mirror of
https://github.com/k2-fsa/icefall.git
synced 2025-08-09 18:12:19 +00:00
* 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) * Update results for torchaudio RNN-T. (#322) * Fix some typos. (#329) * fix fp16 option in example usage (#332) * Support averaging models with weight tying. (#333) * Support specifying iteration number of checkpoints for decoding. (#336) See also #289 * Modified conformer with multi datasets (#312) * Copy files for editing. * Use librispeech + gigaspeech with modified conformer. * Support specifying number of workers for on-the-fly feature extraction. * Feature extraction code for GigaSpeech. * Combine XL splits lazily during training. * Fix warnings in decoding. * Add decoding code for GigaSpeech. * Fix decoding the gigaspeech dataset. We have to use the decoder/joiner networks for the GigaSpeech dataset. * Disable speed perturbe for XL subset. * Compute the Nbest oracle WER for RNN-T decoding. * Minor fixes. * Minor fixes. * Add results. * Update results. * Update CI. * Update results. * Fix style issues. * Update results. * Fix style issues. * Update results. (#340) * Update results. * Typo fixes. * Validate generated manifest files. (#338) * Validate generated manifest files. (#338) * Save batch to disk on OOM. (#343) * Save batch to disk on OOM. * minor fixes * Fixes after review. * Fix style issues. * Fix decoding for gigaspeech in the libri + giga setup. (#345) * Model average (#344) * First upload of model average codes. * minor fix * update decode file * update .flake8 * rename pruned_transducer_stateless3 to pruned_transducer_stateless4 * change epoch number counter starting from 1 instead of 0 * minor fix of pruned_transducer_stateless4/train.py * refactor the checkpoint.py * minor fix, update docs, and modify the epoch number to count from 1 in the pruned_transducer_stateless4/decode.py * update author info * add docs of the scaling in function average_checkpoints_with_averaged_model * Save batch to disk on exception. (#350) * Bug fix (#352) * Keep model_avg on cpu (#348) * keep model_avg on cpu * explicitly convert model_avg to cpu * minor fix * remove device convertion for model_avg * modify usage of the model device in train.py * change model.device to next(model.parameters()).device for decoding * assert params.start_epoch>0 * assert params.start_epoch>0, params.start_epoch * Do some changes for aishell/ASR/transducer stateless/export.py (#347) * do some changes for aishell/ASR/transducer_stateless/export.py * Support decoding with averaged model when using --iter (#353) * support decoding with averaged model when using --iter * minor fix * monir fix of copyright date * Stringify torch.__version__ before serializing it. (#354) * Run decode.py in GitHub actions. (#356) * Ignore padding frames during RNN-T decoding. (#358) * Ignore padding frames during RNN-T decoding. * Fix outdated decoding code. * Minor fixes. * Support --iter in export.py (#360) * GigaSpeech RNN-T experiments (#318) * Copy RNN-T recipe from librispeech * flake8 * flake8 * Update params * gigaspeech decode * black * Update results * syntax highlight * Update RESULTS.md * typo * Update decoding script for gigaspeech and remove duplicate files. (#361) * Validate that there are no OOV tokens in BPE-based lexicons. (#359) * Validate that there are no OOV tokens in BPE-based lexicons. * Typo fixes. * Decode gigaspeech in GitHub actions (#362) * Add CI for gigaspeech. * Update results for libri+giga multi dataset setup. (#363) * Update results for libri+giga multi dataset setup. * Update GigaSpeech reults (#364) * Update decode.py * Update export.py * Update results * Update README.md * Fix GitHub CI for decoding GigaSpeech dev/test datasets (#366) * modify .flake8 * minor fix * minor fix Co-authored-by: Daniel Povey <dpovey@gmail.com> Co-authored-by: Wei Kang <wkang@pku.org.cn> 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> Co-authored-by: whsqkaak <whsqkaak@naver.com> Co-authored-by: pehonnet <pe.honnet@gmail.com>
362 lines
10 KiB
Python
362 lines
10 KiB
Python
#!/usr/bin/env python3
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# Copyright 2021 Xiaomi Corp. (authors: Fangjun Kuang)
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# 2022 Xiaomi Crop. (authors: Mingshuang Luo)
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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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"""
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Usage:
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(1) greedy search
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./pruned_transducer_stateless/pretrained.py \
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--checkpoint ./pruned_transducer_stateless/exp/pretrained.pt \
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--bpe-model ./data/lang_bpe_500/bpe.model \
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--method greedy_search \
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--max-sym-per-frame 1 \
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/path/to/foo.wav \
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/path/to/bar.wav
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(2) beam search
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./pruned_transducer_stateless/pretrained.py \
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--checkpoint ./pruned_transducer_stateless/exp/pretrained.pt \
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--bpe-model ./data/lang_bpe_500/bpe.model \
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--method beam_search \
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--beam-size 4 \
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/path/to/foo.wav \
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/path/to/bar.wav
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(3) modified beam search
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./pruned_transducer_stateless/pretrained.py \
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--checkpoint ./pruned_transducer_stateless/exp/pretrained.pt \
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--bpe-model ./data/lang_bpe_500/bpe.model \
