From 0e57b30495fd802b7794b5c4430e1dd927c9a4a8 Mon Sep 17 00:00:00 2001 From: Mingshuang Luo <37799481+luomingshuang@users.noreply.github.com> Date: Mon, 23 May 2022 17:13:01 +0800 Subject: [PATCH] [Ready to merge] Pruned Transducer Stateless2 for WenetSpeech (char-based) (#349) * add char-based pruned-rnnt2 for wenetspeech * style check * style check * change for export.py * do some changes * do some changes * a small change for .flake8 * solve the conflicts --- README.md | 33 + egs/wenetspeech/ASR/README.md | 19 + egs/wenetspeech/ASR/RESULTS.md | 93 ++ .../ASR/local/compute_fbank_musan.py | 1 + .../compute_fbank_wenetspeech_dev_test.py | 93 ++ .../local/compute_fbank_wenetspeech_splits.py | 181 +++ .../ASR/local/display_manifest_statistics.py | 132 +++ egs/wenetspeech/ASR/local/prepare_char.py | 246 ++++ egs/wenetspeech/ASR/local/prepare_lang.py | 1 + egs/wenetspeech/ASR/local/prepare_words.py | 84 ++ .../ASR/local/preprocess_wenetspeech.py | 120 ++ egs/wenetspeech/ASR/local/text2segments.py | 83 ++ egs/wenetspeech/ASR/local/text2token.py | 196 ++++ egs/wenetspeech/ASR/prepare.sh | 225 ++++ .../pruned_transducer_stateless2/__init__.py | 0 .../asr_datamodule.py | 450 ++++++++ .../beam_search.py | 1 + .../pruned_transducer_stateless2/conformer.py | 1 + .../pruned_transducer_stateless2/decode.py | 623 ++++++++++ .../pruned_transducer_stateless2/decoder.py | 1 + .../encoder_interface.py | 1 + .../pruned_transducer_stateless2/export.py | 178 +++ .../pruned_transducer_stateless2/joiner.py | 1 + .../ASR/pruned_transducer_stateless2/model.py | 1 + .../ASR/pruned_transducer_stateless2/optim.py | 1 + .../pretrained.py | 342 ++++++ .../pruned_transducer_stateless2/scaling.py | 1 + .../ASR/pruned_transducer_stateless2/train.py | 1025 +++++++++++++++++ egs/wenetspeech/ASR/shared | 1 + 29 files changed, 4134 insertions(+) create mode 100644 egs/wenetspeech/ASR/README.md create mode 100644 egs/wenetspeech/ASR/RESULTS.md create mode 120000 egs/wenetspeech/ASR/local/compute_fbank_musan.py create mode 100755 egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_dev_test.py create mode 100755 egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_splits.py create mode 100644 egs/wenetspeech/ASR/local/display_manifest_statistics.py create mode 100755 egs/wenetspeech/ASR/local/prepare_char.py create mode 120000 egs/wenetspeech/ASR/local/prepare_lang.py create mode 100644 egs/wenetspeech/ASR/local/prepare_words.py create mode 100755 egs/wenetspeech/ASR/local/preprocess_wenetspeech.py create mode 100644 egs/wenetspeech/ASR/local/text2segments.py create mode 100755 egs/wenetspeech/ASR/local/text2token.py create mode 100755 egs/wenetspeech/ASR/prepare.sh create mode 100644 egs/wenetspeech/ASR/pruned_transducer_stateless2/__init__.py create mode 100644 egs/wenetspeech/ASR/pruned_transducer_stateless2/asr_datamodule.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/beam_search.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/conformer.py create mode 100755 egs/wenetspeech/ASR/pruned_transducer_stateless2/decode.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/decoder.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/encoder_interface.py create mode 100644 egs/wenetspeech/ASR/pruned_transducer_stateless2/export.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/joiner.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/model.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/optim.py create mode 100644 egs/wenetspeech/ASR/pruned_transducer_stateless2/pretrained.py create mode 120000 egs/wenetspeech/ASR/pruned_transducer_stateless2/scaling.py create mode 100644 egs/wenetspeech/ASR/pruned_transducer_stateless2/train.py create mode 120000 egs/wenetspeech/ASR/shared diff --git a/README.md b/README.md index 8911a43361..d88ed7aac4 100644 --- a/README.md +++ b/README.md @@ -20,6 +20,8 @@ We provide 6 recipes at present: - [TIMIT][timit] - [TED-LIUM3][tedlium3] - [GigaSpeech][gigaspeech] + - [Aidatatang_200zh][aidatatang_200zh] + - [WenetSpeech][wenetspeech] ### yesno @@ -217,6 +219,33 @@ and [Pruned stateless RNN-T: Conformer encoder + Embedding decoder + k2 pruned R | fast beam search | 10.50 | 10.69 | | modified beam search | 10.40 | 10.51 | +### Aidatatang_200zh + +We provide one model for this recipe: [Pruned stateless RNN-T: Conformer encoder + Embedding decoder + k2 pruned RNN-T loss][Aidatatang_200zh_pruned_transducer_stateless2]. + +#### Pruned stateless RNN-T: Conformer encoder + Embedding decoder + k2 pruned RNN-T loss + +| | Dev | Test | +|----------------------|-------|-------| +| greedy search | 5.53 | 6.59 | +| fast beam search | 5.30 | 6.34 | +| modified beam search | 5.27 | 6.33 | + +We provide a Colab notebook to run a pre-trained Pruned Transducer Stateless model: [![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/drive/1wNSnSj3T5oOctbh5IGCa393gKOoQw2GH?usp=sharing) + +### WenetSpeech + +We provide one model for this recipe: [Pruned stateless RNN-T: Conformer encoder + Embedding decoder + k2 pruned RNN-T loss][WenetSpeech_pruned_transducer_stateless2]. + +#### Pruned stateless RNN-T: Conformer encoder + Embedding decoder + k2 pruned RNN-T loss (trained with L subset) + +| | Dev | Test-Net | Test-Meeting | +|----------------------|-------|----------|--------------| +| greedy search | 7.80 | 8.75 | 13.49 | +| fast beam search | 7.94 | 8.74 | 13.80 | +| modified beam search | 7.76 | 8.71 | 13.41 | + +We provide a Colab notebook to run a pre-trained Pruned Transducer Stateless model: [![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/drive/1EV4e1CHa1GZgEF-bZgizqI9RyFFehIiN?usp=sharing) ## Deployment with C++ @@ -243,10 +272,14 @@ Please see: [![Open In Colab](https://colab.research.google.com/assets/colab-bad [TED-LIUM3_pruned_transducer_stateless]: egs/tedlium3/ASR/pruned_transducer_stateless [GigaSpeech_conformer_ctc]: egs/gigaspeech/ASR/conformer_ctc [GigaSpeech_pruned_transducer_stateless2]: egs/gigaspeech/ASR/pruned_transducer_stateless2 +[Aidatatang_200zh_pruned_transducer_stateless2]: egs/aidatatang_200zh/ASR/pruned_transducer_stateless2 +[WenetSpeech_pruned_transducer_stateless2]: egs/wenetspeech/ASR/pruned_transducer_stateless2 [yesno]: egs/yesno/ASR [librispeech]: egs/librispeech/ASR [aishell]: egs/aishell/ASR [timit]: egs/timit/ASR [tedlium3]: egs/tedlium3/ASR [gigaspeech]: egs/gigaspeech/ASR +[aidatatang_200zh]: egs/aidatatang_200zh/ASR +[wenetspeech]: egs/wenetspeech/ASR [k2]: https://github.com/k2-fsa/k2 diff --git a/egs/wenetspeech/ASR/README.md b/egs/wenetspeech/ASR/README.md new file mode 100644 index 0000000000..c92f1b4e67 --- /dev/null +++ b/egs/wenetspeech/ASR/README.md @@ -0,0 +1,19 @@ + +# Introduction + +This recipe includes some different ASR models trained with WenetSpeech. + +[./RESULTS.md](./RESULTS.md) contains the latest results. + +# Transducers + +There are various folders containing the name `transducer` in this folder. +The following table lists the differences among them. + +| | Encoder | Decoder | Comment | +|---------------------------------------|---------------------|--------------------|-----------------------------| +| `pruned_transducer_stateless2` | Conformer(modified) | Embedding + Conv1d | Using k2 pruned RNN-T loss | | + +The decoder in `transducer_stateless` is modified from the paper +[Rnn-Transducer with Stateless Prediction Network](https://ieeexplore.ieee.org/document/9054419/). +We place an additional Conv1d layer right after the input embedding layer. diff --git a/egs/wenetspeech/ASR/RESULTS.md b/egs/wenetspeech/ASR/RESULTS.md new file mode 100644 index 0000000000..ea6658ddbc --- /dev/null +++ b/egs/wenetspeech/ASR/RESULTS.md @@ -0,0 +1,93 @@ +## Results + +### WenetSpeech char-based training results (Pruned Transducer 2) + +#### 2022-05-19 + +Using the codes from this PR https://github.com/k2-fsa/icefall/pull/349. + +When training with the L subset, the WERs are + +| | dev | test-net | test-meeting | comment | +|------------------------------------|-------|----------|--------------|------------------------------------------| +| greedy search | 7.80 | 8.75 | 13.49 | --epoch 10, --avg 2, --max-duration 100 | +| modified beam search (beam size 4) | 7.76 | 8.71 | 13.41 | --epoch 10, --avg 2, --max-duration 100 | +| fast beam search (set as default) | 7.94 | 8.74 | 13.80 | --epoch 10, --avg 2, --max-duration 1500 | + +The training command for reproducing is given below: + +``` +export CUDA_VISIBLE_DEVICES="0,1,2,3,4,5,6,7" + +./pruned_transducer_stateless2/train.py \ + --lang-dir data/lang_char \ + --exp-dir pruned_transducer_stateless2/exp \ + --world-size 8 \ + --num-epochs 15 \ + --start-epoch 0 \ + --max-duration 180 \ + --valid-interval 3000 \ + --model-warm-step 3000 \ + --save-every-n 8000 \ + --training-subset L +``` + +The tensorboard training log can be found at +https://tensorboard.dev/experiment/wM4ZUNtASRavJx79EOYYcg/#scalars + +The decoding command is: +``` +epoch=10 +avg=2 + +## greedy search +./pruned_transducer_stateless2/decode.py \ + --epoch $epoch \ + --avg $avg \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --lang-dir data/lang_char \ + --max-duration 100 \ + --decoding-method greedy_search + +## modified beam search +./pruned_transducer_stateless2/decode.py \ + --epoch $epoch \ + --avg $avg \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --lang-dir data/lang_char \ + --max-duration 100 \ + --decoding-method modified_beam_search \ + --beam-size 4 + +## fast beam search +./pruned_transducer_stateless2/decode.py \ + --epoch $epoch \ + --avg $avg \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --lang-dir data/lang_char \ + --max-duration 1500 \ + --decoding-method fast_beam_search \ + --beam 4 \ + --max-contexts 4 \ + --max-states 8 +``` + +When training with the M subset, the WERs are + +| | dev | test-net | test-meeting | comment | +|------------------------------------|--------|-----------|---------------|-------------------------------------------| +| greedy search | 10.40 | 11.31 | 19.64 | --epoch 29, --avg 11, --max-duration 100 | +| modified beam search (beam size 4) | 9.85 | 11.04 | 18.20 | --epoch 29, --avg 11, --max-duration 100 | +| fast beam search (set as default) | 10.18 | 11.10 | 19.32 | --epoch 29, --avg 11, --max-duration 1500 | + + +When training with the S subset, the WERs are + +| | dev | test-net | test-meeting | comment | +|------------------------------------|--------|-----------|---------------|-------------------------------------------| +| greedy search | 19.92 | 25.20 | 35.35 | --epoch 29, --avg 24, --max-duration 100 | +| modified beam search (beam size 4) | 18.62 | 23.88 | 33.80 | --epoch 29, --avg 24, --max-duration 100 | +| fast beam search (set as default) | 19.31 | 24.41 | 34.87 | --epoch 29, --avg 24, --max-duration 1500 | + + +A pre-trained model and decoding logs can be found at diff --git a/egs/wenetspeech/ASR/local/compute_fbank_musan.py b/egs/wenetspeech/ASR/local/compute_fbank_musan.py new file mode 120000 index 0000000000..5833f2484e --- /dev/null +++ b/egs/wenetspeech/ASR/local/compute_fbank_musan.py @@ -0,0 +1 @@ +../../../librispeech/ASR/local/compute_fbank_musan.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_dev_test.py b/egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_dev_test.py new file mode 100755 index 0000000000..11aaef5c5a --- /dev/null +++ b/egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_dev_test.py @@ -0,0 +1,93 @@ +#!/usr/bin/env python3 +# Copyright 2021 Johns Hopkins University (Piotr Żelasko) +# Copyright 2021 Xiaomi Corp. (Fangjun Kuang) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + +import logging +from pathlib import Path + +import torch +from lhotse import ( + CutSet, + KaldifeatFbank, + KaldifeatFbankConfig, + LilcomHdf5Writer, +) + +# Torch's multithreaded behavior needs to be disabled or +# it wastes a lot of CPU and slow things down. +# Do this outside of main() in case it needs to take effect +# even when we are not invoking the main (e.g. when spawning subprocesses). +torch.set_num_threads(1) +torch.set_num_interop_threads(1) + + +def compute_fbank_wenetspeech_dev_test(): + in_out_dir = Path("data/fbank") + # number of workers in dataloader + num_workers = 42 + + # number of seconds in a batch + batch_duration = 600 + + subsets = ("S", "M", "DEV", "TEST_NET", "TEST_MEETING") + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", 0) + extractor = KaldifeatFbank(KaldifeatFbankConfig(device=device)) + + logging.info(f"device: {device}") + + for partition in subsets: + cuts_path = in_out_dir / f"cuts_{partition}.jsonl.gz" + if cuts_path.is_file(): + logging.info(f"{cuts_path} exists - skipping") + continue + + raw_cuts_path = in_out_dir / f"cuts_{partition}_raw.jsonl.gz" + + logging.info(f"Loading {raw_cuts_path}") + cut_set = CutSet.from_file(raw_cuts_path) + + logging.info("Computing features") + + cut_set = cut_set.compute_and_store_features_batch( + extractor=extractor, + storage_path=f"{in_out_dir}/feats_{partition}", + num_workers=num_workers, + batch_duration=batch_duration, + storage_type=LilcomHdf5Writer, + ) + cut_set = cut_set.trim_to_supervisions( + keep_overlapping=False, min_duration=None + ) + + logging.info(f"Saving to {cuts_path}") + cut_set.to_file(cuts_path) + + +def main(): + formatter = ( + "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s" + ) + logging.basicConfig(format=formatter, level=logging.INFO) + + compute_fbank_wenetspeech_dev_test() + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_splits.py b/egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_splits.py new file mode 100755 index 0000000000..a828bead93 --- /dev/null +++ b/egs/wenetspeech/ASR/local/compute_fbank_wenetspeech_splits.py @@ -0,0 +1,181 @@ +#!