mirror of
https://github.com/k2-fsa/icefall.git
synced 2025-08-10 10:32:17 +00:00
* Fix torch.nn.Embedding error for torch below 1.8.0 * Changes to fbank computation, use lilcom chunky writer * Add min in q,k,v of attention * Remove learnable offset, use relu instead. * Experiments based on SpecAugment change * Merge specaug change from Mingshuang. * Use much more aggressive SpecAug setup * Fix to num_feature_masks bug I introduced; reduce max_frames_mask_fraction 0.4->0.3 * Change p=0.5->0.9, mask_fraction 0.3->0.2 * Change p=0.9 to p=0.8 in SpecAug * Fix num_time_masks code; revert 0.8 to 0.9 * Change max_frames from 0.2 to 0.15 * Remove ReLU in attention * Adding diagnostics code... * Refactor/simplify ConformerEncoder * First version of rand-combine iterated-training-like idea. * Improvements to diagnostics (RE those with 1 dim * Add pelu to this good-performing setup.. * Small bug fixes/imports * Add baseline for the PeLU expt, keeping only the small normalization-related changes. * pelu_base->expscale, add 2xExpScale in subsampling, and in feedforward units. * Double learning rate of exp-scale units * Combine ExpScale and swish for memory reduction * Add import * Fix backprop bug * Fix bug in diagnostics * Increase scale on Scale from 4 to 20 * Increase scale from 20 to 50. * Fix duplicate Swish; replace norm+swish with swish+exp-scale in convolution module * Reduce scale from 50 to 20 * Add deriv-balancing code * Double the threshold in brelu; slightly increase max_factor. * Fix exp dir * Convert swish nonlinearities to ReLU * Replace relu with swish-squared. * Restore ConvolutionModule to state before changes; change all Swish,Swish(Swish) to SwishOffset. * Replace norm on input layer with scale of 0.1. * Extensions to diagnostics code * Update diagnostics * Add BasicNorm module * Replace most normalizations with scales (still have norm in conv) * Change exp dir * Replace norm in ConvolutionModule with a scaling factor. * use nonzero threshold in DerivBalancer * Add min-abs-value 0.2 * Fix dirname * Change min-abs threshold from 0.2 to 0.5 * Scale up pos_bias_u and pos_bias_v before use. * Reduce max_factor to 0.01 * Fix q*scaling logic * Change max_factor in DerivBalancer from 0.025 to 0.01; fix scaling code. * init 1st conv module to smaller variance * Change how scales are applied; fix residual bug * Reduce min_abs from 0.5 to 0.2 * Introduce in_scale=0.5 for SwishExpScale * Fix scale from 0.5 to 2.0 as I really intended.. * Set scaling on SwishExpScale * Add identity pre_norm_final for diagnostics. * Add learnable post-scale for mha * Fix self.post-scale-mha * Another rework, use scales on linear/conv * Change dir name * Reduce initial scaling of modules * Bug-fix RE bias * Cosmetic change * Reduce initial_scale. * Replace ExpScaleRelu with DoubleSwish() * DoubleSwish fix * Use learnable scales for joiner and decoder * Add max-abs-value constraint in DerivBalancer * Add max-abs-value * Change dir name * Remove ExpScale in feedforward layes. * Reduce max-abs limit from 1000 to 100; introduce 2 DerivBalancer modules in conv layer. * Make DoubleSwish more memory efficient * Reduce constraints from deriv-balancer in ConvModule. * Add warmup mode * Remove max-positive constraint in deriv-balancing; add second DerivBalancer in conv module. * Add some extra info to diagnostics * Add deriv-balancer at output of embedding. * Add more stats. * Make epsilon in BasicNorm learnable, optionally. * Draft of 0mean changes.. * Rework of initialization * Fix typo * Remove dead code * Modifying initialization from normal->uniform; add initial_scale when initializing * bug fix re sqrt * Remove xscale from pos_embedding * Remove some dead code. * Cosmetic changes/renaming things * Start adding some files.. * Add more files.. * update decode.py file type * Add remaining files in pruned_transducer_stateless2 * Fix diagnostics-getting code * Scale down pruned loss in warmup mode * Reduce warmup scale on pruned loss form 0.1 to 0.01. * Remove scale_speed, make swish deriv more efficient. * Cosmetic changes to swish * Double warm_step * Fix bug with import * Change initial std from 0.05 to 0.025. * Set also scale for embedding to 0.025. * Remove logging code that broke with newer Lhotse; fix bug with pruned_loss * Add norm+balancer to VggSubsampling * Incorporate changes from master into pruned_transducer_stateless2. * Add max-abs=6, debugged version * Change 0.025,0.05 to 0.01 in initializations * Fix balancer code * Whitespace fix * Reduce initial pruned_loss scale from 0.01 to 0.0 * Increase warm_step (and valid_interval) * Change max-abs from 6 to 10 * Change how warmup works. * Add changes from master to decode.py, train.py * Simplify the warmup code; max_abs 10->6 * Make warmup work by scaling layer contributions; leave residual layer-drop * Fix bug * Fix test mode with random layer dropout * Add random-number-setting function in dataloader * Fix/patch how