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--method modified_beam_search \
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--beam-size 4 \
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/path/to/foo.wav \
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/path/to/bar.wav
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(4) fast beam search
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./pruned_transducer_stateless/pretrained.py \
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--checkpoint ./pruned_transducer_stateless/exp/pretrained.pt \
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--bpe-model ./data/lang_bpe_500/bpe.model \
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--method fast_beam_search \
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--beam 4 \
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--max-contexts 4 \
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--max-states 8 \
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/path/to/foo.wav \
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/path/to/bar.wav
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You can also use `./pruned_transducer_stateless/exp/epoch-xx.pt`.
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Note: ./pruned_transducer_stateless/exp/pretrained.pt is generated by
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./pruned_transducer_stateless/export.py
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"""
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import argparse
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import logging
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import math
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from typing import List
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import k2
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import kaldifeat
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import sentencepiece as spm
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import torch
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import torchaudio
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from beam_search import (
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beam_search,
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fast_beam_search_one_best,
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greedy_search,
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greedy_search_batch,
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modified_beam_search,
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)
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from torch.nn.utils.rnn import pad_sequence
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from train import get_params, get_transducer_model
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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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"--checkpoint",
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type=str,
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required=True,
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help="Path to the checkpoint. "
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"The checkpoint is assumed to be saved by "
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"icefall.checkpoint.save_checkpoint().",
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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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help="""Path to bpe.model.
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Used only when method is ctc-decoding.
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""",
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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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"sound_files",
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type=str,
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nargs="+",
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help="The input sound file(s) to transcribe. "
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"Supported formats are those supported by torchaudio.load(). "
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"For example, wav and flac are supported. "
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"The sample rate has to be 16kHz.",
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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="Used only when --method is beam_search and 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. Used only when
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--method is greedy_search.
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""",
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)
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parser.add_argument(
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"--sample-rate",
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type=int,
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default=16000,
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help="The sample rate of the input sound file",
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)
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return parser
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def read_sound_files(
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filenames: List[str], expected_sample_rate: float
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) -> List[torch.Tensor]:
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"""Read a list of sound files into a list 1-D float32 torch tensors.
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Args:
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filenames:
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A list of sound filenames.
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expected_sample_rate:
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The expected sample rate of the sound files.
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Returns:
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Return a list of 1-D float32 torch tensors.