/usr/bin/env python3 +# Copyright 2021 Johns Hopkins University (Piotr Żelasko) +# Copyright 2021 Xiaomi Corp. (Fangjun Kuang) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + +import argparse +import logging +from datetime import datetime +from pathlib import Path + +import torch +from lhotse import ( + ChunkedLilcomHdf5Writer, + CutSet, + KaldifeatFbank, + KaldifeatFbankConfig, + set_audio_duration_mismatch_tolerance, + set_caching_enabled, +) + +# Torch's multithreaded behavior needs to be disabled or +# it wastes a lot of CPU and slow things down. +# Do this outside of main() in case it needs to take effect +# even when we are not invoking the main (e.g. when spawning subprocesses). +torch.set_num_threads(1) +torch.set_num_interop_threads(1) + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--training-subset", + type=str, + default="L", + help="The training subset for computing fbank feature.", + ) + + parser.add_argument( + "--num-workers", + type=int, + default=20, + help="Number of dataloading workers used for reading the audio.", + ) + + parser.add_argument( + "--batch-duration", + type=float, + default=600.0, + help="The maximum number of audio seconds in a batch." + "Determines batch size dynamically.", + ) + + parser.add_argument( + "--num-splits", + type=int, + required=True, + help="The number of splits of the L subset", + ) + + parser.add_argument( + "--start", + type=int, + default=0, + help="Process pieces starting from this number (inclusive).", + ) + + parser.add_argument( + "--stop", + type=int, + default=-1, + help="Stop processing pieces until this number (exclusive).", + ) + return parser + + +def compute_fbank_wenetspeech_splits(args): + subset = args.training_subset + subset = str(subset) + num_splits = args.num_splits + output_dir = f"data/fbank/{subset}_split_{num_splits}" + output_dir = Path(output_dir) + assert output_dir.exists(), f"{output_dir} does not exist!" + + num_digits = len(str(num_splits)) + + start = args.start + stop = args.stop + if stop < start: + stop = num_splits + + stop = min(stop, num_splits) + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", 0) + extractor = KaldifeatFbank(KaldifeatFbankConfig(device=device)) + logging.info(f"device: {device}") + + set_audio_duration_mismatch_tolerance(0.01) # 10ms tolerance + set_caching_enabled(False) + for i in range(start, stop): + idx = f"{i + 1}".zfill(num_digits) + logging.info(f"Processing {idx}/{num_splits}") + + cuts_path = output_dir / f"cuts_{subset}.{idx}.jsonl.gz" + if cuts_path.is_file(): + logging.info(f"{cuts_path} exists - skipping") + continue + + raw_cuts_path = output_dir / f"cuts_{subset}_raw.{idx}.jsonl.gz" + + logging.info(f"Loading {raw_cuts_path}") + cut_set = CutSet.from_file(raw_cuts_path) + + logging.info("Computing features") + + cut_set = cut_set.compute_and_store_features_batch( + extractor=extractor, + storage_path=f"{output_dir}/feats_{subset}_{idx}", + num_workers=args.num_workers, + batch_duration=args.batch_duration, + storage_type=ChunkedLilcomHdf5Writer, + ) + + logging.info("About to split cuts into smaller chunks.") + cut_set = cut_set.trim_to_supervisions( + keep_overlapping=False, min_duration=None + ) + + logging.info(f"Saving to {cuts_path}") + cut_set.to_file(cuts_path) + logging.info(f"Saved to {cuts_path}") + + +def main(): + now = datetime.now() + date_time = now.strftime("%Y-%m-%d-%H-%M-%S") + + log_filename = "log-compute_fbank_wenetspeech_splits" + formatter = ( + "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s" + ) + log_filename = f"{log_filename}-{date_time}" + + logging.basicConfig( + filename=log_filename, + format=formatter, + level=logging.INFO, + filemode="w", + ) + + console = logging.StreamHandler() + console.setLevel(logging.INFO) + console.setFormatter(logging.Formatter(formatter)) + logging.getLogger("").addHandler(console) + + parser = get_parser() + args = parser.parse_args() + logging.info(vars(args)) + + compute_fbank_wenetspeech_splits(args) + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/local/display_manifest_statistics.py b/egs/wenetspeech/ASR/local/display_manifest_statistics.py new file mode 100644 index 0000000000..30dc5a5ecc --- /dev/null +++ b/egs/wenetspeech/ASR/local/display_manifest_statistics.py @@ -0,0 +1,132 @@ +#!/usr/bin/env python3 +# Copyright 2021 Xiaomi Corp. (authors: Fangjun Kuang +# Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + +""" +This file displays duration statistics of utterances in a manifest. +You can use the displayed value to choose minimum/maximum duration +to remove short and long utterances during the training. +See the function `remove_short_and_long_utt()` +in ../../../librispeech/ASR/transducer/train.py +for usage. +""" + + +from lhotse import load_manifest + + +def main(): + paths = [ + "./data/fbank/cuts_S.jsonl.gz", + "./data/fbank/cuts_M.jsonl.gz", + "./data/fbank/cuts_DEV.jsonl.gz", + "./data/fbank/cuts_TEST_NET.jsonl.gz", + "./data/fbank/cuts_TEST_MEETING.jsonl.gz", + ] + + for path in paths: + print(f"Starting display the statistics for {path}") + cuts = load_manifest(path) + cuts.describe() + + +if __name__ == "__main__": + main() + +""" +Starting display the statistics for ./data/fbank/cuts_S.jsonl.gz +Duration statistics (seconds): +mean 2.4 +std 1.8 +min 0.2 +25% 1.4 +50% 2.0 +75% 2.9 +99% 8.0 +99.5% 8.7 +99.9% 11.9 +max 405.1 + +Starting display the statistics for ./data/fbank/cuts_M.jsonl.gz +Cuts count: 4543341 +Total duration (hours): 3021.1 +Speech duration (hours): 3021.1 (100.0%) +*** +Duration statistics (seconds): +mean 2.4 +std 1.6 +min 0.2 +25% 1.4 +50% 2.0 +75% 2.9 +99% 8.0 +99.5% 8.8 +99.9% 12.1 +max 405.1 + +Starting display the statistics for ./data/fbank/cuts_DEV.jsonl.gz +Cuts count: 13825 +Total duration (hours): 20.0 +Speech duration (hours): 20.0 (100.0%) +*** +Duration statistics (seconds): +mean 5.2 +std 2.2 +min 1.0 +25% 3.3 +50% 4.9 +75% 7.0 +99% 9.6 +99.5% 9.8 +99.9% 10.0 +max 10.0 + +Starting display the statistics for ./data/fbank/cuts_TEST_NET.jsonl.gz +Cuts count: 24774 +Total duration (hours): 23.1 +Speech duration (hours): 23.1 (100.0%) +*** +Duration statistics (seconds): +mean 3.4 +std 2.6 +min 0.1 +25% 1.4 +50% 2.4 +75% 4.8 +99% 13.1 +99.5% 14.5 +99.9% 18.5 +max 33.3 + +Starting display the statistics for ./data/fbank/cuts_TEST_MEETING.jsonl.gz +Cuts count: 8370 +Total duration (hours): 15.2 +Speech duration (hours): 15.2 (100.0%) +*** +Duration statistics (seconds): +mean 6.5 +std 3.5 +min 0.8 +25% 3.7 +50% 5.8 +75% 8.8 +99% 15.2 +99.5% 16.0 +99.9% 18.8 +max 24.6 + +""" diff --git a/egs/wenetspeech/ASR/local/prepare_char.py b/egs/wenetspeech/ASR/local/prepare_char.py new file mode 100755 index 0000000000..8bc073c75f --- /dev/null +++ b/egs/wenetspeech/ASR/local/prepare_char.py @@ -0,0 +1,246 @@ +#!/usr/bin/env python3 +# Copyright 2021 Xiaomi Corp. (authors: Fangjun Kuang, +# Wei Kang, +# Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + + +""" +This script takes as input `lang_dir`, which should contain:: + - lang_dir/text, + - lang_dir/words.txt +and generates the following files in the directory `lang_dir`: + - lexicon.txt + - lexicon_disambig.txt + - L.pt + - L_disambig.pt + - tokens.txt +""" + +import argparse +import re +from pathlib import Path +from typing import Dict, List + +import k2 +import torch +from prepare_lang import ( + Lexicon, + add_disambig_symbols, + add_self_loops, + write_lexicon, + write_mapping, +) + + +def lexicon_to_fst_no_sil( + lexicon: Lexicon, + token2id: Dict[str, int], + word2id: Dict[str, int], + need_self_loops: bool = False, +) -> k2.Fsa: + """Convert a lexicon to an FST (in k2 format). + Args: + lexicon: + The input lexicon. See also :func:`read_lexicon` + token2id: + A dict mapping tokens to IDs. + word2id: + A dict mapping words to IDs. + need_self_loops: + If True, add self-loop to states with non-epsilon output symbols + on at least one arc out of the state. The input label for this + self loop is `token2id["#0"]` and the output label is `word2id["#0"]`. + Returns: + Return an instance of `k2.Fsa` representing the given lexicon. + """ + loop_state = 0 # words enter and leave from here + next_state = 1 # the next un-allocated state, will be incremented as we go + + arcs = [] + + # The blank symbol is defined in local/train_bpe_model.py + assert token2id[""] == 0 + assert word2id[""] == 0 + + eps = 0 + + for word, pieces in lexicon: + assert len(pieces) > 0, f"{word} has no pronunciations" + cur_state = loop_state + + word = word2id[word] + pieces = [ + token2id[i] if i in token2id else token2id[""] for i in pieces + ] + + for i in range(len(pieces) - 1): + w = word if i == 0 else eps + arcs.append([cur_state, next_state, pieces[i], w, 0]) + + cur_state = next_state + next_state += 1 + + # now for the last piece of this word + i = len(pieces) - 1 + w = word if i == 0 else eps + arcs.append([cur_state, loop_state, pieces[i], w, 0]) + + if need_self_loops: + disambig_token = token2id["#0"] + disambig_word = word2id["#0"] + arcs = add_self_loops( + arcs, + disambig_token=disambig_token, + disambig_word=disambig_word, + ) + + final_state = next_state + arcs.append([loop_state, final_state, -1, -1, 0]) + arcs.append([final_state]) + + arcs = sorted(arcs, key=lambda arc: arc[0]) + arcs = [[str(i) for i in arc] for arc in arcs] + arcs = [" ".join(arc) for arc in arcs] + arcs = "\n".join(arcs) + + fsa = k2.Fsa.from_str(arcs, acceptor=False) + return fsa + + +def contain_oov(token_sym_table: Dict[str, int], tokens: List[str]) -> bool: + """Check if all the given tokens are in token symbol table. + Args: + token_sym_table: + Token symbol table that contains all the valid tokens. + tokens: + A list of tokens. + Returns: + Return True if there is any token not in the token_sym_table, + otherwise False. + """ + for tok in tokens: + if tok not in token_sym_table: + return True + return False + + +def generate_lexicon( + token_sym_table: Dict[str, int], words: List[str] +) -> Lexicon: + """Generate a lexicon from a word list and token_sym_table. + Args: + token_sym_table: + Token symbol table that mapping token to token ids. + words: + A list of strings representing words. + Returns: + Return a dict whose keys are words and values are the corresponding + tokens. + """ + lexicon = [] + for word in words: + chars = list(word.strip(" \t")) + if contain_oov(token_sym_table, chars): + continue + lexicon.append((word, chars)) + + # The OOV word is + lexicon.append(("", [""])) + return lexicon + + +def generate_tokens(text_file: str) -> Dict[str, int]: + """Generate tokens from the given text file. + Args: + text_file: + A file that contains text lines to generate tokens. + Returns: + Return a dict whose keys are tokens and values are token ids ranged + from 0 to len(keys) - 1. + """ + tokens: Dict[str, int] = dict() + tokens[""] = 0 + tokens[""] = 1 + tokens[""] = 2 + whitespace = re.compile(r"([ \t\r\n]+)") + with open(text_file, "r", encoding="utf-8") as f: + for line in f: + line = re.sub(whitespace, "", line) + tokens_list = list(line) + for token in tokens_list: + if token not in tokens: + tokens[token] = len(tokens) + return tokens + + +def main(): + parser = argparse.ArgumentParser() + parser.add_argument("--lang-dir", type=str, help="The lang directory.") + args = parser.parse_args() + + lang_dir = Path(args.lang_dir) + text_file = lang_dir / "text" + + word_sym_table = k2.SymbolTable.from_file(lang_dir / "words.txt") + + words = word_sym_table.symbols + + excluded = ["", "!SIL", "", "", "#0", "", ""] + for w in excluded: + if w in words: + words.remove(w) + + token_sym_table = generate_tokens(text_file) + + lexicon = generate_lexicon(token_sym_table, words) + + lexicon_disambig, max_disambig = add_disambig_symbols(lexicon) + + next_token_id = max(token_sym_table.values()) + 1 + for i in range(max_disambig + 1): + disambig = f"#{i}" + assert disambig not in token_sym_table + token_sym_table[disambig] = next_token_id + next_token_id += 1 + + word_sym_table.add("#0") + word_sym_table.add("") + word_sym_table.add("") + + write_mapping(lang_dir / "tokens.txt", token_sym_table) + + write_lexicon(lang_dir / "lexicon.txt", lexicon) + write_lexicon(lang_dir / "lexicon_disambig.txt", lexicon_disambig) + + L = lexicon_to_fst_no_sil( + lexicon, + token2id=token_sym_table, + word2id=word_sym_table, + ) + + L_disambig = lexicon_to_fst_no_sil( + lexicon_disambig, + token2id=token_sym_table, + word2id=word_sym_table, + need_self_loops=True, + ) + torch.save(L.as_dict(), lang_dir / "L.pt") + torch.save(L_disambig.as_dict(), lang_dir / "L_disambig.pt") + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/local/prepare_lang.py b/egs/wenetspeech/ASR/local/prepare_lang.py new file mode 120000 index 0000000000..747f2ab398 --- /dev/null +++ b/egs/wenetspeech/ASR/local/prepare_lang.py @@ -0,0 +1 @@ +../../../librispeech/ASR/local/prepare_lang.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/local/prepare_words.py b/egs/wenetspeech/ASR/local/prepare_words.py new file mode 100644 index 0000000000..65aca29839 --- /dev/null +++ b/egs/wenetspeech/ASR/local/prepare_words.py @@ -0,0 +1,84 @@ +#!/usr/bin/env python +# -*- coding: utf-8 -*- + +# Copyright 2021 Xiaomi Corp. (authors: Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + + +""" +This script takes as input words.txt without ids: + - words_no_ids.txt +and generates the new words.txt with related ids. + - words.txt +""" + + +import argparse +import logging + +from tqdm import tqdm + + +def get_parser(): + parser = argparse.ArgumentParser( + description="Prepare words.txt", + formatter_class=argparse.ArgumentDefaultsHelpFormatter, + ) + parser.add_argument( + "--input-file", + default="data/lang_char/words_no_ids.txt", + type=str, + help="the words file without ids for WenetSpeech", + ) + parser.add_argument( + "--output-file", + default="data/lang_char/words.txt", + type=str, + help="the words file with ids for WenetSpeech", + ) + + return parser + + +def main(): + parser = get_parser() + args = parser.parse_args() + + input_file = args.input_file + output_file = args.output_file + + f = open(input_file, "r", encoding="utf-8") + lines = f.readlines() + new_lines = [] + add_words = [" 0", "!SIL 1", " 2", " 3"] + new_lines.extend(add_words) + + logging.info("Starting reading the input file") + for i in tqdm(range(len(lines))): + x = lines[i] + idx = 4 + i + new_line = str(x.strip("\n")) + " " + str(idx) + new_lines.append(new_line) + + logging.info("Starting writing the words.txt") + f_out = open(output_file, "w", encoding="utf-8") + for line in new_lines: + f_out.write(line) + f_out.write("\n") + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/local/preprocess_wenetspeech.py b/egs/wenetspeech/ASR/local/preprocess_wenetspeech.py new file mode 100755 index 0000000000..6afc56ddaf --- /dev/null +++ b/egs/wenetspeech/ASR/local/preprocess_wenetspeech.py @@ -0,0 +1,120 @@ +#!