False,
sortagrad=False,
batch_size=config.batch_size,
maxlen_in=float('inf'),
maxlen_out=float('inf'),
minibatches=0,
mini_batch_size=self.args.ngpu,
batch_count='auto',
batch_bins=0,
batch_frames_in=0,
batch_frames_out=0,
batch_frames_inout=0,
preprocess_conf=config.preprocess_config,
n_iter_processes=config.num_workers,
subsampling_factor=1,
num_encs=1,
dist_sampler=config.get('dist_sampler', False),
shortest_first=False)
logger.info("Setup train/valid Dataloader!")
else:
decode_batch_size = config.get('decode', dict()).get(
'decode_batch_size', 1)
# test dataset, return raw text
self.test_loader = BatchDataLoader(
json_file=config.test_manifest,
train_mode=False,
sortagrad=False,
batch_size=decode_batch_size,
maxlen_in=float('inf'),
maxlen_out=float('inf'),
minibatches=0,
mini_batch_size=1,
batch_count='auto',
batch_bins=0,
batch_frames_in=0,
batch_frames_out=0,
batch_frames_inout=0,
preprocess_conf=config.preprocess_config,
n_iter_processes=1,
subsampling_factor=1,
num_encs=1)
logger.info("Setup test/align Dataloader!")
class DeepSpeech2Tester(DeepSpeech2Trainer):
def __init__(self, config, args):
super().__init__(config, args)
self._text_featurizer = TextFeaturizer(
unit_type=config.unit_type,
vocab=config.vocab_filepath)
self.vocab_list = self._text_featurizer.vocab_list
def ordid2token(self, texts, texts_len):
""" ord() id to chr() chr """
trans = []
for text, n in zip(texts, texts_len):
n = n.numpy().item()
ids = text[:n]
trans.append(self._text_featurizer.defeaturize(ids.numpy().tolist()))
return trans
def compute_metrics(self,
utts,
audio,
audio_len,
texts,
texts_len,
fout=None):
decode_cfg = self.config.decode
errors_sum, len_refs, num_ins = 0.0, 0, 0
errors_func = error_rate.char_errors if decode_cfg.error_rate_type == 'cer' else error_rate.word_errors
error_rate_func = error_rate.cer if decode_cfg.error_rate_type == 'cer' else error_rate.wer
target_transcripts = self.ordid2token(texts, texts_len)
result_transcripts = self.compute_result_transcripts(audio, audio_len)
for utt, target, result in zip(utts, target_transcripts,
result_transcripts):
errors, len_ref = errors_func(target, result)
errors_sum += errors
len_refs += len_ref
num_ins += 1
if fout:
fout.write({"utt": utt, "refs": [target], "hyps": [result]})
logger.info(f"Utt: {utt}")
logger.info(f"Ref: {target}")
logger.info(f"Hyp: {result}")
logger.info(
"Current error rate [%s] = %f" %
(decode_cfg.error_rate_type, error_rate_func(target, result)))
return dict(
errors_sum=errors_sum,
len_refs=len_refs,
num_ins=num_ins,
error_rate=errors_sum / len_refs,
error_rate_type=decode_cfg.error_rate_type)
def compute_result_transcripts(self, audio, audio_len):
result_transcripts = self.model.decode(audio, audio_len)
return result_transcripts
@mp_tools.rank_zero_only
@paddle.no_grad()
def test(self):
logger.info(f"Test Total Examples: {len(self.test_loader.dataset)}")
self.model.eval()
error_rate_type = None
errors_sum, len_refs, num_ins = 0.0, 0, 0
# Initialized the decoder in model
decode_cfg = self.config.decode
vocab_list = self.vocab_list
decode_batch_size = decode_cfg.decode_batch_size
self.model.decoder.init_decoder(
decode_batch_size, vocab_list, decode_cfg.decoding_method,
decode_cfg.lang_model_path, decode_cfg.alpha, decode_cfg.beta,
decode_cfg.beam_size, decode_cfg.cutoff_prob,
decode_cfg.cutoff_top_n, decode_cfg.num_proc_bsearch)
with jsonlines.open(self.args.result_file, 'w') as fout:
for i, batch in enumerate(self.test_loader):
