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673 lines
27 KiB
673 lines
27 KiB
# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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"""Contains DeepSpeech2 and DeepSpeech2Online model."""
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import os
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import time
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from collections import defaultdict
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from contextlib import nullcontext
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from typing import Optional
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import jsonlines
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import numpy as np
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import paddle
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from paddle import distributed as dist
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from paddle import inference
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from paddle.io import DataLoader
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from yacs.config import CfgNode
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from paddlespeech.s2t.frontend.featurizer.text_featurizer import TextFeaturizer
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from paddlespeech.s2t.io.collator import SpeechCollator
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from paddlespeech.s2t.io.dataset import ManifestDataset
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from paddlespeech.s2t.io.sampler import SortagradBatchSampler
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from paddlespeech.s2t.io.sampler import SortagradDistributedBatchSampler
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from paddlespeech.s2t.models.ds2 import DeepSpeech2InferModel
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from paddlespeech.s2t.models.ds2 import DeepSpeech2Model
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from paddlespeech.s2t.models.ds2_online import DeepSpeech2InferModelOnline
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from paddlespeech.s2t.models.ds2_online import DeepSpeech2ModelOnline
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from paddlespeech.s2t.training.gradclip import ClipGradByGlobalNormWithLog
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from paddlespeech.s2t.training.reporter import report
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from paddlespeech.s2t.training.timer import Timer
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from paddlespeech.s2t.training.trainer import Trainer
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from paddlespeech.s2t.utils import error_rate
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from paddlespeech.s2t.utils import layer_tools
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from paddlespeech.s2t.utils import mp_tools
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from paddlespeech.s2t.utils.log import Log
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from paddlespeech.s2t.utils.utility import UpdateConfig
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logger = Log(__name__).getlog()
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class DeepSpeech2Trainer(Trainer):
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@classmethod
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def params(cls, config: Optional[CfgNode]=None) -> CfgNode:
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# training config
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default = CfgNode(
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dict(
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lr=5e-4, # learning rate
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lr_decay=1.0, # learning rate decay
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weight_decay=1e-6, # the coeff of weight decay
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global_grad_clip=5.0, # the global norm clip
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n_epoch=50, # train epochs
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))
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if config is not None:
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config.merge_from_other_cfg(default)
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return default
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def __init__(self, config, args):
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super().__init__(config, args)
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def train_batch(self, batch_index, batch_data, msg):
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batch_size = self.config.collator.batch_size
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accum_grad = self.config.training.accum_grad
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start = time.time()
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# forward
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utt, audio, audio_len, text, text_len = batch_data
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loss = self.model(audio, audio_len, text, text_len)
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losses_np = {
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'train_loss': float(loss),
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}
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# loss backward
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if (batch_index + 1) % accum_grad != 0:
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# Disable gradient synchronizations across DDP processes.
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# Within this context, gradients will be accumulated on module
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# variables, which will later be synchronized.
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context = self.model.no_sync if (hasattr(self.model, "no_sync") and
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self.parallel) else nullcontext
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else:
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# Used for single gpu training and DDP gradient synchronization
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# processes.
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context = nullcontext
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with context():
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loss.backward()
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layer_tools.print_grads(self.model, print_func=None)
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# optimizer step
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if (batch_index + 1) % accum_grad == 0:
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self.optimizer.step()
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self.optimizer.clear_grad()
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self.iteration += 1
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iteration_time = time.time() - start
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for k, v in losses_np.items():
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report(k, v)
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report("batch_size", batch_size)
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report("accum", accum_grad)
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report("step_cost", iteration_time)
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if dist.get_rank() == 0 and self.visualizer:
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for k, v in losses_np.items():
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# `step -1` since we update `step` after optimizer.step().