/usr/bin/env python3 +# Copyright 2021 Johns Hopkins University (Piotr Żelasko) +# Copyright 2021 Xiaomi Corp. (Fangjun Kuang) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + +import logging +import re +from pathlib import Path + +from lhotse import CutSet, SupervisionSegment +from lhotse.recipes.utils import read_manifests_if_cached + +# Similar text filtering and normalization procedure as in: +# https://github.com/SpeechColab/WenetSpeech/blob/main/toolkits/kaldi/wenetspeech_data_prep.sh + + +def normalize_text( + utt: str, + # punct_pattern=re.compile(r"<(COMMA|PERIOD|QUESTIONMARK|EXCLAMATIONPOINT)>"), + punct_pattern=re.compile(r"<(PERIOD|QUESTIONMARK|EXCLAMATIONPOINT)>"), + whitespace_pattern=re.compile(r"\s\s+"), +) -> str: + return whitespace_pattern.sub(" ", punct_pattern.sub("", utt)) + + +def has_no_oov( + sup: SupervisionSegment, + oov_pattern=re.compile(r"<(SIL|MUSIC|NOISE|OTHER)>"), +) -> bool: + return oov_pattern.search(sup.text) is None + + +def preprocess_wenet_speech(): + src_dir = Path("data/manifests") + output_dir = Path("data/fbank") + output_dir.mkdir(exist_ok=True) + + dataset_parts = ( + "L", + "M", + "S", + "DEV", + "TEST_NET", + "TEST_MEETING", + ) + + logging.info("Loading manifest (may take 10 minutes)") + manifests = read_manifests_if_cached( + dataset_parts=dataset_parts, + output_dir=src_dir, + suffix="jsonl.gz", + ) + assert manifests is not None + + for partition, m in manifests.items(): + logging.info(f"Processing {partition}") + raw_cuts_path = output_dir / f"cuts_{partition}_raw.jsonl.gz" + if raw_cuts_path.is_file(): + logging.info(f"{partition} already exists - skipping") + continue + + # Note this step makes the recipe different than LibriSpeech: + # We must filter out some utterances and remove punctuation + # to be consistent with Kaldi. + logging.info("Filtering OOV utterances from supervisions") + m["supervisions"] = m["supervisions"].filter(has_no_oov) + logging.info(f"Normalizing text in {partition}") + for sup in m["supervisions"]: + text = str(sup.text) + logging.info(f"Original text: {text}") + sup.text = normalize_text(sup.text) + text = str(sup.text) + logging.info(f"Normalize text: {text}") + + # Create long-recording cut manifests. + logging.info(f"Processing {partition}") + cut_set = CutSet.from_manifests( + recordings=m["recordings"], + supervisions=m["supervisions"], + ) + # Run data augmentation that needs to be done in the + # time domain. + if partition not in ["DEV", "TEST_NET", "TEST_MEETING"]: + logging.info( + f"Speed perturb for {partition} with factors 0.9 and 1.1 " + "(Perturbing may take 8 minutes and saving may take 20 minutes)" + ) + cut_set = ( + cut_set + + cut_set.perturb_speed(0.9) + + cut_set.perturb_speed(1.1) + ) + logging.info(f"Saving to {raw_cuts_path}") + cut_set.to_file(raw_cuts_path) + + +def main(): + formatter = ( + "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s" + ) + logging.basicConfig(format=formatter, level=logging.INFO) + + preprocess_wenet_speech() + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/local/text2segments.py b/egs/wenetspeech/ASR/local/text2segments.py new file mode 100644 index 0000000000..acf6f9698b --- /dev/null +++ b/egs/wenetspeech/ASR/local/text2segments.py @@ -0,0 +1,83 @@ +#!/usr/bin/env python +# -*- coding: utf-8 -*- + +# Copyright 2021 Xiaomi Corp. (authors: Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + + +""" +This script takes as input "text", which refers to the transcript file for +WenetSpeech: + - text +and generates the output file text_word_segmentation which is implemented +with word segmenting: + - text_words_segmentation +""" + + +import argparse + +import jieba +from tqdm import tqdm + +jieba.enable_paddle() + + +def get_parser(): + parser = argparse.ArgumentParser( + description="Chinese Word Segmentation for text", + formatter_class=argparse.ArgumentDefaultsHelpFormatter, + ) + parser.add_argument( + "--input-file", + default="data/lang_char/text", + type=str, + help="the input text file for WenetSpeech", + ) + parser.add_argument( + "--output-file", + default="data/lang_char/text_words_segmentation", + type=str, + help="the text implemented with words segmenting for WenetSpeech", + ) + + return parser + + +def main(): + parser = get_parser() + args = parser.parse_args() + + input_file = args.input + output_file = args.output + + f = open(input_file, "r", encoding="utf-8") + lines = f.readlines() + new_lines = [] + for i in tqdm(range(len(lines))): + x = lines[i].rstrip() + seg_list = jieba.cut(x, use_paddle=True) + new_line = " ".join(seg_list) + new_lines.append(new_line) + + f_new = open(output_file, "w", encoding="utf-8") + for line in new_lines: + f_new.write(line) + f_new.write("\n") + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/local/text2token.py b/egs/wenetspeech/ASR/local/text2token.py new file mode 100755 index 0000000000..1c463cf1ca --- /dev/null +++ b/egs/wenetspeech/ASR/local/text2token.py @@ -0,0 +1,196 @@ +#!/usr/bin/env python3 +# Copyright 2017 Johns Hopkins University (authors: Shinji Watanabe) +# 2022 Xiaomi Corp. (authors: Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + + +import argparse +import codecs +import re +import sys +from typing import List + +from pypinyin import lazy_pinyin, pinyin + +is_python2 = sys.version_info[0] == 2 + + +def exist_or_not(i, match_pos): + start_pos = None + end_pos = None + for pos in match_pos: + if pos[0] <= i < pos[1]: + start_pos = pos[0] + end_pos = pos[1] + break + + return start_pos, end_pos + + +def get_parser(): + parser = argparse.ArgumentParser( + description="convert raw text to tokenized text", + formatter_class=argparse.ArgumentDefaultsHelpFormatter, + ) + parser.add_argument( + "--nchar", + "-n", + default=1, + type=int, + help="number of characters to split, i.e., \ + aabb -> a a b b with -n 1 and aa bb with -n 2", + ) + parser.add_argument( + "--skip-ncols", "-s", default=0, type=int, help="skip first n columns" + ) + parser.add_argument( + "--space", default="", type=str, help="space symbol" + ) + parser.add_argument( + "--non-lang-syms", + "-l", + default=None, + type=str, + help="list of non-linguistic symobles, e.g., etc.", + ) + parser.add_argument( + "text", type=str, default=False, nargs="?", help="input text" + ) + parser.add_argument( + "--trans_type", + "-t", + type=str, + default="char", + choices=["char", "pinyin", "lazy_pinyin"], + help="""Transcript type. char/pinyin/lazy_pinyin""", + ) + return parser + + +def token2id( + texts, token_table, token_type: str = "lazy_pinyin", oov: str = "" +) -> List[List[int]]: + """Convert token to id. + Args: + texts: + The input texts, it refers to the chinese text here. + token_table: + The token table is built based on "data/lang_xxx/token.txt" + token_type: + The type of token, such as "pinyin" and "lazy_pinyin". + oov: + Out of vocabulary token. When a word(token) in the transcript + does not exist in the token list, it is replaced with `oov`. + + Returns: + The list of ids for the input texts. + """ + if texts is None: + raise ValueError("texts can't be None!") + else: + oov_id = token_table[oov] + ids: List[List[int]] = [] + for text in texts: + chars_list = list(str(text)) + if token_type == "lazy_pinyin": + text = lazy_pinyin(chars_list) + sub_ids = [ + token_table[txt] if txt in token_table else oov_id + for txt in text + ] + ids.append(sub_ids) + else: # token_type = "pinyin" + text = pinyin(chars_list) + sub_ids = [ + token_table[txt[0]] if txt[0] in token_table else oov_id + for txt in text + ] + ids.append(sub_ids) + return ids + + +def main(): + parser = get_parser() + args = parser.parse_args() + + rs = [] + if args.non_lang_syms is not None: + with codecs.open(args.non_lang_syms, "r", encoding="utf-8") as f: + nls = [x.rstrip() for x in f.readlines()] + rs = [re.compile(re.escape(x)) for x in nls] + + if args.text: + f = codecs.open(args.text, encoding="utf-8") + else: + f = codecs.getreader("utf-8")( + sys.stdin if is_python2 else sys.stdin.buffer + ) + + sys.stdout = codecs.getwriter("utf-8")( + sys.stdout if is_python2 else sys.stdout.buffer + ) + line = f.readline() + n = args.nchar + while line: + x = line.split() + print(" ".join(x[: args.skip_ncols]), end=" ") + a = " ".join(x[args.skip_ncols :]) # noqa E203 + + # get all matched positions + match_pos = [] + for r in rs: + i = 0 + while i >= 0: + m = r.search(a, i) + if m: + match_pos.append([m.start(), m.end()]) + i = m.end() + else: + break + if len(match_pos) > 0: + chars = [] + i = 0 + while i < len(a): + start_pos, end_pos = exist_or_not(i, match_pos) + if start_pos is not None: + chars.append(a[start_pos:end_pos]) + i = end_pos + else: + chars.append(a[i]) + i += 1 + a = chars + + if args.trans_type == "pinyin": + a = pinyin(list(str(a))) + a = [one[0] for one in a] + + if args.trans_type == "lazy_pinyin": + a = lazy_pinyin(list(str(a))) + + a = [a[j : j + n] for j in range(0, len(a), n)] # noqa E203 + + a_flat = [] + for z in a: + a_flat.append("".join(z)) + + a_chars = [z.replace(" ", args.space) for z in a_flat] + + print("".join(a_chars)) + line = f.readline() + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/prepare.sh b/egs/wenetspeech/ASR/prepare.sh new file mode 100755 index 0000000000..d4e550d819 --- /dev/null +++ b/egs/wenetspeech/ASR/prepare.sh @@ -0,0 +1,225 @@ +#!/usr/bin/env bash + +set -eou pipefail + +nj=15 +stage=0 +stop_stage=100 + +# Split L subset to this number of pieces +# This is to avoid OOM during feature extraction. +num_splits=1000 + +# We assume dl_dir (download dir) contains the following +# directories and files. If not, they will be downloaded +# by this script automatically. +# +# - $dl_dir/WenetSpeech +# You can find audio, WenetSpeech.json inside it. +# You can apply for the download credentials by following +# https://github.com/wenet-e2e/WenetSpeech#download +# +# - $dl_dir/musan +# This directory contains the following directories downloaded from +# http://www.openslr.org/17/ +# +# - music +# - noise +# - speech + +dl_dir=$PWD/download + +. shared/parse_options.sh || exit 1 + +# All files generated by this script are saved in "data". +# You can safely remove "data" and rerun this script to regenerate it. +mkdir -p data + +log() { + # This function is from espnet + local fname=${BASH_SOURCE[1]##*/} + echo -e "$(date '+%Y-%m-%d %H:%M:%S') (${fname}:${BASH_LINENO[0]}:${FUNCNAME[1]}) $*" +} + +log "dl_dir: $dl_dir" + +if [ $stage -le 0 ] && [ $stop_stage -ge 0 ]; then + log "Stage 0: Download data" + + [ ! -e $dl_dir/WenetSpeech ] && mkdir -p $dl_dir/WenetSpeech + + # If you have pre-downloaded it to /path/to/WenetSpeech, + # you can create a symlink + # + # ln -sfv /path/to/WenetSpeech $dl_dir/WenetSpeech + # + if [ ! -d $dl_dir/WenetSpeech/wenet_speech ] && [ ! -f $dl_dir/WenetSpeech/metadata/v1.list ]; then + log "Stage 0: should download WenetSpeech first" + exit 1; + fi + + # If you have pre-downloaded it to /path/to/musan, + # you can create a symlink + # + #ln -sfv /path/to/musan $dl_dir/musan + + if [ ! -d $dl_dir/musan ]; then + lhotse download musan $dl_dir + fi +fi + +if [ $stage -le 1 ] && [ $stop_stage -ge 1 ]; then + log "Stage 1: Prepare WenetSpeech manifest" + # We assume that you have downloaded the WenetSpeech corpus + # to $dl_dir/WenetSpeech + mkdir -p data/manifests + lhotse prepare wenet-speech $dl_dir/WenetSpeech data/manifests -j $nj +fi + +if [ $stage -le 2 ] && [ $stop_stage -ge 2 ]; then + log "Stage 2: Prepare musan manifest" + # We assume that you have downloaded the musan corpus + # to data/musan + mkdir -p data/manifests + lhotse prepare musan $dl_dir/musan data/manifests +fi + +if [ $stage -le 3 ] && [ $stop_stage -ge 3 ]; then + log "Stage 3: Preprocess WenetSpeech manifest" + if [ ! -f data/fbank/.preprocess_complete ]; then + python3 ./local/preprocess_wenetspeech.py + touch data/fbank/.preprocess_complete + fi +fi + +if [ $stage -le 4 ] && [ $stop_stage -ge 4 ]; then + log "Stage 4: Compute features for DEV and TEST subsets of WenetSpeech (may take 2 minutes)" + python3 ./local/compute_fbank_wenetspeech_dev_test.py +fi + +if [ $stage -le 5 ] && [ $stop_stage -ge 5 ]; then + log "Stage 5: Split S subset into ${num_splits} pieces" + split_dir=data/fbank/S_split_${num_splits}_test + if [ ! -f $split_dir/.split_completed ]; then + lhotse split $num_splits ./data/fbank/cuts_S_raw.jsonl.gz $split_dir + touch $split_dir/.split_completed + fi +fi + +if [ $stage -le 6 ] && [ $stop_stage -ge 6 ]; then + log "Stage 6: Split M subset into ${num_splits} piece" + split_dir=data/fbank/M_split_${num_splits} + if [ ! -f $split_dir/.split_completed ]; then + lhotse split $num_splits ./data/fbank/cuts_M_raw.jsonl.gz $split_dir + touch $split_dir/.split_completed + fi +fi + +if [ $stage -le 7 ] && [ $stop_stage -ge 7 ]; then + log "Stage 7: Split L subset into ${num_splits} pieces" + split_dir=data/fbank/L_split_${num_splits} + if [ ! -f $split_dir/.split_completed ]; then + lhotse split $num_splits ./data/fbank/cuts_L_raw.jsonl.gz $split_dir + touch $split_dir/.split_completed + fi +fi + +if [ $stage -le 8 ] && [ $stop_stage -ge 8 ]; then + log "Stage 8: Compute features for S" + python3 ./local/compute_fbank_wenetspeech_splits.py \ + --training-subset S \ + --num-workers 20 \ + --batch-duration 600 \ + --start 0 \ + --num-splits $num_splits +fi + +if [ $stage -le 9 ] && [ $stop_stage -ge 9 ]; then + log "Stage 9: Compute features for M" + python3 ./local/compute_fbank_wenetspeech_splits.py \ + --training-subset M \ + --num-workers 20 \ + --batch-duration 600 \ + --start 0 \ + --num-splits $num_splits +fi + +if [ $stage -le 10 ] && [ $stop_stage -ge 10 ]; then + log "Stage 10: Compute features for L" + python3 ./local/compute_fbank_wenetspeech_splits.py \ + --training-subset L \ + --num-workers 20 \ + --batch-duration 600 \ + --start 0 \ + --num-splits $num_splits +fi + +if [ $stage -le 11 ] && [ $stop_stage -ge 11 ]; then + log "Stage 11: Combine features for S" + if [ ! -f data/fbank/cuts_S.jsonl.gz ]; then + pieces=$(find data/fbank/S_split_1000 -name "cuts_S.