utts, audio, audio_len, texts, texts_len = batch
metrics = self.compute_metrics(utts, audio, audio_len, texts,
texts_len, fout)
errors_sum += metrics['errors_sum']
len_refs += metrics['len_refs']
num_ins += metrics['num_ins']
error_rate_type = metrics['error_rate_type']
logger.info("Error rate [%s] (%d/?) = %f" %
(error_rate_type, num_ins, errors_sum / len_refs))
# logging
msg = "Test: "
msg += "epoch: {}, ".format(self.epoch)
msg += "step: {}, ".format(self.iteration)
msg += "Final error rate [%s] (%d/%d) = %f" % (
error_rate_type, num_ins, num_ins, errors_sum / len_refs)
logger.info(msg)
self.model.decoder.del_decoder()
@paddle.no_grad()
def export(self):
infer_model = DeepSpeech2InferModel.from_pretrained(
self.test_loader, self.config, self.args.checkpoint_path)
infer_model.eval()
static_model = infer_model.export()
logger.info(f"Export code: {static_model.forward.code}")
paddle.jit.save(static_model, self.args.export_path)
class DeepSpeech2ExportTester(DeepSpeech2Tester):
def __init__(self, config, args):
super().__init__(config, args)
self.apply_static = True
self.args = args
@mp_tools.rank_zero_only
@paddle.no_grad()
def test(self):
logger.info(f"Test Total Examples: {len(self.test_loader.dataset)}")
if self.args.enable_auto_log is True:
from paddlespeech.s2t.utils.log import Autolog
self.autolog = Autolog(
batch_size=self.config.decode.decode_batch_size,
model_name="deepspeech2",
model_precision="fp32").getlog()
self.model.eval()
error_rate_type = None
errors_sum, len_refs, num_ins = 0.0, 0, 0
# Initialized the decoder in model
decode_cfg = self.config.decode
vocab_list = self.vocab_list
if self.config.rnn_direction == "forward":
decode_batch_size = 1
elif self.config.rnn_direction == "bidirect":
decode_batch_size = self.test_loader.batch_size
else:
raise Exception("wrong model type")
self.model.decoder.init_decoder(
decode_batch_size, vocab_list, decode_cfg.decoding_method,
decode_cfg.lang_model_path, decode_cfg.alpha, decode_cfg.beta,
decode_cfg.beam_size, decode_cfg.cutoff_prob,
decode_cfg.cutoff_top_n, decode_cfg.num_proc_bsearch)
with jsonlines.open(self.args.result_file, 'w') as fout:
for i, batch in enumerate(self.test_loader):
utts, audio, audio_len, texts, texts_len = batch
metrics = self.compute_metrics(utts, audio, audio_len, texts,
texts_len, fout)
errors_sum += metrics['errors_sum']
len_refs += metrics['len_refs']
num_ins += metrics['num_ins']
error_rate_type = metrics['error_rate_type']
logger.info("Error rate [%s] (%d/?) = %f" %
(error_rate_type, num_ins, errors_sum / len_refs))
# logging
msg = "Test: "
msg += "epoch: {}, ".format(self.epoch)
msg += "step: {}, ".format(self.iteration)
msg += "Final error rate [%s] (%d/%d) = %f" % (
error_rate_type, num_ins, num_ins, errors_sum / len_refs)
logger.info(msg)
if self.args.enable_auto_log is True:
self.autolog.report()
self.model.decoder.del_decoder()
def compute_result_transcripts(self, audio, audio_len):
if self.config.rnn_direction == "forward":
output_probs, output_lens, trans_batch = self.static_forward_online(
audio, audio_len, decoder_chunk_size=1)
result_transcripts = [trans[-1] for trans in trans_batch]
elif self.config.rnn_direction == "bidirect":
output_probs, output_lens = self.static_forward_offline(audio,
audio_len)
batch_size = output_probs.shape[0]
self.model.decoder.reset_decoder(batch_size=batch_size)
self.model.decoder.next(output_probs, output_lens)
trans_best, trans_beam = self.model.decoder.decode()