fix_random_seed() is imported. * Reduce layer-drop prob * Reduce layer-drop prob after warmup to 1 in 100 * Change power of lr-schedule from -0.5 to -0.333 * Increase model_warm_step to 4k * Change max-keep-prob to 0.95 * Refactoring and simplifying conformer and frontend * Rework conformer, remove some code. * Reduce 1st conv channels from 64 to 32 * Add another convolutional layer * Fix padding bug * Remove dropout in output layer * Reduce speed of some components * Initial refactoring to remove unnecessary vocab_size * Fix RE identity * Bug-fix * Add final dropout to conformer * Remove some un-used code * Replace nn.Linear with ScaledLinear in simple joiner * Make 2 projections.. * Reduce initial_speed * Use initial_speed=0.5 * Reduce initial_speed further from 0.5 to 0.25 * Reduce initial_speed from 0.5 to 0.25 * Change how warmup is applied. * Bug fix to warmup_scale * Fix test-mode * Remove final dropout * Make layer dropout rate 0.075, was 0.1. * First draft of model rework * Various bug fixes * Change learning speed of simple_lm_proj * Revert transducer_stateless/ to state in upstream/master * Fix to joiner to allow different dims * Some cleanups * Make training more efficient, avoid redoing some projections. * Change how warm-step is set * First draft of new approach to learning rates + init * Some fixes.. * Change initialization to 0.25 * Fix type of parameter * Fix weight decay formula by adding 1/1-beta * Fix weight decay formula by adding 1/1-beta * Fix checkpoint-writing * Fix to reading scheudler from optim * Simplified optimizer, rework somet things.. * Reduce model_warm_step from 4k to 3k * Fix bug in lambda * Bug-fix RE sign of target_rms * Changing initial_speed from 0.25 to 01 * Change some defaults in LR-setting rule. * Remove initial_speed * Set new scheduler * Change exponential part of lrate to be epoch based * Fix bug * Set 2n rule.. * Implement 2o schedule * Make lrate rule more symmetric * Implement 2p version of learning rate schedule. * Refactor how learning rate is set. * Fix import * Modify init (#301) * update icefall/__init__.py to import more common functions. * update icefall/__init__.py * make imports style consistent. * exclude black check for icefall/__init__.py in pyproject.toml. * Minor fixes for logging (#296) * Minor fixes for logging * Minor fix * Fix dir names * Modify beam search to be efficient with current joienr * Fix adding learning rate to tensorboard * Fix docs in optim.py * Support mix precision training on the reworked model (#305) * Add mix precision support * Minor fixes * Minor fixes * Minor fixes * Tedlium3 pruned transducer stateless (#261) * update tedlium3-pruned-transducer-stateless-codes * update README.md * update README.md * add fast beam search for decoding * do a change for RESULTS.md * do a change for RESULTS.md * do a fix * do some changes for pruned RNN-T * Add mix precision support * Minor fixes * Minor fixes * Updating RESULTS.md; fix in beam_search.py * Fix rebase * Code style check for librispeech pruned transducer stateless2 (#308) * Update results for tedlium3 pruned RNN-T (#307) * Update README.md * Fix CI errors. (#310) * Add more results * Fix tensorboard log location * Add one more epoch of full expt * fix comments * Add results for mixed precision with max-duration 300 * Changes for pretrained.py (tedlium3 pruned RNN-T) (#311) * GigaSpeech recipe (#120) * initial commit * support download, data prep, and fbank * on-the-fly feature extraction by default * support BPE based lang * support HLG for BPE * small fix * small fix * chunked feature extraction by default * Compute features for GigaSpeech by splitting the manifest. * Fixes after review. * Split manifests into 2000 pieces. * set audio duration mismatch tolerance to 0.01 * small fix * add conformer training recipe * Add conformer.py without pre-commit checking * lazy loading and use SingleCutSampler * DynamicBucketingSampler * use KaldifeatFbank to compute fbank for musan * use pretrained language model and lexicon * use 3gram to decode, 4gram to rescore * Add decode.py * Update .flake8 * Delete compute_fbank_gigaspeech.py * Use BucketingSampler for valid and test dataloader * Update params in train.py * Use bpe_500 * update params in decode.py * Decrease num_paths while CUDA OOM * Added README * Update RESULTS * black * Decrease num_paths while CUDA OOM * Decode with post-processing * Update results * Remove lazy_load option * Use default `storage_type` * Keep the original tolerance * Use split-lazy * black * Update pretrained model Co-authored-by: Fangjun Kuang <csukuangfj@gmail.com> * Add LG decoding (#277) * Add LG decoding * Add log weight pushing * Minor fixes * Support computing RNN-T loss with torchaudio (#316) * Support modified beam search decoding for streaming inference with Emformer model. * Formatted imports. * Update results for torchaudio RNN-T. (#322) * Fixed streaming decoding codes for emformer model. * Fixed docs. * Sorted imports for transducer_emformer/streaming_feature_extractor.py * Minor fix for transducer_emformer/streaming_feature_extractor.py Co-authored-by: pkufool <wkang@pku.org.cn> Co-authored-by: Daniel Povey <dpovey@gmail.com> Co-authored-by: Mingshuang Luo <37799481+luomingshuang@users.noreply.github.com> Co-authored-by: Fangjun Kuang <csukuangfj@gmail.com> Co-authored-by: Guo Liyong <guonwpu@qq.com> Co-authored-by: Wang, Guanbo <wgb14@outlook.com>