*.jsonl.gz") + lhotse combine $pieces data/fbank/cuts_S.jsonl.gz + fi +fi + +if [ $stage -le 12 ] && [ $stop_stage -ge 12 ]; then + log "Stage 12: Combine features for M" + if [ ! -f data/fbank/cuts_M.jsonl.gz ]; then + pieces=$(find data/fbank/M_split_1000 -name "cuts_M.*.jsonl.gz") + lhotse combine $pieces data/fbank/cuts_M.jsonl.gz + fi +fi + +if [ $stage -le 13 ] && [ $stop_stage -ge 13 ]; then + log "Stage 13: Combine features for L" + if [ ! -f data/fbank/cuts_L.jsonl.gz ]; then + pieces=$(find data/fbank/L_split_1000 -name "cuts_L.*.jsonl.gz") + lhotse combine $pieces data/fbank/cuts_L.jsonl.gz + fi +fi + +if [ $stage -le 14 ] && [ $stop_stage -ge 14 ]; then + log "Stage 14: Compute fbank for musan" + mkdir -p data/fbank + ./local/compute_fbank_musan.py +fi + +if [ $stage -le 15 ] && [ $stop_stage -ge 15 ]; then + log "Stage 15: Prepare char based lang" + lang_char_dir=data/lang_char + mkdir -p $lang_char_dir + + # Prepare text. + # Note: in Linux, you can install jq with the following command: + # wget -O jq https://github.com/stedolan/jq/release/download/jq-1.6/jq-linux64 + if [ ! -f $lang_char_dir/text ]; then + gunzip -c data/manifests/supervisions_L.jsonl.gz \ + | jq 'text' | sed 's/"//g' \ + | ./local/text2token.py -t "char" > $lang_char_dir/text + fi + + # The implementation of chinese word segmentation for text, + # and it will take about 15 minutes. + if [ ! -f $lang_char_dir/text_words_segmentation ]; then + python ./local/text2segments.py \ + --input-file $lang_char_dir/text \ + --output-file $lang_char_dir/text_words_segmentation + fi + + cat $lang_char_dir/text_words_segmentation | sed 's/ /\n/g' \ + | sort -u | sed '/^$/d' | uniq > $lang_char_dir/words_no_ids.txt + + if [ ! -f $lang_char_dir/words.txt ]; then + python ./local/prepare_words.py \ + --input-file $lang_char_dir/words_no_ids.txt \ + --output-file $lang_char_dir/words.txt + fi +fi + +if [ $stage -le 16 ] && [ $stop_stage -ge 16 ]; then + log "Stage 16: Prepare char based L_disambig.pt" + if [ ! -f data/lang_char/L_disambig.pt ]; then + python ./local/prepare_char.py \ + --lang-dir data/lang_char + fi +fi diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/__init__.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/__init__.py new file mode 100644 index 0000000000..e69de29bb2 diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/asr_datamodule.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/asr_datamodule.py new file mode 100644 index 0000000000..d2f8d85ce1 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/asr_datamodule.py @@ -0,0 +1,450 @@ +# Copyright 2021 Piotr Żelasko +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + + +import argparse +import inspect +import logging +from functools import lru_cache +from pathlib import Path +from typing import Any, Dict, List, Optional + +import torch +from lhotse import ( + CutSet, + Fbank, + FbankConfig, + load_manifest, + set_caching_enabled, +) +from lhotse.dataset import ( + CutConcatenate, + CutMix, + DynamicBucketingSampler, + K2SpeechRecognitionDataset, + PrecomputedFeatures, + SingleCutSampler, + SpecAugment, +) +from lhotse.dataset.input_strategies import OnTheFlyFeatures +from lhotse.utils import fix_random_seed +from torch.utils.data import DataLoader + +from icefall.utils import str2bool + +set_caching_enabled(False) +torch.set_num_threads(1) + + +class _SeedWorkers: + def __init__(self, seed: int): + self.seed = seed + + def __call__(self, worker_id: int): + fix_random_seed(self.seed + worker_id) + + +class WenetSpeechAsrDataModule: + """ + DataModule for k2 ASR experiments. + It assumes there is always one train and valid dataloader, + but there can be multiple test dataloaders (e.g. LibriSpeech test-clean + and test-other). + It contains all the common data pipeline modules used in ASR + experiments, e.g.: + - dynamic batch size, + - bucketing samplers, + - cut concatenation, + - augmentation, + - on-the-fly feature extraction + This class should be derived for specific corpora used in ASR tasks. + """ + + def __init__(self, args: argparse.Namespace): + self.args = args + + @classmethod + def add_arguments(cls, parser: argparse.ArgumentParser): + group = parser.add_argument_group( + title="ASR data related options", + description="These options are used for the preparation of " + "PyTorch DataLoaders from Lhotse CutSet's -- they control the " + "effective batch sizes, sampling strategies, applied data " + "augmentations, etc.", + ) + group.add_argument( + "--manifest-dir", + type=Path, + default=Path("data/fbank"), + help="Path to directory with train/valid/test cuts.", + ) + group.add_argument( + "--max-duration", + type=int, + default=200.0, + help="Maximum pooled recordings duration (seconds) in a " + "single batch. You can reduce it if it causes CUDA OOM.", + ) + group.add_argument( + "--bucketing-sampler", + type=str2bool, + default=True, + help="When enabled, the batches will come from buckets of " + "similar duration (saves padding frames).", + ) + group.add_argument( + "--num-buckets", + type=int, + default=300, + help="The number of buckets for the DynamicBucketingSampler" + "(you might want to increase it for larger datasets).", + ) + group.add_argument( + "--concatenate-cuts", + type=str2bool, + default=False, + help="When enabled, utterances (cuts) will be concatenated " + "to minimize the amount of padding.", + ) + group.add_argument( + "--duration-factor", + type=float, + default=1.0, + help="Determines the maximum duration of a concatenated cut " + "relative to the duration of the longest cut in a batch.", + ) + group.add_argument( + "--gap", + type=float, + default=1.0, + help="The amount of padding (in seconds) inserted between " + "concatenated cuts. This padding is filled with noise when " + "noise augmentation is used.", + ) + group.add_argument( + "--on-the-fly-feats", + type=str2bool, + default=False, + help="When enabled, use on-the-fly cut mixing and feature " + "extraction. Will drop existing precomputed feature manifests " + "if available.", + ) + group.add_argument( + "--shuffle", + type=str2bool, + default=True, + help="When enabled (=default), the examples will be " + "shuffled for each epoch.", + ) + group.add_argument( + "--return-cuts", + type=str2bool, + default=True, + help="When enabled, each batch will have the " + "field: batch['supervisions']['cut'] with the cuts that " + "were used to construct it.", + ) + + group.add_argument( + "--num-workers", + type=int, + default=2, + help="The number of training dataloader workers that " + "collect the batches.", + ) + + group.add_argument( + "--enable-spec-aug", + type=str2bool, + default=True, + help="When enabled, use SpecAugment for training dataset.", + ) + + group.add_argument( + "--spec-aug-time-warp-factor", + type=int, + default=80, + help="Used only when --enable-spec-aug is True. " + "It specifies the factor for time warping in SpecAugment. " + "Larger values mean more warping. " + "A value less than 1 means to disable time warp.", + ) + + group.add_argument( + "--enable-musan", + type=str2bool, + default=True, + help="When enabled, select noise from MUSAN and mix it" + "with training dataset. ", + ) + + group.add_argument( + "--lazy-load", + type=str2bool, + default=True, + help="lazily open CutSets to avoid OOM (for L|XL subset)", + ) + + group.add_argument( + "--training-subset", + type=str, + default="L", + help="The training subset for using", + ) + + def train_dataloaders( + self, + cuts_train: CutSet, + sampler_state_dict: Optional[Dict[str, Any]] = None, + ) -> DataLoader: + """ + Args: + cuts_train: + CutSet for training. + sampler_state_dict: + The state dict for the training sampler. + """ + logging.info("About to get Musan cuts") + cuts_musan = load_manifest( + self.args.manifest_dir / "cuts_musan.json.gz" + ) + + transforms = [] + if self.args.enable_musan: + logging.info("Enable MUSAN") + transforms.append( + CutMix( + cuts=cuts_musan, prob=0.5, snr=(10, 20), preserve_id=True + ) + ) + else: + logging.info("Disable MUSAN") + + if self.args.concatenate_cuts: + logging.info( + f"Using cut concatenation with duration factor " + f"{self.args.duration_factor} and gap {self.args.gap}." + ) + # Cut concatenation should be the first transform in the list, + # so that if we e.g. mix noise in, it will fill the gaps between + # different utterances. + transforms = [ + CutConcatenate( + duration_factor=self.args.duration_factor, gap=self.args.gap + ) + ] + transforms + + input_transforms = [] + if self.args.enable_spec_aug: + logging.info("Enable SpecAugment") + logging.info( + f"Time warp factor: {self.args.spec_aug_time_warp_factor}" + ) + # Set the value of num_frame_masks according to Lhotse's version. + # In different Lhotse's versions, the default of num_frame_masks is + # different. + num_frame_masks = 10 + num_frame_masks_parameter = inspect.signature( + SpecAugment.__init__ + ).parameters["num_frame_masks"] + if num_frame_masks_parameter.default == 1: + num_frame_masks = 2 + logging.info(f"Num frame mask: {num_frame_masks}") + input_transforms.append( + SpecAugment( + time_warp_factor=self.args.spec_aug_time_warp_factor, + num_frame_masks=num_frame_masks, + features_mask_size=27, + num_feature_masks=2, + frames_mask_size=100, + ) + ) + else: + logging.info("Disable SpecAugment") + + logging.info("About to create train dataset") + train = K2SpeechRecognitionDataset( + cut_transforms=transforms, + input_transforms=input_transforms, + return_cuts=self.args.return_cuts, + ) + + if self.args.on_the_fly_feats: + # NOTE: the PerturbSpeed transform should be added only if we + # remove it from data prep stage. + # Add on-the-fly speed perturbation; since originally it would + # have increased epoch size by 3, we will apply prob 2/3 and use + # 3x more epochs. + # Speed perturbation probably should come first before + # concatenation, but in principle the transforms order doesn't have + # to be strict (e.g. could be randomized) + # transforms = [PerturbSpeed(factors=[0.9, 1.1], p=2/3)] + transforms # noqa + # Drop feats to be on the safe side. + train = K2SpeechRecognitionDataset( + cut_transforms=transforms, + input_strategy=OnTheFlyFeatures( + Fbank(FbankConfig(num_mel_bins=80)) + ), + input_transforms=input_transforms, + return_cuts=self.args.return_cuts, + ) + + if self.args.bucketing_sampler: + logging.info("Using DynamicBucketingSampler.") + train_sampler = DynamicBucketingSampler( + cuts_train, + max_duration=self.args.max_duration, + shuffle=self.args.shuffle, + num_buckets=self.args.num_buckets, + buffer_size=30000, + drop_last=True, + ) + else: + logging.info("Using SingleCutSampler.") + train_sampler = SingleCutSampler( + cuts_train, + max_duration=self.args.max_duration, + shuffle=self.args.shuffle, + ) + logging.info("About to create train dataloader") + + # 'seed' is derived from the current random state, which will have + # previously been set in the main process. + seed = torch.randint(0, 100000, ()).item() + worker_init_fn = _SeedWorkers(seed) + + train_dl = DataLoader( + train, + sampler=train_sampler, + batch_size=None, + num_workers=self.args.num_workers, + persistent_workers=False, + worker_init_fn=worker_init_fn, + ) + + if sampler_state_dict is not None: + logging.info("Loading sampler state dict") + train_dl.sampler.load_state_dict(sampler_state_dict) + + return train_dl + + def valid_dataloaders(self, cuts_valid: CutSet) -> DataLoader: + transforms = [] + if self.args.concatenate_cuts: + transforms = [ + CutConcatenate( + duration_factor=self.args.duration_factor, gap=self.args.gap + ) + ] + transforms + + logging.info("About to create dev dataset") + if self.args.on_the_fly_feats: + validate = K2SpeechRecognitionDataset( + cut_transforms=transforms, + input_strategy=OnTheFlyFeatures( + Fbank(FbankConfig(num_mel_bins=80)) + ), + return_cuts=self.args.return_cuts, + ) + else: + validate = K2SpeechRecognitionDataset( + cut_transforms=transforms, + return_cuts=self.args.return_cuts, + ) + valid_sampler = DynamicBucketingSampler( + cuts_valid, + max_duration=self.args.max_duration, + rank=0, + world_size=1, + shuffle=False, + ) + logging.info("About to create dev dataloader") + + from lhotse.dataset.iterable_dataset import IterableDatasetWrapper + + dev_iter_dataset = IterableDatasetWrapper( + dataset=validate, + sampler=valid_sampler, + ) + valid_dl = DataLoader( + dev_iter_dataset, + batch_size=None, + num_workers=self.args.num_workers, + persistent_workers=False, + ) + + return valid_dl + + def test_dataloaders(self, cuts: CutSet) -> DataLoader: + logging.debug("About to create test dataset") + test = K2SpeechRecognitionDataset( + input_strategy=OnTheFlyFeatures(Fbank(FbankConfig(num_mel_bins=80))) + if self.args.on_the_fly_feats + else PrecomputedFeatures(), + return_cuts=self.args.return_cuts, + ) + sampler = DynamicBucketingSampler( + cuts, + max_duration=self.args.max_duration, + rank=0, + world_size=1, + shuffle=False, + ) + from lhotse.dataset.iterable_dataset import IterableDatasetWrapper + + test_iter_dataset = IterableDatasetWrapper( + dataset=test, + sampler=sampler, + ) + test_dl = DataLoader( + test_iter_dataset, + batch_size=None, + num_workers=self.args.num_workers, + ) + return test_dl + + @lru_cache() + def train_cuts(self) -> CutSet: + logging.info("About to get train cuts") + if self.args.lazy_load: + logging.info("use lazy cuts") + cuts_train = CutSet.from_jsonl_lazy( + self.args.manifest_dir + / f"cuts_{self.args.training_subset}.jsonl.gz" + ) + else: + cuts_train = CutSet.from_file( + self.args.manifest_dir + / f"cuts_{self.args.training_subset}.jsonl.gz" + ) + return cuts_train + + @lru_cache() + def valid_cuts(self) -> CutSet: + logging.info("About to get dev cuts") + return load_manifest(self.args.manifest_dir / "cuts_DEV.jsonl.gz") + + @lru_cache() + def test_net_cuts(self) -> List[CutSet]: + logging.info("About to get TEST_NET cuts") + return load_manifest(self.args.manifest_dir / "cuts_TEST_NET.jsonl.gz") + + @lru_cache() + def test_meeting_cuts(self) -> List[CutSet]: + logging.info("About to get TEST_MEETING cuts") + return load_manifest( + self.args.manifest_dir / "cuts_TEST_MEETING.jsonl.gz" + ) diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/beam_search.