# 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 from typing import Optional import jsonlines import numpy as np import paddle from paddle import distributed as dist from paddle import inference from paddle.io import DataLoader from yacs.config import CfgNode from paddlespeech.s2t.frontend.featurizer.text_featurizer import TextFeaturizer from paddlespeech.s2t.io.collator import SpeechCollator from paddlespeech.s2t.io.dataset import ManifestDataset from paddlespeech.s2t.io.sampler import SortagradBatchSampler from paddlespeech.s2t.io.sampler import SortagradDistributedBatchSampler from paddlespeech.s2t.models.ds2 import DeepSpeech2InferModel from paddlespeech.s2t.models.ds2 import DeepSpeech2Model from paddlespeech.s2t.models.ds2_online import DeepSpeech2InferModelOnline from paddlespeech.s2t.models.ds2_online import DeepSpeech2ModelOnline 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): @classmethod def params(cls, config: Optional[CfgNode]=None) -> CfgNode: # training config default = CfgNode( dict( lr=5e-4, # learning rate lr_decay=1.0, # learning rate decay weight_decay=1e-6, # the coeff of weight decay global_grad_clip=5.0, # the global norm clip n_epoch=50, # train epochs )) if config is not None: config.merge_from_other_cfg(default) return default def __init__(self, config, args): super().__init__(config, args) def train_batch(self, batch_index, batch_data, msg): batch_size = self.config.collator.batch_size accum_grad = self.config.training.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.training.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.model.feat_size = self.train_loader.collate_fn.feature_size config.model.dict_size = self.train_loader.collate_fn.vocab_size else: config.model.feat_size = self.test_loader.collate_fn.feature_size config.model.dict_size = self.test_loader.collate_fn.vocab_size if self.args.model_type == 'offline': model = DeepSpeech2Model.from_config(config.model) elif self.args.model_type == 'online': model = DeepSpeech2ModelOnline.from_config(config.model) else: raise Exception("wrong model type") 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.training.global_grad_clip) lr_scheduler = paddle.optimizer.lr.ExponentialDecay( learning_rate=config.training.lr, gamma=config.training.lr_decay, verbose=True) optimizer = paddle.optimizer.Adam( learning_rate=lr_scheduler, parameters=model.parameters(), weight_decay=paddle.regularizer.L2Decay( config.training.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 config.data.manifest = config.data.train_manifest train_dataset = ManifestDataset.from_config(config) if self.parallel: batch_sampler = SortagradDistributedBatchSampler( train_dataset, batch_size=config.collator.batch_size, num_replicas=None, rank=None, shuffle=True, drop_last=True, sortagrad=config.collator.sortagrad, shuffle_method=config.collator.shuffle_method) else: batch_sampler = SortagradBatchSampler( train_dataset, shuffle=True, batch_size=config.collator.batch_size, drop_last=True, sortagrad=config.collator.sortagrad, shuffle_method=config.collator.shuffle_method) config.collator.keep_transcription_text = False collate_fn_train = SpeechCollator.from_config(config) self.train_loader = DataLoader( train_dataset, batch_sampler=batch_sampler, collate_fn=collate_fn_train, num_workers=config.collator.num_workers) # dev config.data.manifest = config.data.dev_manifest dev_dataset = ManifestDataset.from_config(config) config.collator.augmentation_config = "" config.collator.keep_transcription_text = False collate_fn_dev = SpeechCollator.from_config(config) self.valid_loader = DataLoader( dev_dataset, batch_size=int(config.collator.batch_size), shuffle=False, drop_last=False, collate_fn=collate_fn_dev, num_workers=config.collator.num_workers) logger.info("Setup train/valid Dataloader!") else: # test config.data.manifest = config.data.test_manifest test_dataset = ManifestDataset.from_config(config) config.collator.augmentation_config = "" config.collator.keep_transcription_text = True collate_fn_test = SpeechCollator.from_config(config) self.test_loader = DataLoader( test_dataset, batch_size=config.decoding.batch_size, shuffle=False, drop_last=False, collate_fn=collate_fn_test, num_workers=config.collator.num_workers) logger.info("Setup test Dataloader!") class