result_transcripts = trans_best
else:
raise Exception("wrong model type")
self.predictor.clear_intermediate_tensor()
self.predictor.try_shrink_memory()
#replace the <space> with ' '
result_transcripts = [
self._text_featurizer.detokenize(sentence)
for sentence in result_transcripts
]
return result_transcripts
def run_test(self):
"""Do Test/Decode"""
try:
with Timer("Test/Decode Done: {}"):
with self.eval():
self.test()
except KeyboardInterrupt:
exit(-1)
def static_forward_online(self, audio, audio_len,
decoder_chunk_size: int=1):
"""
Parameters
----------
audio (Tensor): shape[B, T, D]
audio_len (Tensor): shape[B]
decoder_chunk_size(int)
Returns
-------
output_probs(numpy.array): shape[B, T, vocab_size]
output_lens(numpy.array): shape[B]
trans(list(list(str))): shape[B, T]
"""
output_probs_list = []
output_lens_list = []
subsampling_rate = self.model.encoder.conv.subsampling_rate
receptive_field_length = self.model.encoder.conv.receptive_field_length
chunk_stride = subsampling_rate * decoder_chunk_size
chunk_size = (decoder_chunk_size - 1
) * subsampling_rate + receptive_field_length
x_batch = audio.numpy()
batch_size, Tmax, x_dim = x_batch.shape
x_len_batch = audio_len.numpy().astype(np.int64)
if (Tmax - chunk_size) % chunk_stride != 0:
# The length of padding for the batch
padding_len_batch = chunk_stride - (Tmax - chunk_size
) % chunk_stride
else:
padding_len_batch = 0
x_list = np.split(x_batch, batch_size, axis=0)
x_len_list = np.split(x_len_batch, batch_size, axis=0)
trans_batch = []
for x, x_len in zip(x_list, x_len_list):
if self.args.enable_auto_log is True:
self.autolog.times.start()
x_len = x_len[0]
assert (chunk_size <= x_len)
if (x_len - chunk_size) % chunk_stride != 0:
padding_len_x = chunk_stride - (x_len - chunk_size
) % chunk_stride
else:
padding_len_x = 0
padding = np.zeros(
(x.shape[0], padding_len_x, x.shape[2]), dtype=x.dtype)
padded_x = np.concatenate([x, padding], axis=1)
num_chunk = (x_len + padding_len_x - chunk_size) / chunk_stride + 1
num_chunk = int(num_chunk)
chunk_state_h_box = np.zeros(
(self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
dtype=x.dtype)
chunk_state_c_box = np.zeros(
(self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
dtype=x.dtype)
input_names = self.predictor.get_input_names()
audio_handle = self.predictor.get_input_handle(input_names[0])
audio_len_handle = self.predictor.get_input_handle(input_names[1])
h_box_handle = self.predictor.get_input_handle(input_names[2])
c_box_handle = self.predictor.get_input_handle(input_names[3])
trans = []
probs_chunk_list = []
probs_chunk_lens_list = []
if self.args.enable_auto_log is True:
# record the model preprocessing time
self.autolog.times.stamp()
self.model.decoder.reset_decoder(batch_size=1)
for i in range(0, num_chunk):
start = i * chunk_stride
end = start + chunk_size
x_chunk = padded_x[:, start:end, :]
if x_len < i * chunk_stride:
x_chunk_lens = 0
else:
x_chunk_lens = min(x_len - i * chunk_stride, chunk_size)
#means the number of input frames in the chunk is not enough for predicting one prob
if (x_chunk_lens < receptive_field_length):
break
x_chunk_lens = np.array([x_chunk_lens])
audio_handle.reshape(x_chunk.shape)
audio_handle.copy_from_cpu(x_chunk)
audio_len_handle.reshape(x_chunk_lens.shape)
audio_len_handle.copy_from_cpu(x_chunk_lens)
h_box_handle.reshape(chunk_state_h_box.shape)
h_box_handle.copy_from_cpu(chunk_state_h_box)
c_box_handle.reshape(chunk_state_c_box.shape)