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/beam_search.py new file mode 120000 index 0000000000..e24eca39f2 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/beam_search.py @@ -0,0 +1 @@ +../../../librispeech/ASR/pruned_transducer_stateless2/beam_search.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/conformer.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/conformer.py new file mode 120000 index 0000000000..a659571806 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/conformer.py @@ -0,0 +1 @@ +../../../librispeech/ASR/pruned_transducer_stateless2/conformer.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/decode.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/decode.py new file mode 100755 index 0000000000..be3d01f6a2 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/decode.py @@ -0,0 +1,623 @@ +#!/usr/bin/env python3 +# +# Copyright 2021 Xiaomi Corporation (Author: Fangjun Kuang) +# Copyright 2022 Xiaomi Corporation (Author: Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. +""" +When training with the L subset, usage: +(1) greedy search +./pruned_transducer_stateless2/decode.py \ + --epoch 10 \ + --avg 2 \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --lang-dir data/lang_char \ + --max-duration 100 \ + --decoding-method greedy_search + +(2) modified beam search +./pruned_transducer_stateless2/decode.py \ + --epoch 10 \ + --avg 2 \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --lang-dir data/lang_char \ + --max-duration 100 \ + --decoding-method modified_beam_search \ + --beam-size 4 + +(3) fast beam search +./pruned_transducer_stateless2/decode.py \ + --epoch 10 \ + --avg 2 \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --lang-dir data/lang_char \ + --max-duration 1500 \ + --decoding-method fast_beam_search \ + --beam 4 \ + --max-contexts 4 \ + --max-states 8 +""" + + +import argparse +import logging +from collections import defaultdict +from pathlib import Path +from typing import Dict, List, Optional, Tuple + +import k2 +import torch +import torch.nn as nn +from asr_datamodule import WenetSpeechAsrDataModule +from beam_search import ( + beam_search, + fast_beam_search, + greedy_search, + greedy_search_batch, + modified_beam_search, +) +from train import get_params, get_transducer_model + +from icefall.checkpoint import ( + average_checkpoints, + find_checkpoints, + load_checkpoint, +) +from icefall.lexicon import Lexicon +from icefall.utils import ( + AttributeDict, + setup_logger, + store_transcripts, + write_error_stats, +) + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--epoch", + type=int, + default=28, + help="It specifies the checkpoint to use for decoding." + "Note: Epoch counts from 0.", + ) + + parser.add_argument( + "--batch", + type=int, + default=None, + help="It specifies the batch checkpoint to use for decoding." + "Note: Epoch counts from 0.", + ) + + parser.add_argument( + "--avg", + type=int, + default=15, + help="Number of checkpoints to average. Automatically select " + "consecutive checkpoints before the checkpoint specified by " + "'--epoch'. ", + ) + + parser.add_argument( + "--avg-last-n", + type=int, + default=0, + help="""If positive, --epoch and --avg are ignored and it + will use the last n checkpoints exp_dir/checkpoint-xxx.pt + where xxx is the number of processed batches while + saving that checkpoint. + """, + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="pruned_transducer_stateless2/exp", + help="The experiment dir", + ) + + parser.add_argument( + "--lang-dir", + type=str, + default="data/lang_char", + help="""The lang dir + It contains language related input files such as + "lexicon.txt" + """, + ) + + parser.add_argument( + "--decoding-method", + type=str, + default="greedy_search", + help="""Possible values are: + - greedy_search + - beam_search + - modified_beam_search + - fast_beam_search + """, + ) + + parser.add_argument( + "--beam-size", + type=int, + default=4, + help="""An interger indicating how many candidates we will keep for each + frame. Used only when --decoding-method is beam_search or + modified_beam_search.""", + ) + + parser.add_argument( + "--beam", + type=float, + default=4, + help="""A floating point value to calculate the cutoff score during beam + search (i.e., `cutoff = max-score - beam`), which is the same as the + `beam` in Kaldi. + Used only when --decoding-method is fast_beam_search""", + ) + + parser.add_argument( + "--max-contexts", + type=int, + default=4, + help="""Used only when --decoding-method is + fast_beam_search""", + ) + + parser.add_argument( + "--max-states", + type=int, + default=8, + help="""Used only when --decoding-method is + fast_beam_search""", + ) + + parser.add_argument( + "--context-size", + type=int, + default=2, + help="The context size in the decoder. 1 means bigram; " + "2 means tri-gram", + ) + parser.add_argument( + "--max-sym-per-frame", + type=int, + default=1, + help="""Maximum number of symbols per frame. + Used only when --decoding_method is greedy_search""", + ) + + return parser + + +def decode_one_batch( + params: AttributeDict, + model: nn.Module, + lexicon: Lexicon, + batch: dict, + decoding_graph: Optional[k2.Fsa] = None, +) -> Dict[str, List[List[str]]]: + """Decode one batch and return the result in a dict. The dict has the + following format: + + - key: It indicates the setting used for decoding. For example, + if greedy_search is used, it would be "greedy_search" + If beam search with a beam size of 7 is used, it would be + "beam_7" + - value: It contains the decoding result. `len(value)` equals to + batch size. `value[i]` is the decoding result for the i-th + utterance in the given batch. + Args: + params: + It's the return value of :func:`get_params`. + model: + The neural model. + batch: + It is the return value from iterating + `lhotse.dataset.K2SpeechRecognitionDataset`. See its documentation + for the format of the `batch`. + decoding_graph: + The decoding graph. Can be either a `k2.trivial_graph` or HLG, Used + only when --decoding_method is fast_beam_search. + Returns: + Return the decoding result. See above description for the format of + the returned dict. + """ + device = model.device + feature = batch["inputs"] + assert feature.ndim == 3 + + feature = feature.to(device) + # at entry, feature is (N, T, C) + + supervisions = batch["supervisions"] + feature_lens = supervisions["num_frames"].to(device) + + encoder_out, encoder_out_lens = model.encoder( + x=feature, x_lens=feature_lens + ) + hyps = [] + + if params.decoding_method == "fast_beam_search": + hyp_tokens = fast_beam_search( + model=model, + decoding_graph=decoding_graph, + encoder_out=encoder_out, + encoder_out_lens=encoder_out_lens, + beam=params.beam, + max_contexts=params.max_contexts, + max_states=params.max_states, + ) + for i in range(encoder_out.size(0)): + hyps.append([lexicon.token_table[idx] for idx in hyp_tokens[i]]) + elif ( + params.decoding_method == "greedy_search" + and params.max_sym_per_frame == 1 + ): + hyp_tokens = greedy_search_batch( + model=model, + encoder_out=encoder_out, + ) + for i in range(encoder_out.size(0)): + hyps.append([lexicon.token_table[idx] for idx in hyp_tokens[i]]) + elif params.decoding_method == "modified_beam_search": + hyp_tokens = modified_beam_search( + model=model, + encoder_out=encoder_out, + beam=params.beam_size, + ) + for i in range(encoder_out.size(0)): + hyps.append([lexicon.token_table[idx] for idx in hyp_tokens[i]]) + 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([lexicon.token_table[idx] for idx in hyp]) + + if params.decoding_method == "greedy_search": + return {"greedy_search": hyps} + elif params.decoding_method == "fast_beam_search": + return { + ( + f"beam_{params.beam}_" + f"max_contexts_{params.max_contexts}_" + f"max_states_{params.max_states}" + ): hyps + } + else: + return {f"beam_size_{params.beam_size}": hyps} + + +def decode_dataset( + dl: torch.utils.data.DataLoader, + params: AttributeDict, + model: nn.Module, + lexicon: Lexicon, + decoding_graph: Optional[k2.Fsa] = None, +) -> Dict[str, List[Tuple[List[str], List[str]]]]: + """Decode dataset. + + Args: + dl: + PyTorch's dataloader containing the dataset to decode. + params: + It is returned by :func:`get_params`. + model: + The neural model. + decoding_graph: + The decoding graph. Can be either a `k2.trivial_graph` or HLG, Used + only when --decoding_method is fast_beam_search. + Returns: + Return a dict, whose key may be "greedy_search" if greedy search + is used, or it may be "beam_7" if beam size of 7 is used. + Its value is a list of tuples. Each tuple contains two elements: + The first is the reference transcript, and the second is the + predicted result. + """ + num_cuts = 0 + + try: + num_batches = len(dl) + except TypeError: + num_batches = "?" + + if params.decoding_method == "greedy_search": + log_interval = 100 + else: + log_interval = 2 + + results = defaultdict(list) + for batch_idx, batch in enumerate(dl): + texts = batch["supervisions"]["text"] + texts = [list(str(text)) for text in texts] + + hyps_dict = decode_one_batch( + params=params, + model=model, + lexicon=lexicon, + decoding_graph=decoding_graph, + batch=batch, + ) + + for name, hyps in hyps_dict.items(): + this_batch = [] + assert len(hyps) == len(texts) + for hyp_words, ref_text in zip(hyps, texts): + this_batch.append((ref_text, hyp_words)) + + results[name].extend(this_batch) + + num_cuts += len(texts) + + if batch_idx % log_interval == 0: + batch_str = f"{batch_idx}/{num_batches}" + + logging.info( + f"batch {batch_str}, cuts processed until now is {num_cuts}" + ) + return results + + +def save_results( + params: AttributeDict, + test_set_name: str, + results_dict: Dict[str, List[Tuple[List[int], List[int]]]], +): + test_set_wers = dict() + for key, results in results_dict.items(): + recog_path = ( + params.res_dir / f"recogs-{test_set_name}-{key}-{params.suffix}.txt" + ) + store_transcripts(filename=recog_path, texts=results) + logging.info(f"The transcripts are stored in {recog_path}") + + # The following prints out WERs, per-word error statistics and aligned + # ref/hyp pairs. + errs_filename = ( + params.res_dir / f"errs-{test_set_name}-{key}-{params.suffix}.txt" + ) + with open(errs_filename, "w") as f: + wer = write_error_stats( + f, f"{test_set_name}-{key}", results, enable_log=True + ) + test_set_wers[key] = wer + + logging.info("Wrote detailed error stats to {}".format(errs_filename)) + + test_set_wers = sorted(test_set_wers.items(), key=lambda x: x[1]) + errs_info = ( + params.res_dir + / f"wer-summary-{test_set_name}-{key}-{params.suffix}.txt" + ) + with open(errs_info, "w") as f: + print("settings\tWER", file=f) + for key, val in test_set_wers: + print("{}\t{}".format(key, val), file=f) + + s = "\nFor {}, WER of different settings are:\n".format(test_set_name) + note = "\tbest for {}".format(test_set_name) + for key, val in test_set_wers: + s += "{}\t{}{}\n".format(key, val, note) + note = "" + logging.info(s) + + +@torch.no_grad() +def main(): + parser = get_parser() + WenetSpeechAsrDataModule.add_arguments(parser) + args = parser.parse_args() + args.exp_dir = Path(args.exp_dir) + + params = get_params() + params.update(vars(args)) + + assert params.decoding_method in ( + "greedy_search", + "beam_search", + "fast_beam_search", + "modified_beam_search", + ) + params.res_dir = params.exp_dir / params.decoding_method + + params.suffix = f"epoch-{params.epoch}-avg-{params.avg}" + if "fast_beam_search" in params.decoding_method: + params.suffix += f"-beam-{params.beam}" + params.suffix += f"-max-contexts-{params.max_contexts}" + params.suffix += f"-max-states-{params.max_states}" + elif "beam_search" in params.decoding_method: + params.suffix += f"-beam-{params.beam_size}" + else: + params.suffix += f"-context-{params.context_size}" + params.suffix += f"-max-sym-per-frame-{params.max_sym_per_frame}" + + setup_logger(f"{params.res_dir}/log-decode-{params.suffix}") + logging.info("Decoding started") + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", 0) + + logging.info(f"Device: {device}") + + lexicon = Lexicon(params.lang_dir) + params.blank_id = lexicon.token_table[""] + params.vocab_size = max(lexicon.tokens) + 1 + + 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) + elif params.batch is not None: + filenames = f"{params.exp_dir}/checkpoint-{params.batch}.pt" + logging.info(f"averaging {filenames}") + model.to(device) + model.load_state_dict(average_checkpoints([filenames], device=device)) + 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 + + if params.decoding_method == "fast_beam_search": + decoding_graph = k2.trivial_graph(params.vocab_size - 1, device=device) + else: + decoding_graph = None + + num_param = sum([p.numel() for p in model.parameters()]) + logging.info(f"Number of model parameters: {num_param}") + + # Note: Please use "pip install webdataset==0.1.103" + # for installing the webdataset. + import glob + import os + + from lhotse import CutSet + from lhotse.dataset.webdataset import export_to_webdataset + + wenetspeech = WenetSpeechAsrDataModule(args) + + dev = "dev" + test_net = "test_net" + test_meeting = "test_meeting" + + if not os.path.exists(f"{dev}/shared-0.tar"): + os.makedirs(dev) + dev_cuts = wenetspeech.valid_cuts() + export_to_webdataset( + dev_cuts, + output_path=f"{dev}/shared-%d.tar", + shard_size=300, + ) + + if not os.path.exists(f"{test_net}/shared-0.tar"): + os.makedirs(test_net) + test_net_cuts = wenetspeech.test_net_cuts() + export_to_webdataset( + test_net_cuts, + output_path=f"{test_net}/shared-%d.tar", + shard_size=300, + ) + + if not os.path.exists(f"{test_meeting}/shared-0.tar"): + os.makedirs(test_meeting) + test_meeting_cuts = wenetspeech.test_meeting_cuts() + export_to_webdataset( + test_meeting_cuts, + output_path=f"{test_meeting}/shared-%d.tar", + shard_size=300, + ) + + dev_shards = [ + str(path) + for path in sorted(glob.glob(os.path.join(dev, "shared-*.tar"))) + ] + cuts_dev_webdataset = CutSet.from_webdataset( + dev_shards, + split_by_worker=True, + split_by_node=True, + shuffle_shards=True, + ) + + test_net_shards = [ + str(path) + for path in sorted(glob.glob(os.path.join(test_net, "shared-*.tar"))) + ] + cuts_test_net_webdataset = CutSet.from_webdataset( + test_net_shards, + split_by_worker=True, + split_by_node=True, + shuffle_shards=True, + ) + + test_meeting_shards = [ + str(path) + for path in sorted( + glob.glob(os.path.join(test_meeting, "shared-*.tar")) + ) + ] + cuts_test_meeting_webdataset = CutSet.from_webdataset( + test_meeting_shards, + split_by_worker=True, + split_by_node=True, + shuffle_shards=True, + ) + + dev_dl = wenetspeech.valid_dataloaders(cuts_dev_webdataset) + test_net_dl = wenetspeech.test_dataloaders(cuts_test_net_webdataset) + test_meeting_dl = wenetspeech.test_dataloaders(cuts_test_meeting_webdataset) + + test_sets = ["DEV", "TEST_NET", "TEST_MEETING"] + test_dl = [dev_dl, test_net_dl, test_meeting_dl] + + for test_set, test_dl in zip(test_sets, test_dl): + results_dict = decode_dataset( + dl=test_dl, + params=params, + model=model, + lexicon=lexicon, + decoding_graph=decoding_graph, + ) + save_results( + params=params, + test_set_name=test_set, + results_dict=results_dict, + ) + + logging.info("Done!") + + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/decoder.