749 lines
22 KiB
Python
Executable File
749 lines
22 KiB
Python
Executable File
#!/usr/bin/env python3
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#
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# Copyright 2021 Xiaomi Corporation (Author: Fangjun Kuang)
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#
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# See ../../../../LICENSE for clarification regarding multiple authors
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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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import argparse
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import logging
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import warnings
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from pathlib import Path
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from typing import List, Optional, Tuple
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import k2
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import numpy as np
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import sentencepiece as spm
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import torch
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import torch.nn as nn
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from asr_datamodule import LibriSpeechAsrDataModule
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from beam_search import Hypothesis, HypothesisList, get_hyps_shape
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from emformer import LOG_EPSILON, stack_states, unstack_states
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from streaming_feature_extractor import FeatureExtractionStream
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from train import add_model_arguments, get_params, get_transducer_model
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from icefall.checkpoint import (
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average_checkpoints,
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find_checkpoints,
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load_checkpoint,
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)
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from icefall.utils import AttributeDict, setup_logger
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def get_parser():
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parser = argparse.ArgumentParser(
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formatter_class=argparse.ArgumentDefaultsHelpFormatter
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)
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parser.add_argument(
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"--epoch",
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type=int,
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default=28,
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help="It specifies the checkpoint to use for decoding."
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"Note: Epoch counts from 0.",
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)
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parser.add_argument(
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"--avg",
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type=int,
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default=15,
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help="Number of checkpoints to average. Automatically select "
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"consecutive checkpoints before the checkpoint specified by "
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"'--epoch'. ",
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)
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parser.add_argument(
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"--avg-last-n",
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type=int,
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default=0,
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help="""If positive, --epoch and --avg are ignored and it
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will use the last n checkpoints exp_dir/checkpoint-xxx.pt
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where xxx is the number of processed batches while
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saving that checkpoint.