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self.visualizer.add_scalar("train/{}".format(k), v,
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self.iteration - 1)
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@paddle.no_grad()
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def valid(self):
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logger.info(f"Valid Total Examples: {len(self.valid_loader.dataset)}")
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self.model.eval()
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valid_losses = defaultdict(list)
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num_seen_utts = 1
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total_loss = 0.0
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for i, batch in enumerate(self.valid_loader):
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utt, audio, audio_len, text, text_len = batch
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loss = self.model(audio, audio_len, text, text_len)
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if paddle.isfinite(loss):
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num_utts = batch[1].shape[0]
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num_seen_utts += num_utts
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total_loss += float(loss) * num_utts
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valid_losses['val_loss'].append(float(loss))
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if (i + 1) % self.config.training.log_interval == 0:
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valid_dump = {k: np.mean(v) for k, v in valid_losses.items()}
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valid_dump['val_history_loss'] = total_loss / num_seen_utts
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# logging
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msg = f"Valid: Rank: {dist.get_rank()}, "
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msg += "epoch: {}, ".format(self.epoch)
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msg += "step: {}, ".format(self.iteration)
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msg += "batch : {}/{}, ".format(i + 1, len(self.valid_loader))
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msg += ', '.join('{}: {:>.6f}'.format(k, v)
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for k, v in valid_dump.items())
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logger.info(msg)
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logger.info('Rank {} Val info val_loss {}'.format(
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dist.get_rank(), total_loss / num_seen_utts))
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return total_loss, num_seen_utts
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def setup_model(self):
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config = self.config.clone()
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with UpdateConfig(config):
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if self.train:
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config.model.input_dim = self.train_loader.collate_fn.feature_size
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config.model.output_dim = self.train_loader.collate_fn.vocab_size
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else:
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config.model.input_dim = self.test_loader.collate_fn.feature_size
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config.model.output_dim = self.test_loader.collate_fn.vocab_size
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if self.args.model_type == 'offline':
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model = DeepSpeech2Model.from_config(config.model)
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elif self.args.model_type == 'online':
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model = DeepSpeech2ModelOnline.from_config(config.model)
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else:
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raise Exception("wrong model type")
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if self.parallel:
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model = paddle.DataParallel(model)
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logger.info(f"{model}")
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layer_tools.print_params(model, logger.info)
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self.model = model
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logger.info("Setup model!")
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if not self.train:
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return
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grad_clip = ClipGradByGlobalNormWithLog(
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config.training.global_grad_clip)
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lr_scheduler = paddle.optimizer.lr.ExponentialDecay(
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learning_rate=config.training.lr,
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gamma=config.training.lr_decay,
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verbose=True)
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optimizer = paddle.optimizer.Adam(
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learning_rate=lr_scheduler,
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parameters=model.parameters(),
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weight_decay=paddle.regularizer.L2Decay(
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config.training.weight_decay),
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grad_clip=grad_clip)
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self.optimizer = optimizer
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self.lr_scheduler = lr_scheduler
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logger.info("Setup optimizer/lr_scheduler!")
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def setup_dataloader(self):
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config = self.config.clone()
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config.defrost()
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if self.train:
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# train
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config.data.manifest = config.data.train_manifest
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train_dataset = ManifestDataset.from_config(config)
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if self.parallel:
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batch_sampler = SortagradDistributedBatchSampler(
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train_dataset,
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batch_size=config.collator.batch_size,
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num_replicas=None,
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rank=None,
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shuffle=True,
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drop_last=True,
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sortagrad=config.collator.sortagrad,
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shuffle_method=config.collator.shuffle_method)
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else:
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batch_sampler = SortagradBatchSampler(
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train_dataset,
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shuffle=True,
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batch_size=config.collator.batch_size,
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drop_last=True,
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sortagrad=config.collator.sortagrad,
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shuffle_method=config.collator.shuffle_method)
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config.collator.keep_transcription_text = False
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collate_fn_train = SpeechCollator.from_config(config)
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self.train_loader = DataLoader(
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train_dataset,
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batch_sampler=batch_sampler,
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collate_fn=collate_fn_train,
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num_workers=config.collator.num_workers)
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# dev
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config.data.manifest = config.data.dev_manifest
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dev_dataset = ManifestDataset.from_config(config)
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config.collator.augmentation_config = ""
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config.collator.keep_transcription_text = False
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collate_fn_dev = SpeechCollator.from_config(config)
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self.valid_loader = DataLoader(
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dev_dataset,
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batch_size=int(config.collator.batch_size),
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shuffle=False,
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drop_last=False,
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collate_fn=collate_fn_dev,
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num_workers=config.collator.num_workers)
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logger.info("Setup train/valid Dataloader!")
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else:
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# test
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config.data.manifest = config.data.test_manifest
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test_dataset = ManifestDataset.from_config(config)
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config.collator.augmentation_config = ""
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config.collator.keep_transcription_text = True
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collate_fn_test = SpeechCollator.from_config(config)
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self.test_loader = DataLoader(
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test_dataset,
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batch_size=config.decoding.batch_size,
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shuffle=False,
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drop_last=False,
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collate_fn=collate_fn_test,
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num_workers=config.collator.num_workers)
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logger.info("Setup test Dataloader!")