DeepSpeech2Tester(DeepSpeech2Trainer): @classmethod def params(cls, config: Optional[CfgNode]=None) -> CfgNode: # testing config default = CfgNode( dict( alpha=2.5, # Coef of LM for beam search. beta=0.3, # Coef of WC for beam search. cutoff_prob=1.0, # Cutoff probability for pruning. cutoff_top_n=40, # Cutoff number for pruning. lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm', # Filepath for language model. decoding_method='ctc_beam_search', # Decoding method. Options: ctc_beam_search, ctc_greedy error_rate_type='wer', # Error rate type for evaluation. Options `wer`, 'cer' num_proc_bsearch=8, # # of CPUs for beam search. beam_size=500, # Beam search width. batch_size=128, # decoding batch size )) if config is not None: config.merge_from_other_cfg(default) return default def __init__(self, config, args): super().__init__(config, args) self._text_featurizer = TextFeaturizer( unit_type=config.collator.unit_type, vocab_filepath=None) 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(''.join([chr(i) for i in ids])) return trans def compute_metrics(self, utts, audio, audio_len, texts, texts_len, fout=None): cfg = self.config.decoding errors_sum, len_refs, num_ins = 0.0, 0, 0 errors_func = error_rate.char_errors if cfg.error_rate_type == 'cer' else error_rate.word_errors error_rate_func = error_rate.cer if cfg.error_rate_type == 'cer' else error_rate.wer vocab_list = self.test_loader.collate_fn.vocab_list target_transcripts = self.ordid2token(texts, texts_len) result_transcripts = self.compute_result_transcripts(audio, audio_len, vocab_list, cfg) 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, "ref": target, "hyp": result}) logger.info(f"Utt: {utt}") logger.info(f"Ref: {target}") logger.info(f"Hyp: {result}") logger.info("Current error rate [%s] = %f" % (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=cfg.error_rate_type) def compute_result_transcripts(self, audio, audio_len, vocab_list, cfg): result_transcripts = self.model.decode( audio, audio_len, vocab_list, decoding_method=cfg.decoding_method, lang_model_path=cfg.lang_model_path, beam_alpha=cfg.alpha, beam_beta=cfg.beta, beam_size=cfg.beam_size, cutoff_prob=cfg.cutoff_prob, cutoff_top_n=cfg.cutoff_top_n, num_processes=cfg.num_proc_bsearch) 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() cfg = self.config error_rate_type = None errors_sum, len_refs, num_ins = 0.0, 0, 0 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) @paddle.no_grad() def export(self): if self.args.model_type == 'offline': infer_model = DeepSpeech2InferModel.from_pretrained( self.test_loader, self.config, self.args.checkpoint_path) elif self.args.model_type == 'online': infer_model = DeepSpeech2InferModelOnline.from_pretrained( self.test_loader, self.config, self.args.checkpoint_path) else: raise Exception("wrong model type") infer_model.eval() feat_dim = self.test_loader.collate_fn.feature_size 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 == True: from paddlespeech.s2t.utils.log import Autolog self.autolog = Autolog( batch_size=self.config.decoding.batch_size, model_name="deepspeech2", model_precision="fp32").getlog() self.model.eval() cfg = self.config error_rate_type = None errors_sum, len_refs, num_ins = 0.0, 0, 0 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 == True: self.autolog.report() def compute_result_transcripts(self, audio, audio_len, vocab_list, cfg): if self.args.model_type == "online": output_probs, output_lens = self.static_forward_online(audio, audio_len) elif self.args.model_type == "offline": output_probs, output_lens = self.static_forward_offline(audio, audio_len) else: raise Exception("wrong model type") self.predictor.clear_intermediate_tensor() self.predictor.try_shrink_memory() self.model.decoder.init_decode(cfg.alpha, cfg.beta, cfg.lang_model_path, vocab_list, cfg.decoding_method) 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 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 == 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 == 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 == 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 == True: self.autolog.times.start() # record the prefix processing time self.autolog.times.stamp() self.predictor.run() if self.args.enable_auto_log == 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