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""",
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)
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parser.add_argument(
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"--exp-dir",
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type=str,
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default="transducer_emformer/exp",
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help="The experiment dir",
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)
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parser.add_argument(
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"--bpe-model",
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type=str,
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default="data/lang_bpe_500/bpe.model",
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help="Path to the BPE model",
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)
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parser.add_argument(
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"--decoding-method",
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type=str,
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default="greedy_search",
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help="""Possible values are:
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- greedy_search
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- beam_search
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- modified_beam_search
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- fast_beam_search
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""",
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)
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parser.add_argument(
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"--beam-size",
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type=int,
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default=4,
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help="""An interger indicating how many candidates we will keep for each
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frame. Used only when --decoding-method is beam_search or
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modified_beam_search.""",
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)
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parser.add_argument(
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"--beam",
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type=float,
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default=4,
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help="""A floating point value to calculate the cutoff score during beam
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search (i.e., `cutoff = max-score - beam`), which is the same as the
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`beam` in Kaldi.
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Used only when --decoding-method is fast_beam_search""",
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)
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parser.add_argument(
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"--max-contexts",
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type=int,
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default=4,
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help="""Used only when --decoding-method is
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fast_beam_search""",
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)
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parser.add_argument(
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"--max-states",
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type=int,
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default=8,
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help="""Used only when --decoding-method is
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fast_beam_search""",
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)
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parser.add_argument(
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"--context-size",
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type=int,
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default=2,
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help="The context size in the decoder. 1 means bigram; "
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"2 means tri-gram",
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)
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parser.add_argument(
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"--max-sym-per-frame",
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type=int,
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default=1,
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help="""Maximum number of symbols per frame.
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Used only when --decoding_method is greedy_search""",
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)
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parser.add_argument(
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"--sampling-rate",
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type=float,
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default=16000,
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help="Sample rate of the audio",
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)
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add_model_arguments(parser)
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return parser
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class StreamingAudioSamples(object):
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"""This class takes as input a list of audio samples and returns
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them in a streaming fashion.
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"""
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def __init__(self, samples: List[torch.Tensor]) -> None:
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"""
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Args:
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samples:
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A list of audio samples. Each entry is a 1-D tensor of dtype
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torch.float32, containing the audio samples of an utterance.
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"""
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self.samples = samples
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self.cur_indexes = [0] * len(self.samples)
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@property
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def done(self) -> bool:
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"""Return True if all samples have been processed.
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Return False otherwise.
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"""
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for i, samples in zip(self.cur_indexes, self.samples):
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if i < samples.numel():
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return False
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return True
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def get_next(self) -> List[torch.Tensor]:
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"""Return a list of audio samples. Each entry may have different
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lengths. It is OK if an entry contains no samples at all, which
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means it reaches the end of the utterance.
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"""
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ans = []
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num = [1024] * len(self.samples)
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for i in range(len(self.samples)):
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start = self.cur_indexes[i]
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end = start + num[i]
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self.cur_indexes[i] = end
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s = self.samples[i][start:end]
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ans.append(s)
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return ans
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class StreamList(object):
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def __init__(
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self,
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batch_size: int,
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context_size: int,
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decoding_method: str,
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):
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"""
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Args:
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batch_size:
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Size of this batch.
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context_size:
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Context size of the RNN-T decoder model.
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decoding_method:
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Decoding method. The possible values are:
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- greedy_search
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- modified_beam_search
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"""
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self.streams = [
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FeatureExtractionStream(
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context_size=context_size, decoding_method=decoding_method
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)
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for _ in range(batch_size)
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]
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@property
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def done(self) -> bool:
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"""Return True if all streams have reached end of utterance.
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That is, no more audio samples are available for all utterances.
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"""
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return all(stream.done for stream in self.streams)
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def accept_waveform(
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self,
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audio_samples: List[torch.Tensor],
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sampling_rate: float,
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):
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"""Feed audio samples to each stream.
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Args:
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audio_samples:
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A list of 1-D tensors containing the audio samples for each
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utterance in the batch. If an entry is empty, it means
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end-of-utterance has been reached.
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sampling_rate:
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Sampling rate of the given audio samples.
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"""
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assert len(audio_samples) == len(self.streams)
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for stream, samples in zip(self.streams, audio_samples):
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if stream.done:
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assert samples.numel() == 0
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continue
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stream.accept_waveform(
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sampling_rate=sampling_rate,
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waveform=samples,
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)
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if samples.numel() == 0:
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stream.input_finished()
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def build_batch(
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self,
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chunk_length: int,
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segment_length: int,
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) -> Tuple[Optional[torch.Tensor], Optional[List[FeatureExtractionStream]]]:
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"""
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Args:
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chunk_length:
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Number of frames for each chunk. It equals to
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``segment_length + right_context_length``.