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class DeepSpeech2Tester(DeepSpeech2Trainer):
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@classmethod
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def params(cls, config: Optional[CfgNode]=None) -> CfgNode:
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# testing config
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default = CfgNode(
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dict(
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alpha=2.5, # Coef of LM for beam search.
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beta=0.3, # Coef of WC for beam search.
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cutoff_prob=1.0, # Cutoff probability for pruning.
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cutoff_top_n=40, # Cutoff number for pruning.
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lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm', # Filepath for language model.
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decoding_method='ctc_beam_search', # Decoding method. Options: ctc_beam_search, ctc_greedy
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error_rate_type='wer', # Error rate type for evaluation. Options `wer`, 'cer'
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num_proc_bsearch=8, # # of CPUs for beam search.
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beam_size=500, # Beam search width.
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batch_size=128, # decoding batch size
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))
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if config is not None:
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config.merge_from_other_cfg(default)
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return default
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def __init__(self, config, args):
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super().__init__(config, args)
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self._text_featurizer = TextFeaturizer(
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unit_type=config.collator.unit_type, vocab_filepath=None)
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def ordid2token(self, texts, texts_len):
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""" ord() id to chr() chr """
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trans = []
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for text, n in zip(texts, texts_len):
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n = n.numpy().item()
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ids = text[:n]
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trans.append(''.join([chr(i) for i in ids]))
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return trans
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def compute_metrics(self,
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utts,
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audio,
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audio_len,
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texts,
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texts_len,
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fout=None):
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cfg = self.config.decoding
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errors_sum, len_refs, num_ins = 0.0, 0, 0
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errors_func = error_rate.char_errors if cfg.error_rate_type == 'cer' else error_rate.word_errors
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error_rate_func = error_rate.cer if cfg.error_rate_type == 'cer' else error_rate.wer
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vocab_list = self.test_loader.collate_fn.vocab_list
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target_transcripts = self.ordid2token(texts, texts_len)
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result_transcripts = self.compute_result_transcripts(audio, audio_len,
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vocab_list, cfg)
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for utt, target, result in zip(utts, target_transcripts,
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result_transcripts):
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errors, len_ref = errors_func(target, result)
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errors_sum += errors
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len_refs += len_ref
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num_ins += 1
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if fout:
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fout.write({"utt": utt, "ref": target, "hyp": result})
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logger.info(f"Utt: {utt}")
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logger.info(f"Ref: {target}")
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logger.info(f"Hyp: {result}")
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logger.info("Current error rate [%s] = %f" %
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(cfg.error_rate_type, error_rate_func(target, result)))
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return dict(
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errors_sum=errors_sum,
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len_refs=len_refs,
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num_ins=num_ins,
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error_rate=errors_sum / len_refs,
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error_rate_type=cfg.error_rate_type)
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def compute_result_transcripts(self, audio, audio_len, vocab_list, cfg):
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result_transcripts = self.model.decode(
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audio,
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audio_len,
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vocab_list,
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decoding_method=cfg.decoding_method,
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lang_model_path=cfg.lang_model_path,
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beam_alpha=cfg.alpha,
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beam_beta=cfg.beta,
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beam_size=cfg.beam_size,
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cutoff_prob=cfg.cutoff_prob,
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cutoff_top_n=cfg.cutoff_top_n,
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num_processes=cfg.num_proc_bsearch)
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return result_transcripts
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@mp_tools.rank_zero_only
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@paddle.no_grad()
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def test(self):
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logger.info(f"Test Total Examples: {len(self.test_loader.dataset)}")
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self.model.eval()
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cfg = self.config
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error_rate_type = None
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errors_sum, len_refs, num_ins = 0.0, 0, 0
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with jsonlines.open(self.args.result_file, 'w') as fout:
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for i, batch in enumerate(self.test_loader):
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utts, audio, audio_len, texts, texts_len = batch
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metrics = self.compute_metrics(utts, audio, audio_len, texts,
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texts_len, fout)
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errors_sum += metrics['errors_sum']
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len_refs += metrics['len_refs']
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num_ins += metrics['num_ins']
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error_rate_type = metrics['error_rate_type']