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/decoder.py new file mode 120000 index 0000000000..722e1c8941 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/decoder.py @@ -0,0 +1 @@ +../../../librispeech/ASR/pruned_transducer_stateless2/decoder.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/encoder_interface.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/encoder_interface.py new file mode 120000 index 0000000000..653c5b09af --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/encoder_interface.py @@ -0,0 +1 @@ +../../../librispeech/ASR/transducer_stateless/encoder_interface.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/export.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/export.py new file mode 100644 index 0000000000..8c4f92c81d --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/export.py @@ -0,0 +1,178 @@ +# Copyright 2021 Xiaomi Corporation (Author: Fangjun Kuang) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. + +# This script converts several saved checkpoints +# to a single one using model averaging. +""" +Usage: +./pruned_transducer_stateless2/export.py \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --lang-dir data/lang_char \ + --epoch 10 \ + --avg 2 + +It will generate a file exp_dir/pretrained.pt + +To use the generated file with `pruned_transducer_stateless2/decode.py`, +you can do: + + cd /path/to/exp_dir + ln -s pretrained.pt epoch-9999.pt + + cd /path/to/egs/wenetspeech/ASR + ./pruned_transducer_stateless2/decode.py \ + --exp-dir ./pruned_transducer_stateless2/exp \ + --epoch 10 \ + --avg 2 \ + --max-duration 100 \ + --lang-dir data/lang_char +""" + +import argparse +import logging +from pathlib import Path + +import torch +from train import get_params, get_transducer_model + +from icefall.checkpoint import average_checkpoints, load_checkpoint +from icefall.lexicon import Lexicon +from icefall.utils import str2bool + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--epoch", + type=int, + default=28, + help="It specifies the checkpoint to use for decoding." + "Note: Epoch counts from 0.", + ) + + parser.add_argument( + "--avg", + type=int, + default=15, + help="Number of checkpoints to average. Automatically select " + "consecutive checkpoints before the checkpoint specified by " + "'--epoch'. ", + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="pruned_transducer_stateless2/exp", + help="""It specifies the directory where all training related + files, e.g., checkpoints, log, etc, are saved + """, + ) + + parser.add_argument( + "--lang-dir", + type=str, + default="data/lang_char", + help="The lang dir", + ) + + parser.add_argument( + "--jit", + type=str2bool, + default=False, + help="""True to save a model after applying torch.jit.script. + """, + ) + + parser.add_argument( + "--context-size", + type=int, + default=2, + help="The context size in the decoder. 1 means bigram; " + "2 means tri-gram", + ) + + return parser + + +def main(): + args = get_parser().parse_args() + args.exp_dir = Path(args.exp_dir) + + assert args.jit is False, "Support torchscript will be added later" + + params = get_params() + params.update(vars(args)) + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", 0) + + logging.info(f"device: {device}") + + lexicon = Lexicon(params.lang_dir) + + params.blank_id = 0 + params.vocab_size = max(lexicon.tokens) + 1 + + logging.info(params) + + logging.info("About to create model") + model = get_transducer_model(params) + + model.to(device) + + if 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.eval() + + model.to("cpu") + model.eval() + + if params.jit: + logging.info("Using torch.jit.script") + model = torch.jit.script(model) + filename = params.exp_dir / "cpu_jit.pt" + model.save(str(filename)) + logging.info(f"Saved to {filename}") + else: + logging.info("Not using torch.jit.script") + # Save it using a format so that it can be loaded + # by :func:`load_checkpoint` + filename = params.exp_dir / "pretrained.pt" + torch.save({"model": model.state_dict()}, str(filename)) + logging.info(f"Saved to {filename}") + + +if __name__ == "__main__": + formatter = ( + "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s" + ) + + logging.basicConfig(format=formatter, level=logging.INFO) + main() diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/joiner.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/joiner.py new file mode 120000 index 0000000000..9052f3cbb9 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/joiner.py @@ -0,0 +1 @@ +../../../librispeech/ASR/pruned_transducer_stateless2/joiner.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/model.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/model.py new file mode 120000 index 0000000000..a99e743342 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/model.py @@ -0,0 +1 @@ +../../../librispeech/ASR/pruned_transducer_stateless2/model.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/optim.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/optim.py new file mode 120000 index 0000000000..0a2f285aa4 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/optim.py @@ -0,0 +1 @@ +../../../librispeech/ASR/pruned_transducer_stateless2/optim.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/pretrained.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/pretrained.py new file mode 100644 index 0000000000..27ffc3bfc0 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/pretrained.py @@ -0,0 +1,342 @@ +#!/usr/bin/env python3 +# Copyright 2021 Xiaomi Corp. (authors: Fangjun Kuang) +# 2022 Xiaomi Crop. (authors: Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. +""" +Usage: +(1) greedy search +./pruned_transducer_stateless2/pretrained.py \ + --checkpoint ./pruned_transducer_stateless2/exp/pretrained.pt \ + --lang-dir ./data/lang_char \ + --method greedy_search \ + --max-sym-per-frame 1 \ + /path/to/foo.wav \ + /path/to/bar.wav +(2) modified beam search +./pruned_transducer_stateless2/pretrained.py \ + --checkpoint ./pruned_transducer_stateless2/exp/pretrained.pt \ + --lang-dir ./data/lang_char \ + --method modified_beam_search \ + --beam-size 4 \ + /path/to/foo.wav \ + /path/to/bar.wav +(3) fast beam search +./pruned_transducer_stateless2/pretrained.py \ + --checkpoint ./pruned_transducer_stateless/exp/pretrained.pt \ + --lang-dir ./data/lang_char \ + --method fast_beam_search \ + --beam 4 \ + --max-contexts 4 \ + --max-states 8 \ + /path/to/foo.wav \ + /path/to/bar.wav +You can also use `./pruned_transducer_stateless2/exp/epoch-xx.pt`. +Note: ./pruned_transducer_stateless2/exp/pretrained.pt is generated by +./pruned_transducer_stateless2/export.py +""" + + +import argparse +import logging +import math +from typing import List + +import k2 +import kaldifeat +import torch +import torchaudio +from beam_search import ( + beam_search, + fast_beam_search_one_best, + greedy_search, + greedy_search_batch, + modified_beam_search, +) +from torch.nn.utils.rnn import pad_sequence +from train import get_params, get_transducer_model + +from icefall.lexicon import Lexicon + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--checkpoint", + type=str, + required=True, + help="Path to the checkpoint. " + "The checkpoint is assumed to be saved by " + "icefall.checkpoint.save_checkpoint().", + ) + + parser.add_argument( + "--lang-dir", + type=str, + help="""Path to lang. + """, + ) + + parser.add_argument( + "--decoding-method", + type=str, + default="greedy_search", + help="""Possible values are: + - greedy_search + - modified_beam_search + - fast_beam_search + """, + ) + + parser.add_argument( + "sound_files", + type=str, + nargs="+", + help="The input sound file(s) to transcribe. " + "Supported formats are those supported by torchaudio.load(). " + "For example, wav and flac are supported. " + "The sample rate has to be 16kHz.", + ) + + parser.add_argument( + "--sample-rate", + type=int, + default=48000, + help="The sample rate of the input sound file", + ) + + parser.add_argument( + "--beam-size", + type=int, + default=4, + help="Used only when --method is beam_search and modified_beam_search ", + ) + + parser.add_argument( + "--beam", + type=float, + default=4, + help="""A floating point value to calculate the cutoff score during beam + search (i.e., `cutoff = max-score - beam`), which is the same as the + `beam` in Kaldi. + Used only when --decoding-method is fast_beam_search""", + ) + + parser.add_argument( + "--max-contexts", + type=int, + default=4, + help="""Used only when --decoding-method is + fast_beam_search""", + ) + + parser.add_argument( + "--max-states", + type=int, + default=8, + help="""Used only when --decoding-method is + fast_beam_search""", + ) + + parser.add_argument( + "--context-size", + type=int, + default=2, + help="The context size in the decoder. 1 means bigram; " + "2 means tri-gram", + ) + + parser.add_argument( + "--max-sym-per-frame", + type=int, + default=1, + help="""Maximum number of symbols per frame. Used only when + --method is greedy_search. + """, + ) + + return parser + + +def read_sound_files( + filenames: List[str], expected_sample_rate: float +) -> List[torch.Tensor]: + """Read a list of sound files into a list 1-D float32 torch tensors. + Args: + filenames: + A list of sound filenames. + expected_sample_rate: + The expected sample rate of the sound files. + Returns: + Return a list of 1-D float32 torch tensors. + """ + ans = [] + for f in filenames: + wave, sample_rate = torchaudio.load(f) + assert sample_rate == expected_sample_rate, ( + f"expected sample rate: {expected_sample_rate}. " + f"Given: {sample_rate}" + ) + # We use only the first channel + ans.append(wave[0]) + return ans + + +@torch.no_grad() +def main(): + parser = get_parser() + args = parser.parse_args() + + params = get_params() + + params.update(vars(args)) + + lexicon = Lexicon(params.lang_dir) + params.blank_id = lexicon.token_table[""] + params.vocab_size = max(lexicon.tokens) + 1 + + logging.info(f"{params}") + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", 0) + + logging.info(f"device: {device}") + + logging.info("Creating model") + model = get_transducer_model(params) + + checkpoint = torch.load(args.checkpoint, map_location="cpu") + model.load_state_dict(checkpoint["model"], strict=False) + model.to(device) + model.eval() + model.device = device + + if params.decoding_method == "fast_beam_search": + decoding_graph = k2.trivial_graph(params.vocab_size - 1, device=device) + else: + decoding_graph = None + + logging.info("Constructing Fbank computer") + opts = kaldifeat.FbankOptions() + opts.device = device + opts.frame_opts.dither = 0 + opts.frame_opts.snip_edges = False + opts.frame_opts.samp_freq = params.sample_rate + opts.mel_opts.num_bins = params.feature_dim + + fbank = kaldifeat.Fbank(opts) + + logging.info(f"Reading sound files: {params.sound_files}") + waves = read_sound_files( + filenames=params.sound_files, expected_sample_rate=params.sample_rate + ) + waves = [w.to(device) for w in waves] + + logging.info("Decoding started") + features = fbank(waves) + feature_lengths = [f.size(0) for f in features] + + features = pad_sequence( + features, batch_first=True, padding_value=math.log(1e-10) + ) + + feature_lengths = torch.tensor(feature_lengths, device=device) + + with torch.no_grad(): + encoder_out, encoder_out_lens = model.encoder( + x=features, x_lens=feature_lengths + ) + + hyps = [] + msg = f"Using {params.decoding_method}" + logging.info(msg) + + if params.decoding_method == "fast_beam_search": + hyp_tokens = fast_beam_search_one_best( + model=model, + decoding_graph=decoding_graph, + encoder_out=encoder_out, + encoder_out_lens=encoder_out_lens, + beam=params.beam, + max_contexts=params.max_contexts, + max_states=params.max_states, + ) + for i in range(encoder_out.size(0)): + hyps.append([lexicon.token_table[idx] for idx in hyp_tokens[i]]) + elif ( + params.decoding_method == "greedy_search" + and params.max_sym_per_frame == 1 + ): + hyp_tokens = greedy_search_batch( + model=model, + encoder_out=encoder_out, + encoder_out_lens=encoder_out_lens, + ) + for i in range(encoder_out.size(0)): + hyps.append([lexicon.token_table[idx] for idx in hyp_tokens[i]]) + 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 i in range(encoder_out.size(0)): + hyps.append([lexicon.token_table[idx] for idx in hyp_tokens[i]]) + 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([lexicon.token_table[idx] for idx in hyp]) + + 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() diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/scaling.