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segment_length
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Number of frames for each segment.
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Returns:
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Return a tuple containing:
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- features, a 3-D tensor of shape ``(num_active_streams, T, C)``
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- active_streams, a list of active streams. We say a stream is
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active when it has enough feature frames to be fed into the
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encoder model.
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"""
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feature_list = []
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stream_list = []
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for stream in self.streams:
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if len(stream.feature_frames) >= chunk_length:
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# this_chunk is a list of tensors, each of which
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# has a shape (1, feature_dim)
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chunk = stream.feature_frames[:chunk_length]
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stream.feature_frames = stream.feature_frames[segment_length:]
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features = torch.cat(chunk, dim=0)
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feature_list.append(features)
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stream_list.append(stream)
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elif stream.done and len(stream.feature_frames) > 0:
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chunk = stream.feature_frames[:chunk_length]
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stream.feature_frames = []
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features = torch.cat(chunk, dim=0)
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features = torch.nn.functional.pad(
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features,
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(0, 0, 0, chunk_length - features.size(0)),
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mode="constant",
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value=LOG_EPSILON,
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)
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feature_list.append(features)
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stream_list.append(stream)
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if len(feature_list) == 0:
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return None, None
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features = torch.stack(feature_list, dim=0)
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return features, stream_list
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def greedy_search(
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model: nn.Module,
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streams: List[FeatureExtractionStream],
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encoder_out: torch.Tensor,
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sp: spm.SentencePieceProcessor,
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):
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"""
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Args:
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model:
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The RNN-T model.
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streams:
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A list of stream objects.
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encoder_out:
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A 3-D tensor of shape (N, T, encoder_out_dim) containing the output of
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the encoder model.
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sp:
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The BPE model.
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"""
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assert len(streams) == encoder_out.size(0)
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assert encoder_out.ndim == 3
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blank_id = model.decoder.blank_id
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context_size = model.decoder.context_size
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device = model.device
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T = encoder_out.size(1)
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if streams[0].decoder_out is None:
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for stream in streams:
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stream.hyp = [blank_id] * context_size
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decoder_input = torch.tensor(
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[stream.hyp[-context_size:] for stream in streams],
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device=device,
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dtype=torch.int64,
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)
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decoder_out = model.decoder(decoder_input, need_pad=False).squeeze(1)
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# decoder_out is of shape (N, decoder_out_dim)
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else:
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decoder_out = torch.stack(
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[stream.decoder_out for stream in streams],
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dim=0,
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)
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for t in range(T):
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current_encoder_out = encoder_out[:, t]
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# current_encoder_out's shape: (batch_size, encoder_out_dim)
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logits = model.joiner(current_encoder_out, decoder_out)
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# logits'shape (batch_size, vocab_size)
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assert logits.ndim == 2, logits.shape
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y = logits.argmax(dim=1).tolist()
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emitted = False
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for i, v in enumerate(y):
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if v != blank_id:
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streams[i].hyp.append(v)
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emitted = True
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if emitted:
|
|
# update decoder output
|
|
decoder_input = torch.tensor(
|
|
[stream.hyp[-context_size:] for stream in streams],
|
|
device=device,
|
|
dtype=torch.int64,
|
|
)
|
|
decoder_out = model.decoder(
|
|
decoder_input,
|
|
need_pad=False,
|
|
).squeeze(1)
|
|
|
|
for k, stream in enumerate(streams):
|
|
result = sp.decode(stream.decoding_result())
|
|
logging.info(f"Partial result {k}:\n{result}")
|
|
|
|
decoder_out_list = decoder_out.unbind(dim=0)
|
|
for i, d in enumerate(decoder_out_list):
|
|
streams[i].decoder_out = d
|
|
|
|
|
|
def modified_beam_search(
|
|
model: nn.Module,
|
|
streams: List[FeatureExtractionStream],
|
|
encoder_out: torch.Tensor,
|
|
sp: spm.SentencePieceProcessor,
|
|
beam: int = 4,
|
|
):
|
|
"""
|
|
Args:
|
|
model:
|
|
The RNN-T model.