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logger.info("Error rate [%s] (%d/?) = %f" %
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(error_rate_type, num_ins, errors_sum / len_refs))
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# logging
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msg = "Test: "
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msg += "epoch: {}, ".format(self.epoch)
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msg += "step: {}, ".format(self.iteration)
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msg += "Final error rate [%s] (%d/%d) = %f" % (
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error_rate_type, num_ins, num_ins, errors_sum / len_refs)
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logger.info(msg)
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@paddle.no_grad()
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def export(self):
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if self.args.model_type == 'offline':
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infer_model = DeepSpeech2InferModel.from_pretrained(
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self.test_loader, self.config, self.args.checkpoint_path)
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elif self.args.model_type == 'online':
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infer_model = DeepSpeech2InferModelOnline.from_pretrained(
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self.test_loader, self.config, self.args.checkpoint_path)
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else:
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raise Exception("wrong model type")
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infer_model.eval()
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feat_dim = self.test_loader.collate_fn.feature_size
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static_model = infer_model.export()
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logger.info(f"Export code: {static_model.forward.code}")
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paddle.jit.save(static_model, self.args.export_path)
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class DeepSpeech2ExportTester(DeepSpeech2Tester):
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def __init__(self, config, args):
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super().__init__(config, args)
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self.apply_static = True
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self.args = args
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@mp_tools.rank_zero_only
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@paddle.no_grad()
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def test(self):
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logger.info(f"Test Total Examples: {len(self.test_loader.dataset)}")
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if self.args.enable_auto_log is True:
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from paddlespeech.s2t.utils.log import Autolog
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self.autolog = Autolog(
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batch_size=self.config.decoding.batch_size,
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model_name="deepspeech2",
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model_precision="fp32").getlog()
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self.model.eval()
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cfg = self.config
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error_rate_type = None
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errors_sum, len_refs, num_ins = 0.0, 0, 0
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with jsonlines.open(self.args.result_file, 'w') as fout:
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for i, batch in enumerate(self.test_loader):
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utts, audio, audio_len, texts, texts_len = batch
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metrics = self.compute_metrics(utts, audio, audio_len, texts,
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texts_len, fout)
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errors_sum += metrics['errors_sum']
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len_refs += metrics['len_refs']
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num_ins += metrics['num_ins']
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error_rate_type = metrics['error_rate_type']
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logger.info("Error rate [%s] (%d/?) = %f" %
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(error_rate_type, num_ins, errors_sum / len_refs))
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# logging
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msg = "Test: "
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msg += "epoch: {}, ".format(self.epoch)
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msg += "step: {}, ".format(self.iteration)
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msg += "Final error rate [%s] (%d/%d) = %f" % (
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error_rate_type, num_ins, num_ins, errors_sum / len_refs)
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logger.info(msg)
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if self.args.enable_auto_log is True:
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self.autolog.report()
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def compute_result_transcripts(self, audio, audio_len, vocab_list, cfg):
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if self.args.model_type == "online":
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output_probs, output_lens = self.static_forward_online(audio,
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audio_len)
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elif self.args.model_type == "offline":
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output_probs, output_lens = self.static_forward_offline(audio,
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audio_len)
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else:
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raise Exception("wrong model type")
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self.predictor.clear_intermediate_tensor()
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self.predictor.try_shrink_memory()
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self.model.decoder.init_decode(cfg.alpha, cfg.beta, cfg.lang_model_path,
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vocab_list, cfg.decoding_method)
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result_transcripts = self.model.decoder.decode_probs(
|
|
output_probs, output_lens, vocab_list, cfg.decoding_method,
|
|
cfg.lang_model_path, cfg.alpha, cfg.beta, cfg.beam_size,
|
|
cfg.cutoff_prob, cfg.cutoff_top_n, cfg.num_proc_bsearch)
|
|
#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]
|
|
"""
|
|
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:
|
|
padding_len_batch = chunk_stride - (
|
|
Tmax - chunk_size
|
|
) % chunk_stride # The length of padding for the batch
|
|
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)
|
|
|
|
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.model.num_rnn_layers, 1,
|
|
self.config.model.rnn_layer_size),
|
|
dtype=x.dtype)
|
|
chunk_state_c_box = np.zeros(
|
|
(self.config.model.num_rnn_layers, 1,
|
|
self.config.model.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])
|
|
|
|
probs_chunk_list = []
|
|
probs_chunk_lens_list = []
|
|
if self.args.enable_auto_log is True:
|
|
# record the model preprocessing time
|
|
self.autolog.times.stamp()
|
|
|
|
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)
|
|
|
|
if (x_chunk_lens <
|
|
receptive_field_length): #means the number of input frames in the chunk is not enough for predicting one prob
|
|
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()
|
|
|
|
probs_chunk_list.append(output_chunk_probs)
|
|
probs_chunk_lens_list.append(output_chunk_lens)
|
|
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
|
|
|
|
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
|