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/scaling.py new file mode 120000 index 0000000000..c10cdfe12e --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/scaling.py @@ -0,0 +1 @@ +../../../librispeech/ASR/pruned_transducer_stateless2/scaling.py \ No newline at end of file diff --git a/egs/wenetspeech/ASR/pruned_transducer_stateless2/train.py b/egs/wenetspeech/ASR/pruned_transducer_stateless2/train.py new file mode 100644 index 0000000000..4db874c8d3 --- /dev/null +++ b/egs/wenetspeech/ASR/pruned_transducer_stateless2/train.py @@ -0,0 +1,1025 @@ +#!/usr/bin/env python3 +# Copyright 2021 Xiaomi Corp. (authors: Fangjun Kuang, +# Wei Kang +# Mingshuang Luo) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# 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. +""" +Usage: + +For training with the L subset: + +export CUDA_VISIBLE_DEVICES="0,1,2,3,4,5,6,7" + +./pruned_transducer_stateless2/train.py \ + --lang-dir data/lang_char \ + --exp-dir pruned_transducer_stateless2/exp \ + --world-size 8 \ + --num-epochs 15 \ + --start-epoch 0 \ + --max-duration 180 \ + --valid-interval 3000 \ + --model-warm-step 3000 \ + --save-every-n 8000 \ + --training-subset L + +# For mix precision training: + +./pruned_transducer_stateless2/train.py \ + --lang-dir data/lang_char \ + --exp-dir pruned_transducer_stateless2/exp \ + --world-size 8 \ + --num-epochs 10 \ + --start-epoch 0 \ + --max-duration 180 \ + --valid-interval 3000 \ + --model-warm-step 3000 \ + --save-every-n 8000 \ + --use-fp16 True \ + --training-subset L + +For training with the M subset: + +./pruned_transducer_stateless2/train.py \ + --lang-dir data/lang_char \ + --exp-dir pruned_transducer_stateless2/exp \ + --world-size 8 \ + --num-epochs 29 \ + --start-epoch 0 \ + --max-duration 180 \ + --valid-interval 1000 \ + --model-warm-step 500 \ + --save-every-n 1000 \ + --training-subset M + +For training with the S subset: + +./pruned_transducer_stateless2/train.py \ + --lang-dir data/lang_char \ + --exp-dir pruned_transducer_stateless2/exp \ + --world-size 8 \ + --num-epochs 29 \ + --start-epoch 0 \ + --max-duration 180 \ + --valid-interval 400 \ + --model-warm-step 100 \ + --save-every-n 1000 \ + --training-subset S +""" + +import argparse +import logging +import os +import warnings +from pathlib import Path +from shutil import copyfile +from typing import Any, Dict, Optional, Tuple, Union + +import k2 +import optim +import torch +import torch.multiprocessing as mp +import torch.nn as nn +from asr_datamodule import WenetSpeechAsrDataModule +from conformer import Conformer +from decoder import Decoder +from joiner import Joiner +from lhotse.cut import Cut +from lhotse.dataset.sampling.base import CutSampler +from lhotse.utils import fix_random_seed +from model import Transducer +from optim import Eden, Eve +from torch import Tensor +from torch.cuda.amp import GradScaler +from torch.nn.parallel import DistributedDataParallel as DDP +from torch.utils.tensorboard import SummaryWriter + +from icefall import diagnostics +from icefall.char_graph_compiler import CharCtcTrainingGraphCompiler +from icefall.checkpoint import load_checkpoint, remove_checkpoints +from icefall.checkpoint import save_checkpoint as save_checkpoint_impl +from icefall.checkpoint import save_checkpoint_with_global_batch_idx +from icefall.dist import cleanup_dist, setup_dist +from icefall.env import get_env_info +from icefall.lexicon import Lexicon +from icefall.utils import AttributeDict, MetricsTracker, setup_logger, str2bool + +LRSchedulerType = Union[ + torch.optim.lr_scheduler._LRScheduler, optim.LRScheduler +] + +os.environ["CUDA_LAUNCH_BLOCKING"] = "1" + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--world-size", + type=int, + default=1, + help="Number of GPUs for DDP training.", + ) + + parser.add_argument( + "--master-port", + type=int, + default=12354, + help="Master port to use for DDP training.", + ) + + parser.add_argument( + "--tensorboard", + type=str2bool, + default=True, + help="Should various information be logged in tensorboard.", + ) + + parser.add_argument( + "--num-epochs", + type=int, + default=30, + help="Number of epochs to train.", + ) + + parser.add_argument( + "--start-epoch", + type=int, + default=0, + help="""Resume training from from this epoch. + If it is positive, it will load checkpoint from + transducer_stateless2/exp/epoch-{start_epoch-1}.pt + """, + ) + + parser.add_argument( + "--start-batch", + type=int, + default=0, + help="""If positive, --start-epoch is ignored and + it loads the checkpoint from exp-dir/checkpoint-{start_batch}.pt + """, + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="pruned_transducer_stateless2/exp", + help="""The experiment dir. + It specifies the directory where all training related + files, e.g., checkpoints, log, etc, are saved + """, + ) + + parser.add_argument( + "--lang-dir", + type=str, + default="data/lang_char", + help="""The lang dir + It contains language related input files such as + "lexicon.txt" + """, + ) + + parser.add_argument( + "--initial-lr", + type=float, + default=0.003, + help="The initial learning rate. This value should not need to be changed.", + ) + + parser.add_argument( + "--lr-batches", + type=float, + default=5000, + help="""Number of steps that affects how rapidly the learning rate decreases. + We suggest not to change this.""", + ) + + parser.add_argument( + "--lr-epochs", + type=float, + default=6, + help="""Number of epochs that affects how rapidly the learning rate decreases. + """, + ) + + parser.add_argument( + "--context-size", + type=int, + default=2, + help="The context size in the decoder. 1 means bigram; " + "2 means tri-gram", + ) + + parser.add_argument( + "--prune-range", + type=int, + default=5, + help="The prune range for rnnt loss, it means how many symbols(context)" + "we are using to compute the loss", + ) + + parser.add_argument( + "--lm-scale", + type=float, + default=0.25, + help="The scale to smooth the loss with lm " + "(output of prediction network) part.", + ) + + parser.add_argument( + "--am-scale", + type=float, + default=0.0, + help="The scale to smooth the loss with am (output of encoder network)" + "part.", + ) + + parser.add_argument( + "--simple-loss-scale", + type=float, + default=0.5, + help="To get pruning ranges, we will calculate a simple version" + "loss(joiner is just addition), this simple loss also uses for" + "training (as a regularization item). We will scale the simple loss" + "with this parameter before adding to the final loss.", + ) + + parser.add_argument( + "--seed", + type=int, + default=42, + help="The seed for random generators intended for reproducibility", + ) + + parser.add_argument( + "--print-diagnostics", + type=str2bool, + default=False, + help="Accumulate stats on activations, print them and exit.", + ) + + parser.add_argument( + "--save-every-n", + type=int, + default=8000, + help="""Save checkpoint after processing this number of batches" + periodically. We save checkpoint to exp-dir/ whenever + params.batch_idx_train % save_every_n == 0. The checkpoint filename + has the form: f'exp-dir/checkpoint-{params.batch_idx_train}.pt' + Note: It also saves checkpoint to `exp-dir/epoch-xxx.pt` at the + end of each epoch where `xxx` is the epoch number counting from 0. + """, + ) + + parser.add_argument( + "--keep-last-k", + type=int, + default=20, + help="""Only keep this number of checkpoints on disk. + For instance, if it is 3, there are only 3 checkpoints + in the exp-dir with filenames `checkpoint-xxx.pt`. + It does not affect checkpoints with name `epoch-xxx.pt`. + """, + ) + + parser.add_argument( + "--use-fp16", + type=str2bool, + default=False, + help="Whether to use half precision training.", + ) + + parser.add_argument( + "--valid-interval", + type=int, + default=3000, + help="""When training_subset is L, set the valid_interval to 3000. + When training_subset is M, set the valid_interval to 1000. + When training_subset is S, set the valid_interval to 400. + """, + ) + + parser.add_argument( + "--model-warm-step", + type=int, + default=3000, + help="""When training_subset is L, set the model_warm_step to 3000. + When training_subset is M, set the model_warm_step to 500. + When training_subset is S, set the model_warm_step to 100. + """, + ) + + return parser + + +def get_params() -> AttributeDict: + """Return a dict containing training parameters. + All training related parameters that are not passed from the commandline + are saved in the variable `params`. + Commandline options are merged into `params` after they are parsed, so + you can also access them via `params`. + Explanation of options saved in `params`: + - best_train_loss: Best training loss so far. It is used to select + the model that has the lowest training loss. It is + updated during the training. + - best_valid_loss: Best validation loss so far. It is used to select + the model that has the lowest validation loss. It is + updated during the training. + - best_train_epoch: It is the epoch that has the best training loss. + - best_valid_epoch: It is the epoch that has the best validation loss. + - batch_idx_train: Used to writing statistics to tensorboard. It + contains number of batches trained so far across + epochs. + - log_interval: Print training loss if batch_idx % log_interval` is 0 + - reset_interval: Reset statistics if batch_idx % reset_interval is 0 + - valid_interval: Run validation if batch_idx % valid_interval is 0 + - feature_dim: The model input dim. It has to match the one used + in computing features. + - subsampling_factor: The subsampling factor for the model. + - encoder_dim: Hidden dim for multi-head attention model. + - num_decoder_layers: Number of decoder layer of transformer decoder. + - warm_step: The warm_step for Noam optimizer. + """ + params = AttributeDict( + { + "best_train_loss": float("inf"), + "best_valid_loss": float("inf"), + "best_train_epoch": -1, + "best_valid_epoch": -1, + "batch_idx_train": 10, + "log_interval": 1, + "reset_interval": 200, + # parameters for conformer + "feature_dim": 80, + "subsampling_factor": 4, + "encoder_dim": 512, + "nhead": 8, + "dim_feedforward": 2048, + "num_encoder_layers": 12, + # parameters for decoder + "decoder_dim": 512, + # parameters for joiner + "joiner_dim": 512, + # parameters for Noam + "env_info": get_env_info(), + } + ) + + return params + + +def get_encoder_model(params: AttributeDict) -> nn.Module: + # TODO: We can add an option to switch between Conformer and Transformer + encoder = Conformer( + num_features=params.feature_dim, + subsampling_factor=params.subsampling_factor, + d_model=params.encoder_dim, + nhead=params.nhead, + dim_feedforward=params.dim_feedforward, + num_encoder_layers=params.num_encoder_layers, + ) + return encoder + + +def get_decoder_model(params: AttributeDict) -> nn.Module: + decoder = Decoder( + vocab_size=params.vocab_size, + decoder_dim=params.decoder_dim, + blank_id=params.blank_id, + context_size=params.context_size, + ) + return decoder + + +def get_joiner_model(params: AttributeDict) -> nn.Module: + joiner = Joiner( + encoder_dim=params.encoder_dim, + decoder_dim=params.decoder_dim, + joiner_dim=params.joiner_dim, + vocab_size=params.vocab_size, + ) + return joiner + + +def get_transducer_model(params: AttributeDict) -> nn.Module: + encoder = get_encoder_model(params) + decoder = get_decoder_model(params) + joiner = get_joiner_model(params) + + model = Transducer( + encoder=encoder, + decoder=decoder, + joiner=joiner, + encoder_dim=params.encoder_dim, + decoder_dim=params.decoder_dim, + joiner_dim=params.joiner_dim, + vocab_size=params.vocab_size, + ) + return model + + +def load_checkpoint_if_available( + params: AttributeDict, + model: nn.Module, + optimizer: Optional[torch.optim.Optimizer] = None, + scheduler: Optional[LRSchedulerType] = None, +) -> Optional[Dict[str, Any]]: + """Load checkpoint from file. + If params.start_batch is positive, it will load the checkpoint from + `params.exp_dir/checkpoint-{params.start_batch}.pt`. Otherwise, if + params.start_epoch is positive, it will load the checkpoint from + `params.start_epoch - 1`. + Apart from loading state dict for `model` and `optimizer` it also updates + `best_train_epoch`, `best_train_loss`, `best_valid_epoch`, + and `best_valid_loss` in `params`. + Args: + params: + The return value of :func:`get_params`. + model: + The training model. + optimizer: + The optimizer that we are using. + scheduler: + The scheduler that we are using. + Returns: + Return a dict containing previously saved training info. + """ + if params.start_batch > 0: + filename = params.exp_dir / f"checkpoint-{params.start_batch}.pt" + elif params.start_epoch > 0: + filename = params.exp_dir / f"epoch-{params.start_epoch-1}.pt" + else: + return None + + assert filename.is_file(), f"{filename} does not exist!" + + saved_params = load_checkpoint( + filename, + model=model, + optimizer=optimizer, + scheduler=scheduler, + ) + + keys = [ + "best_train_epoch", + "best_valid_epoch", + "batch_idx_train", + "best_train_loss", + "best_valid_loss", + ] + for k in keys: + params[k] = saved_params[k] + + if params.start_batch > 0: + if "cur_epoch" in saved_params: + params["start_epoch"] = saved_params["cur_epoch"] + + return saved_params + + +def save_checkpoint( + params: AttributeDict, + model: nn.Module, + optimizer: Optional[torch.optim.Optimizer] = None, + scheduler: Optional[LRSchedulerType] = None, + sampler: Optional[CutSampler] = None, + scaler: Optional[GradScaler] = None, + rank: int = 0, +) -> None: + """Save model, optimizer, scheduler and training stats to file. + Args: + params: + It is returned by :func:`get_params`. + model: + The training model. + optimizer: + The optimizer used in the training. + sampler: + The sampler