|
|
streams:
|
|
A list of stream objects.
|
|
encoder_out:
|
|
A 3-D tensor of shape (N, T, encoder_out_dim) containing the output of
|
|
the encoder model.
|
|
sp:
|
|
The BPE model.
|
|
beam:
|
|
Number of active paths during the beam search.
|
|
"""
|
|
assert encoder_out.ndim == 3, encoder_out.shape
|
|
assert len(streams) == encoder_out.size(0)
|
|
|
|
blank_id = model.decoder.blank_id
|
|
context_size = model.decoder.context_size
|
|
device = model.device
|
|
batch_size = len(streams)
|
|
T = encoder_out.size(1)
|
|
|
|
for stream in streams:
|
|
if len(stream.hyps) == 0:
|
|
stream.hyps.add(
|
|
Hypothesis(
|
|
ys=[blank_id] * context_size,
|
|
log_prob=torch.zeros(1, dtype=torch.float32, device=device),
|
|
)
|
|
)
|
|
B = [stream.hyps for stream in streams]
|
|
for t in range(T):
|
|
current_encoder_out = encoder_out[:, t]
|
|
# current_encoder_out's shape: (batch_size, encoder_out_dim)
|
|
|
|
hyps_shape = get_hyps_shape(B).to(device)
|
|
|
|
A = [list(b) for b in B]
|
|
B = [HypothesisList() for _ in range(batch_size)]
|
|
|
|
ys_log_probs = torch.stack(
|
|
[hyp.log_prob.reshape(1) for hyps in A for hyp in hyps], dim=0
|
|
) # (num_hyps, 1)
|
|
|
|
decoder_input = torch.tensor(
|
|
[hyp.ys[-context_size:] for hyps in A for hyp in hyps],
|
|
device=device,
|
|
dtype=torch.int64,
|
|
) # (num_hyps, context_size)
|
|
|
|
decoder_out = model.decoder(decoder_input, need_pad=False).squeeze(1)
|
|
# decoder_out is of shape (num_hyps, decoder_output_dim)
|
|
|
|
# Note: For torch 1.7.1 and below, it requires a torch.int64 tensor
|
|
# as index, so we use `to(torch.int64)` below.
|
|
current_encoder_out = torch.index_select(
|
|
current_encoder_out,
|
|
dim=0,
|
|
index=hyps_shape.row_ids(1).to(torch.int64),
|
|
) # (num_hyps, encoder_out_dim)
|
|
|
|
logits = model.joiner(current_encoder_out, decoder_out)
|
|
# logits is of shape (num_hyps, vocab_size)
|
|
|
|
log_probs = logits.log_softmax(dim=-1) # (num_hyps, vocab_size)
|
|
|
|
log_probs.add_(ys_log_probs)
|
|
|
|
vocab_size = log_probs.size(-1)
|
|
|
|
log_probs = log_probs.reshape(-1)
|
|
|
|
row_splits = hyps_shape.row_splits(1) * vocab_size
|
|
log_probs_shape = k2.ragged.create_ragged_shape2(
|
|
row_splits=row_splits, cached_tot_size=log_probs.numel()
|
|
)
|
|
ragged_log_probs = k2.RaggedTensor(
|
|
shape=log_probs_shape, value=log_probs
|
|
)
|
|
|
|
for i in range(batch_size):
|
|
topk_log_probs, topk_indexes = ragged_log_probs[i].topk(beam)
|
|
|
|
with warnings.catch_warnings():
|
|
warnings.simplefilter("ignore")
|
|
topk_hyp_indexes = (topk_indexes // vocab_size).tolist()
|
|
topk_token_indexes = (topk_indexes % vocab_size).tolist()
|
|
|
|
for k in range(len(topk_hyp_indexes)):
|
|
hyp_idx = topk_hyp_indexes[k]
|
|
hyp = A[i][hyp_idx]
|
|
|
|
new_ys = hyp.ys[:]
|
|
new_token = topk_token_indexes[k]
|
|
if new_token != blank_id:
|
|
new_ys.append(new_token)
|
|
|
|
new_log_prob = topk_log_probs[k]
|
|
new_hyp = Hypothesis(ys=new_ys, log_prob=new_log_prob)
|
|
B[i].add(new_hyp)
|
|
|
|
streams[i].hyps = B[i]
|
|
result = sp.decode(streams[i].decoding_result())
|
|
logging.info(f"Partial result {i}:\n{result}")
|
|
|
|
|
|
def process_features(
|
|
model: nn.Module,
|
|
features: torch.Tensor,
|
|
streams: List[FeatureExtractionStream],
|
|
params: AttributeDict,
|
|
sp: spm.SentencePieceProcessor,
|
|
) -> None:
|
|
"""Process features for each stream in parallel.