c_box_handle.copy_from_cpu(chunk_state_c_box)
output_names = self.predictor.get_output_names()
output_handle = self.predictor.get_output_handle(
output_names[0])
output_lens_handle = self.predictor.get_output_handle(
output_names[1])
output_state_h_handle = self.predictor.get_output_handle(
output_names[2])
output_state_c_handle = self.predictor.get_output_handle(
output_names[3])
self.predictor.run()
output_chunk_probs = output_handle.copy_to_cpu()
output_chunk_lens = output_lens_handle.copy_to_cpu()
chunk_state_h_box = output_state_h_handle.copy_to_cpu()
chunk_state_c_box = output_state_c_handle.copy_to_cpu()
self.model.decoder.next(output_chunk_probs, output_chunk_lens)
probs_chunk_list.append(output_chunk_probs)
probs_chunk_lens_list.append(output_chunk_lens)
trans_best, trans_beam = self.model.decoder.decode()
trans.append(trans_best[0])
trans_batch.append(trans)
output_probs = np.concatenate(probs_chunk_list, axis=1)
output_lens = np.sum(probs_chunk_lens_list, axis=0)
vocab_size = output_probs.shape[2]
output_probs_padding_len = Tmax + padding_len_batch - output_probs.shape[
1]
output_probs_padding = np.zeros(
(1, output_probs_padding_len, vocab_size),
dtype=output_probs.
dtype) # The prob padding for a piece of utterance
output_probs = np.concatenate(
[output_probs, output_probs_padding], axis=1)
output_probs_list.append(output_probs)
output_lens_list.append(output_lens)
if self.args.enable_auto_log is True:
# record the model inference time
self.autolog.times.stamp()
# record the post processing time
self.autolog.times.stamp()
self.autolog.times.end()
output_probs = np.concatenate(output_probs_list, axis=0)
output_lens = np.concatenate(output_lens_list, axis=0)
return output_probs, output_lens, trans_batch
def static_forward_offline(self, audio, audio_len):
"""
Parameters
----------
audio (Tensor): shape[B, T, D]
audio_len (Tensor): shape[B]
Returns
-------
output_probs(numpy.array): shape[B, T, vocab_size]
output_lens(numpy.array): shape[B]
"""
x = audio.numpy()
x_len = audio_len.numpy().astype(np.int64)
input_names = self.predictor.get_input_names()
audio_handle = self.predictor.get_input_handle(input_names[0])
audio_len_handle = self.predictor.get_input_handle(input_names[1])
audio_handle.reshape(x.shape)
audio_handle.copy_from_cpu(x)
audio_len_handle.reshape(x_len.shape)
audio_len_handle.copy_from_cpu(x_len)
if self.args.enable_auto_log is True:
self.autolog.times.start()
# record the prefix processing time
self.autolog.times.stamp()
self.predictor.run()
if self.args.enable_auto_log is True:
# record the model inference time
self.autolog.times.stamp()
# record the post processing time
self.autolog.times.stamp()
self.autolog.times.end()
output_names = self.predictor.get_output_names()
output_handle = self.predictor.get_output_handle(output_names[0])
output_lens_handle = self.predictor.get_output_handle(output_names[1])
output_probs = output_handle.copy_to_cpu()
output_lens = output_lens_handle.copy_to_cpu()
return output_probs, output_lens
def setup_model(self):
super().setup_model()
deepspeech_config = inference.Config(
self.args.export_path + ".pdmodel",
self.args.export_path + ".pdiparams")
if (os.environ['CUDA_VISIBLE_DEVICES'].strip() != ''):
deepspeech_config.enable_use_gpu(100, 0)
deepspeech_config.enable_memory_optim()
deepspeech_predictor = inference.create_predictor(deepspeech_config)
self.predictor = deepspeech_predictor
Reference in new issue View Git Blame Copy Permalink