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"""
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ans = []
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for f in filenames:
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wave, sample_rate = torchaudio.load(f)
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assert sample_rate == expected_sample_rate, (
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f"expected sample rate: {expected_sample_rate}. "
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f"Given: {sample_rate}"
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)
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# We use only the first channel
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ans.append(wave[0])
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return ans
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@torch.no_grad()
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def main():
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parser = get_parser()
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args = parser.parse_args()
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params = get_params()
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params.update(vars(args))
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sp = spm.SentencePieceProcessor()
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sp.load(params.bpe_model)
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# <blk> and <unk> are defined in local/train_bpe_model.py
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params.blank_id = sp.piece_to_id("<blk>")
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params.unk_id = sp.piece_to_id("<unk>")
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params.vocab_size = sp.get_piece_size()
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logging.info(f"{params}")
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device = torch.device("cpu")
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if torch.cuda.is_available():
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device = torch.device("cuda", 0)
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logging.info(f"device: {device}")
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logging.info("Creating model")
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model = get_transducer_model(params)
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checkpoint = torch.load(args.checkpoint, map_location="cpu")
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model.load_state_dict(checkpoint["model"], strict=False)
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model.to(device)
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model.eval()
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model.device = device
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if params.decoding_method == "fast_beam_search":
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decoding_graph = k2.trivial_graph(params.vocab_size - 1, device=device)
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else:
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decoding_graph = None
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logging.info("Constructing Fbank computer")
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opts = kaldifeat.FbankOptions()
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opts.device = device
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opts.frame_opts.dither = 0
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opts.frame_opts.snip_edges = False
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opts.frame_opts.samp_freq = params.sample_rate
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opts.mel_opts.num_bins = params.feature_dim
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fbank = kaldifeat.Fbank(opts)
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logging.info(f"Reading sound files: {params.sound_files}")
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waves = read_sound_files(
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filenames=params.sound_files, expected_sample_rate=params.sample_rate
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)
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waves = [w.to(device) for w in waves]
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logging.info("Decoding started")
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features = fbank(waves)
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feature_lengths = [f.size(0) for f in features]
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features = pad_sequence(
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features, batch_first=True, padding_value=math.log(1e-10)
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)
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feature_lengths = torch.tensor(feature_lengths, device=device)
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with torch.no_grad():
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encoder_out, encoder_out_lens = model.encoder(
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x=features, x_lens=feature_lengths
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)
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hyps = []
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msg = f"Using {params.decoding_method}"
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logging.info(msg)
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if params.decoding_method == "fast_beam_search":
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hyp_tokens = fast_beam_search_one_best(
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model=model,
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decoding_graph=decoding_graph,
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encoder_out=encoder_out,
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encoder_out_lens=encoder_out_lens,
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beam=params.beam,
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max_contexts=params.max_contexts,
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max_states=params.max_states,
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)
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for hyp in sp.decode(hyp_tokens):
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hyps.append(hyp.split())
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elif (
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params.decoding_method == "greedy_search"
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and params.max_sym_per_frame == 1
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):
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hyp_tokens = greedy_search_batch(
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model=model,
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encoder_out=encoder_out,
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encoder_out_lens=encoder_out_lens,
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)
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for hyp in sp.decode(hyp_tokens):
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hyps.append(hyp.split())
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elif params.decoding_method == "modified_beam_search":
|
|
hyp_tokens = modified_beam_search(
|
|
model=model,
|
|
encoder_out=encoder_out,
|
|
encoder_out_lens=encoder_out_lens,
|
|
beam=params.beam_size,
|
|
)
|
|
for hyp in sp.decode(hyp_tokens):
|
|
hyps.append(hyp.split())
|
|
else:
|
|
batch_size = encoder_out.size(0)
|
|
|
|
for i in range(batch_size):
|
|
# fmt: off
|
|
encoder_out_i = encoder_out[i:i+1, :encoder_out_lens[i]]
|
|
# fmt: on
|
|
if params.decoding_method == "greedy_search":
|
|
hyp = greedy_search(
|
|
model=model,
|
|
encoder_out=encoder_out_i,
|
|
max_sym_per_frame=params.max_sym_per_frame,
|
|
)
|
|
elif params.decoding_method == "beam_search":
|
|
hyp = beam_search(
|
|
model=model,
|
|
encoder_out=encoder_out_i,
|
|
beam=params.beam_size,
|
|
)
|
|
else:
|
|
raise ValueError(
|
|
f"Unsupported decoding method: {params.decoding_method}"
|
|
)
|
|
hyps.append(sp.decode(hyp).split())
|
|
|
|
s = "\n"
|
|
for filename, hyp in zip(params.sound_files, hyps):
|
|
words = " ".join(hyp)
|
|
s += f"{filename}:\n{words}\n\n"
|
|
logging.info(s)
|
|
|
|
logging.info("Decoding Done")
|
|
|
|
|
|
if __name__ == "__main__":
|
|
formatter = (
|
|
"%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
|
|
)
|
|
|
|
logging.basicConfig(format=formatter, level=logging.INFO)
|
|
main()
|