|
|
|
|
Args:
|
|
model:
|
|
The RNN-T model.
|
|
features:
|
|
A 3-D tensor of shape (N, T, C).
|
|
streams:
|
|
A list of streams of size (N,).
|
|
params:
|
|
It is the return value of :func:`get_params`.
|
|
sp:
|
|
The BPE model.
|
|
"""
|
|
assert features.ndim == 3
|
|
assert features.size(0) == len(streams)
|
|
batch_size = features.size(0)
|
|
|
|
device = model.device
|
|
features = features.to(device)
|
|
feature_lens = torch.full(
|
|
(batch_size,),
|
|
fill_value=features.size(1),
|
|
device=device,
|
|
)
|
|
|
|
# Caution: It has a limitation as it assumes that
|
|
# if one of the stream has an empty state, then all other
|
|
# streams also have empty states.
|
|
if streams[0].states is None:
|
|
states = None
|
|
else:
|
|
state_list = [stream.states for stream in streams]
|
|
states = stack_states(state_list)
|
|
|
|
(encoder_out, encoder_out_lens, states,) = model.encoder.streaming_forward(
|
|
features,
|
|
feature_lens,
|
|
states,
|
|
)
|
|
state_list = unstack_states(states)
|
|
for i, s in enumerate(state_list):
|
|
streams[i].states = s
|
|
|
|
if params.decoding_method == "greedy_search":
|
|
greedy_search(
|
|
model=model,
|
|
streams=streams,
|
|
encoder_out=encoder_out,
|
|
sp=sp,
|
|
)
|
|
elif params.decoding_method == "modified_beam_search":
|
|
modified_beam_search(
|
|
model=model,
|
|
streams=streams,
|
|
encoder_out=encoder_out,
|
|
sp=sp,
|
|
beam=params.beam_size,
|
|
)
|
|
else:
|
|
raise ValueError(
|
|
f"Unsupported decoding method: {params.decoding_method}"
|
|
)
|
|
|
|
|
|
def decode_batch(
|
|
batched_samples: List[torch.Tensor],
|
|
model: nn.Module,
|
|
params: AttributeDict,
|
|
sp: spm.SentencePieceProcessor,
|
|
) -> List[str]:
|
|
"""
|
|
Args:
|
|
batched_samples:
|
|
A list of 1-D tensors containing the audio samples of each utterance.
|
|
model:
|
|
The RNN-T model.
|
|
params:
|
|
It is the return value of :func:`get_params`.
|
|
sp:
|
|
The BPE model.
|
|
"""
|
|
# number of frames before subsampling
|
|
segment_length = model.encoder.segment_length
|
|
|
|
right_context_length = model.encoder.right_context_length
|
|
|
|
# We add 3 here since the subsampling method is using
|
|
# ((len - 1) // 2 - 1) // 2)
|
|
chunk_length = (segment_length + 3) + right_context_length
|
|
|
|
batch_size = len(batched_samples)
|
|
streaming_audio_samples = StreamingAudioSamples(batched_samples)
|
|
|
|
stream_list = StreamList(
|
|
batch_size=batch_size,
|
|
context_size=params.context_size,
|
|
decoding_method=params.decoding_method,
|
|
)
|
|
|
|
while not streaming_audio_samples.done:
|
|
samples = streaming_audio_samples.get_next()
|
|
stream_list.accept_waveform(
|
|
audio_samples=samples,
|
|
sampling_rate=params.sampling_rate,
|
|
)
|
|
features, active_streams = stream_list.build_batch(
|
|
chunk_length=chunk_length,
|
|
segment_length=segment_length,
|
|
)
|
|
if features is not None:
|
|
process_features(
|
|
model=model,
|
|
features=features,
|
|
streams=active_streams,
|
|
params=params,
|
|
sp=sp,
|
|
)
|
|
results = []
|
|
for stream in stream_list.streams:
|
|
text = sp.decode(stream.decoding_result())
|
|
results.append(text)
|
|
return results
|
|
|
|
|
|
@torch.no_grad()
|
|
def main():
|
|
parser = get_parser()
|
|
LibriSpeechAsrDataModule.add_arguments(parser)
|
|
args = parser.parse_args()
|
|
args.exp_dir = Path(args.exp_dir)
|
|
|
|
params = get_params()
|
|
params.update(vars(args))