for the training dataset. + scaler: + The scaler used for mix precision training. + """ + if rank != 0: + return + filename = params.exp_dir / f"epoch-{params.cur_epoch}.pt" + save_checkpoint_impl( + filename=filename, + model=model, + params=params, + optimizer=optimizer, + scheduler=scheduler, + sampler=sampler, + scaler=scaler, + rank=rank, + ) + + if params.best_train_epoch == params.cur_epoch: + best_train_filename = params.exp_dir / "best-train-loss.pt" + copyfile(src=filename, dst=best_train_filename) + + if params.best_valid_epoch == params.cur_epoch: + best_valid_filename = params.exp_dir / "best-valid-loss.pt" + copyfile(src=filename, dst=best_valid_filename) + + +def compute_loss( + params: AttributeDict, + model: nn.Module, + graph_compiler: CharCtcTrainingGraphCompiler, + batch: dict, + is_training: bool, + warmup: float = 1.0, +) -> Tuple[Tensor, MetricsTracker]: + """ + Compute CTC loss given the model and its inputs. + Args: + params: + Parameters for training. See :func:`get_params`. + model: + The model for training. It is an instance of Conformer in our case. + batch: + A batch of data. See `lhotse.dataset.K2SpeechRecognitionDataset()` + for the content in it. + is_training: + True for training. False for validation. When it is True, this + function enables autograd during computation; when it is False, it + disables autograd. + warmup: a floating point value which increases throughout training; + values >= 1.0 are fully warmed up and have all modules present. + """ + device = model.device + feature = batch["inputs"] + # at entry, feature is (N, T, C) + assert feature.ndim == 3 + feature = feature.to(device) + + supervisions = batch["supervisions"] + feature_lens = supervisions["num_frames"].to(device) + + texts = batch["supervisions"]["text"] + + y = graph_compiler.texts_to_ids(texts) + if type(y) == list: + y = k2.RaggedTensor(y).to(device) + else: + y = y.to(device) + + with torch.set_grad_enabled(is_training): + simple_loss, pruned_loss = model( + x=feature, + x_lens=feature_lens, + y=y, + prune_range=params.prune_range, + am_scale=params.am_scale, + lm_scale=params.lm_scale, + warmup=warmup, + ) + # after the main warmup step, we keep pruned_loss_scale small + # for the same amount of time (model_warm_step), to avoid + # overwhelming the simple_loss and causing it to diverge, + # in case it had not fully learned the alignment yet. + pruned_loss_scale = ( + 0.0 + if warmup < 1.0 + else (0.1 if warmup > 1.0 and warmup < 2.0 else 1.0) + ) + loss = ( + params.simple_loss_scale * simple_loss + + pruned_loss_scale * pruned_loss + ) + assert loss.requires_grad == is_training + + info = MetricsTracker() + with warnings.catch_warnings(): + warnings.simplefilter("ignore") + info["frames"] = ( + (feature_lens // params.subsampling_factor).sum().item() + ) + + # Note: We use reduction=sum while computing the loss. + info["loss"] = loss.detach().cpu().item() + info["simple_loss"] = simple_loss.detach().cpu().item() + info["pruned_loss"] = pruned_loss.detach().cpu().item() + + return loss, info + + +def compute_validation_loss( + params: AttributeDict, + model: nn.Module, + graph_compiler: CharCtcTrainingGraphCompiler, + valid_dl: torch.utils.data.DataLoader, + world_size: int = 1, +) -> MetricsTracker: + """Run the validation process.""" + model.eval() + + tot_loss = MetricsTracker() + + for batch_idx, batch in enumerate(valid_dl): + loss, loss_info = compute_loss( + params=params, + model=model, + graph_compiler=graph_compiler, + batch=batch, + is_training=False, + ) + assert loss.requires_grad is False + tot_loss = tot_loss + loss_info + + if world_size > 1: + tot_loss.reduce(loss.device) + + loss_value = tot_loss["loss"] / tot_loss["frames"] + if loss_value < params.best_valid_loss: + params.best_valid_epoch = params.cur_epoch + params.best_valid_loss = loss_value + + return tot_loss + + +def train_one_epoch( + params: AttributeDict, + model: nn.Module, + optimizer: torch.optim.Optimizer, + scheduler: LRSchedulerType, + graph_compiler: CharCtcTrainingGraphCompiler, + train_dl: torch.utils.data.DataLoader, + valid_dl: torch.utils.data.DataLoader, + scaler: GradScaler, + tb_writer: Optional[SummaryWriter] = None, + world_size: int = 1, + rank: int = 0, +) -> None: + """Train the model for one epoch. + The training loss from the mean of all frames is saved in + `params.train_loss`. It runs the validation process every + `params.valid_interval` batches. + Args: + params: + It is returned by :func:`get_params`. + model: + The model for training. + optimizer: + The optimizer we are using. + scheduler: + The learning rate scheduler, we call step() every step. + train_dl: + Dataloader for the training dataset. + valid_dl: + Dataloader for the validation dataset. + scaler: + The scaler used for mix precision training. + tb_writer: + Writer to write log messages to tensorboard. + world_size: + Number of nodes in DDP training. If it is 1, DDP is disabled. + rank: + The rank of the node in DDP training. If no DDP is used, it should + be set to 0. + """ + model.train() + + tot_loss = MetricsTracker() + + for batch_idx, batch in enumerate(train_dl): + + params.batch_idx_train += 1 + batch_size = len(batch["supervisions"]["text"]) + + with torch.cuda.amp.autocast(enabled=params.use_fp16): + loss, loss_info = compute_loss( + params=params, + model=model, + graph_compiler=graph_compiler, + batch=batch, + is_training=True, + warmup=(params.batch_idx_train / params.model_warm_step), + ) + # summary stats + tot_loss = (tot_loss * (1 - 1 / params.reset_interval)) + loss_info + + # NOTE: We use reduction==sum and loss is computed over utterances + # in the batch and there is no normalization to it so far. + scaler.scale(loss).backward() + scheduler.step_batch(params.batch_idx_train) + scaler.step(optimizer) + scaler.update() + optimizer.zero_grad() + + if params.print_diagnostics and batch_idx == 5: + return + + if ( + params.batch_idx_train > 0 + and params.batch_idx_train % params.save_every_n == 0 + ): + save_checkpoint_with_global_batch_idx( + out_dir=params.exp_dir, + global_batch_idx=params.batch_idx_train, + model=model, + params=params, + optimizer=optimizer, + scheduler=scheduler, + sampler=train_dl.sampler, + scaler=scaler, + rank=rank, + ) + remove_checkpoints( + out_dir=params.exp_dir, + topk=params.keep_last_k, + rank=rank, + ) + + if batch_idx % params.log_interval == 0: + cur_lr = scheduler.get_last_lr()[0] + logging.info( + f"Epoch {params.cur_epoch}, " + f"batch {batch_idx}, loss[{loss_info}], " + f"tot_loss[{tot_loss}], batch size: {batch_size}, " + f"lr: {cur_lr:.2e}" + ) + + if tb_writer is not None: + tb_writer.add_scalar( + "train/learning_rate", cur_lr, params.batch_idx_train + ) + + loss_info.write_summary( + tb_writer, "train/current_", params.batch_idx_train + ) + tot_loss.write_summary( + tb_writer, "train/tot_", params.batch_idx_train + ) + + if batch_idx > 0 and batch_idx % params.valid_interval == 0: + logging.info("Computing validation loss") + valid_info = compute_validation_loss( + params=params, + model=model, + graph_compiler=graph_compiler, + valid_dl=valid_dl, + world_size=world_size, + ) + model.train() + logging.info(f"Epoch {params.cur_epoch}, validation: {valid_info}") + if tb_writer is not None: + valid_info.write_summary( + tb_writer, "train/valid_", params.batch_idx_train + ) + + loss_value = tot_loss["loss"] / tot_loss["frames"] + params.train_loss = loss_value + if params.train_loss < params.best_train_loss: + params.best_train_epoch = params.cur_epoch + params.best_train_loss = params.train_loss + + +def run(rank, world_size, args): + """ + Args: + rank: + It is a value between 0 and `world_size-1`, which is + passed automatically by `mp.spawn()` in :func:`main`. + The node with rank 0 is responsible for saving checkpoint. + world_size: + Number of GPUs for DDP training. + args: + The return value of get_parser().parse_args() + """ + params = get_params() + params.update(vars(args)) + + fix_random_seed(params.seed) + if world_size > 1: + setup_dist(rank, world_size, params.master_port) + + setup_logger(f"{params.exp_dir}/log/log-train") + logging.info("Training started") + + if args.tensorboard and rank == 0: + tb_writer = SummaryWriter(log_dir=f"{params.exp_dir}/tensorboard") + else: + tb_writer = None + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", rank) + logging.info(f"Device: {device}") + + lexicon = Lexicon(params.lang_dir) + graph_compiler = CharCtcTrainingGraphCompiler( + lexicon=lexicon, + device=device, + ) + + params.blank_id = lexicon.token_table[""] + params.vocab_size = max(lexicon.tokens) + 1 + + logging.info(params) + + logging.info("About to create model") + model = get_transducer_model(params) + + num_param = sum([p.numel() for p in model.parameters()]) + logging.info(f"Number of model parameters: {num_param}") + + checkpoints = load_checkpoint_if_available(params=params, model=model) + + model.to(device) + if world_size > 1: + logging.info("Using DDP") + model = DDP(model, device_ids=[rank]) + model.device = device + + optimizer = Eve(model.parameters(), lr=params.initial_lr) + + scheduler = Eden(optimizer, params.lr_batches, params.lr_epochs) + + if checkpoints and "optimizer" in checkpoints: + logging.info("Loading optimizer state dict") + optimizer.load_state_dict(checkpoints["optimizer"]) + + if ( + checkpoints + and "scheduler" in checkpoints + and checkpoints["scheduler"] is not None + ): + logging.info("Loading scheduler state dict") + scheduler.load_state_dict(checkpoints["scheduler"]) + + if params.print_diagnostics: + opts = diagnostics.TensorDiagnosticOptions( + 2 ** 22 + ) # allow 4 megabytes per sub-module + diagnostic = diagnostics.attach_diagnostics(model, opts) + + wenetspeech = WenetSpeechAsrDataModule(args) + + train_cuts = wenetspeech.train_cuts() + valid_cuts = wenetspeech.valid_cuts() + + def remove_short_and_long_utt(c: Cut): + # Keep only utterances with duration between 1 second and 15.0 seconds + # + # Caution: There is a reason to select 15.0 here. Please see + # ../local/display_manifest_statistics.py + # + # You should use ../local/display_manifest_statistics.py to get + # an utterance duration distribution for your dataset to select + # the threshold + return 1.0 <= c.duration <= 15.0 + + train_cuts = train_cuts.filter(remove_short_and_long_utt) + + valid_dl = wenetspeech.valid_dataloaders(valid_cuts) + + if params.start_batch > 0 and checkpoints and "sampler" in checkpoints: + # We only load the sampler's state dict when it loads a checkpoint + # saved in the middle of an epoch + sampler_state_dict = checkpoints["sampler"] + else: + sampler_state_dict = None + + train_dl = wenetspeech.train_dataloaders( + train_cuts, sampler_state_dict=sampler_state_dict + ) + + if not params.print_diagnostics and params.start_batch == 0: + scan_pessimistic_batches_for_oom( + model=model, + train_dl=train_dl, + optimizer=optimizer, + graph_compiler=graph_compiler, + params=params, + ) + + scaler = GradScaler(enabled=params.use_fp16) + if checkpoints and "grad_scaler" in checkpoints: + logging.info("Loading grad scaler state dict") + scaler.load_state_dict(checkpoints["grad_scaler"]) + + for epoch in range(params.start_epoch, params.num_epochs): + scheduler.step_epoch(epoch) + fix_random_seed(params.seed + epoch) + train_dl.sampler.set_epoch(epoch) + + if tb_writer is not None: + tb_writer.add_scalar("train/epoch", epoch, params.batch_idx_train) + + params.cur_epoch = epoch + + train_one_epoch( + params=params, + model=model, + optimizer=optimizer, + scheduler=scheduler, + graph_compiler=graph_compiler, + train_dl=train_dl, + valid_dl=valid_dl, + scaler=scaler, + tb_writer=tb_writer, + world_size=world_size, + rank=rank, + ) + + if params.print_diagnostics: + diagnostic.print_diagnostics() + break + + save_checkpoint( + params=params, + model=model, + optimizer=optimizer, + scheduler=scheduler, + sampler=train_dl.sampler, + scaler=scaler, + rank=rank, + ) + + logging.info("Done!") + + if world_size > 1: + torch.distributed.barrier() + cleanup_dist() + + +def scan_pessimistic_batches_for_oom( + model: nn.Module, + train_dl: torch.utils.data.DataLoader, + optimizer: torch.optim.Optimizer, + graph_compiler: CharCtcTrainingGraphCompiler, + params: AttributeDict, +): + from lhotse.dataset import find_pessimistic_batches + + logging.info( + "Sanity check -- see if any of the batches in epoch 0 would cause OOM." + ) + batches, crit_values = find_pessimistic_batches(train_dl.sampler) + for criterion, cuts in batches.items(): + batch = train_dl.dataset[cuts] + try: + # warmup = 0.0 is so that the derivs for the pruned loss stay zero + # (i.e. are not remembered by the decaying-average in adam), because + # we want to avoid these params being subject to shrinkage in adam. + with torch.cuda.amp.autocast(enabled=params.use_fp16): + loss, _ = compute_loss( + params=params, + model=model, + graph_compiler=graph_compiler, + batch=batch, + is_training=True, + warmup=0.0, + ) + loss.backward() + optimizer.step() + optimizer.zero_grad() + except RuntimeError as e: + if "CUDA out of memory" in str(e): + logging.error( + "Your GPU ran out of memory with the current " + "max_duration setting. We recommend decreasing " + "max_duration and trying again.\n" + f"Failing criterion: {criterion} " + f"(={crit_values[criterion]}) ..." + ) + raise + + +def main(): + parser = get_parser() + WenetSpeechAsrDataModule.add_arguments(parser) + args = parser.parse_args() + args.lang_dir = Path(args.lang_dir) + args.exp_dir = Path(args.exp_dir) + + world_size = args.world_size + assert world_size >= 1 + if world_size > 1: + mp.spawn(run, args=(world_size, args), nprocs=world_size, join=True) + else: + run(rank=0, world_size=1, args=args) + + +torch.set_num_threads(1) +torch.set_num_interop_threads(1) + +if __name__ == "__main__": + main() diff --git a/egs/wenetspeech/ASR/shared b/egs/wenetspeech/ASR/shared new file mode 120000 index 0000000000..e9461a6d7e --- /dev/null +++ b/egs/wenetspeech/ASR/shared @@ -0,0 +1 @@ +../../librispeech/ASR/shared \ No newline at end of file