|
|
|
|
# Note: params.decoding_method is currently not used.
|
|
params.res_dir = params.exp_dir / "streaming" / params.decoding_method
|
|
|
|
setup_logger(f"{params.res_dir}/log-streaming-decode")
|
|
logging.info("Decoding started")
|
|
|
|
device = torch.device("cpu")
|
|
if torch.cuda.is_available():
|
|
device = torch.device("cuda", 0)
|
|
|
|
logging.info(f"Device: {device}")
|
|
|
|
sp = spm.SentencePieceProcessor()
|
|
sp.load(params.bpe_model)
|
|
|
|
# <blk> is defined in local/train_bpe_model.py
|
|
params.blank_id = sp.piece_to_id("<blk>")
|
|
params.vocab_size = sp.get_piece_size()
|
|
|
|
params.device = device
|
|
|
|
logging.info(params)
|
|
|
|
logging.info("About to create model")
|
|
model = get_transducer_model(params)
|
|
|
|
if params.avg_last_n > 0:
|
|
filenames = find_checkpoints(params.exp_dir)[: params.avg_last_n]
|
|
logging.info(f"averaging {filenames}")
|
|
model.to(device)
|
|
model.load_state_dict(average_checkpoints(filenames, device=device))
|
|
elif params.avg == 1:
|
|
load_checkpoint(f"{params.exp_dir}/epoch-{params.epoch}.pt", model)
|
|
else:
|
|
start = params.epoch - params.avg + 1
|
|
filenames = []
|
|
for i in range(start, params.epoch + 1):
|
|
if start >= 0:
|
|
filenames.append(f"{params.exp_dir}/epoch-{i}.pt")
|
|
logging.info(f"averaging {filenames}")
|
|
model.to(device)
|
|
model.load_state_dict(average_checkpoints(filenames, device=device))
|
|
|
|
model.to(device)
|
|
model.eval()
|
|
model.device = device
|
|
|
|
num_param = sum([p.numel() for p in model.parameters()])
|
|
logging.info(f"Number of model parameters: {num_param}")
|
|
|
|
librispeech = LibriSpeechAsrDataModule(args)
|
|
|
|
test_clean_cuts = librispeech.test_clean_cuts()
|
|
|
|
batch_size = 3
|
|
|
|
ground_truth = []
|
|
batched_samples = []
|
|
for num, cut in enumerate(test_clean_cuts):
|
|
audio: np.ndarray = cut.load_audio()
|
|
# audio.shape: (1, num_samples)
|
|
assert len(audio.shape) == 2
|
|
assert audio.shape[0] == 1, "Should be single channel"
|
|
assert audio.dtype == np.float32, audio.dtype
|
|
|
|
# The trained model is using normalized samples
|
|
assert audio.max() <= 1, "Should be normalized to [-1, 1])"
|
|
|
|
samples = torch.from_numpy(audio).squeeze(0)
|
|
|
|
batched_samples.append(samples)
|
|
ground_truth.append(cut.supervisions[0].text)
|
|
|
|
if len(batched_samples) >= batch_size:
|
|
decoded_results = decode_batch(
|
|
batched_samples=batched_samples,
|
|
model=model,
|
|
params=params,
|
|
sp=sp,
|
|
)
|
|
s = "\n"
|
|
for i, (hyp, ref) in enumerate(zip(decoded_results, ground_truth)):
|
|
s += f"hyp {i}:\n{hyp}\n"
|
|
s += f"ref {i}:\n{ref}\n\n"
|
|
logging.info(s)
|
|
batched_samples = []
|
|
ground_truth = []
|
|
# break after processing the first batch for test purposes
|
|
break
|
|
|
|
|
|
if __name__ == "__main__":
|
|
torch.manual_seed(20220410)
|
|
main()
|