diff --git a/.github/workflows/run-pretrained-transducer-stateless-librispeech-100h.yml b/.github/workflows/run-pretrained-transducer-stateless-librispeech-100h.yml new file mode 100644 index 000000000..efea5366b --- /dev/null +++ b/.github/workflows/run-pretrained-transducer-stateless-librispeech-100h.yml @@ -0,0 +1,152 @@ +# Copyright 2021 Fangjun Kuang (csukuangfj@gmail.com) + +# 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. + +name: run-pre-trained-trandsucer-stateless-multi-datasets-librispeech-100h + +on: + push: + branches: + - master + pull_request: + types: [labeled] + +jobs: + run_pre_trained_transducer_stateless_multi_datasets_librispeech_100h: + if: github.event.label.name == 'ready' || github.event_name == 'push' + runs-on: ${{ matrix.os }} + strategy: + matrix: + os: [ubuntu-18.04] + python-version: [3.7, 3.8, 3.9] + torch: ["1.10.0"] + torchaudio: ["0.10.0"] + k2-version: ["1.9.dev20211101"] + + fail-fast: false + + steps: + - uses: actions/checkout@v2 + with: + fetch-depth: 0 + + - name: Setup Python ${{ matrix.python-version }} + uses: actions/setup-python@v1 + with: + python-version: ${{ matrix.python-version }} + + - name: Install Python dependencies + run: | + python3 -m pip install --upgrade pip pytest + # numpy 1.20.x does not support python 3.6 + pip install numpy==1.19 + pip install torch==${{ matrix.torch }}+cpu torchaudio==${{ matrix.torchaudio }}+cpu -f https://download.pytorch.org/whl/cpu/torch_stable.html + pip install k2==${{ matrix.k2-version }}+cpu.torch${{ matrix.torch }} -f https://k2-fsa.org/nightly/ + + python3 -m pip install git+https://github.com/lhotse-speech/lhotse + python3 -m pip install kaldifeat + # We are in ./icefall and there is a file: requirements.txt in it + pip install -r requirements.txt + + - name: Install graphviz + shell: bash + run: | + python3 -m pip install -qq graphviz + sudo apt-get -qq install graphviz + + - name: Download pre-trained model + shell: bash + run: | + sudo apt-get -qq install git-lfs tree sox + cd egs/librispeech/ASR + mkdir tmp + cd tmp + git lfs install + git clone https://huggingface.co/csukuangfj/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21 + + cd .. + tree tmp + soxi tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/*.wav + ls -lh tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/*.wav + + - name: Run greedy search decoding (max-sym-per-frame 1) + shell: bash + run: | + export PYTHONPATH=$PWD:PYTHONPATH + cd egs/librispeech/ASR + ./transducer_stateless_multi_datasets/pretrained.py \ + --method greedy_search \ + --max-sym-per-frame 1 \ + --checkpoint ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/exp/pretrained.pt \ + --bpe-model ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/data/lang_bpe_500/bpe.model \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1089-134686-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0002.wav + + - name: Run greedy search decoding (max-sym-per-frame 2) + shell: bash + run: | + export PYTHONPATH=$PWD:PYTHONPATH + cd egs/librispeech/ASR + ./transducer_stateless_multi_datasets/pretrained.py \ + --method greedy_search \ + --max-sym-per-frame 2 \ + --checkpoint ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/exp/pretrained.pt \ + --bpe-model ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/data/lang_bpe_500/bpe.model \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1089-134686-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0002.wav + + - name: Run greedy search decoding (max-sym-per-frame 3) + shell: bash + run: | + export PYTHONPATH=$PWD:PYTHONPATH + cd egs/librispeech/ASR + ./transducer_stateless_multi_datasets/pretrained.py \ + --method greedy_search \ + --max-sym-per-frame 3 \ + --checkpoint ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/exp/pretrained.pt \ + --bpe-model ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/data/lang_bpe_500/bpe.model \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1089-134686-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0002.wav + + - name: Run beam search decoding + shell: bash + run: | + export PYTHONPATH=$PWD:$PYTHONPATH + cd egs/librispeech/ASR + ./transducer_stateless_multi_datasets/pretrained.py \ + --method beam_search \ + --beam-size 4 \ + --checkpoint ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/exp/pretrained.pt \ + --bpe-model ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/data/lang_bpe_500/bpe.model \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1089-134686-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0002.wav + + - name: Run modified beam search decoding + shell: bash + run: | + export PYTHONPATH=$PWD:$PYTHONPATH + cd egs/librispeech/ASR + ./transducer_stateless_multi_datasets/pretrained.py \ + --method modified_beam_search \ + --beam-size 4 \ + --checkpoint ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/exp/pretrained.pt \ + --bpe-model ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/data/lang_bpe_500/bpe.model \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1089-134686-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0001.wav \ + ./tmp/icefall-asr-librispeech-100h-transducer-stateless-multi-datasets-bpe-500-2022-02-21/test_wavs/1221-135766-0002.wav diff --git a/egs/librispeech/ASR/README.md b/egs/librispeech/ASR/README.md index c8ee98d7d..211a7d120 100644 --- a/egs/librispeech/ASR/README.md +++ b/egs/librispeech/ASR/README.md @@ -9,11 +9,12 @@ for how to run models in this recipe. There are various folders containing the name `transducer` in this folder. The following table lists the differences among them. -| | Encoder | Decoder | -|------------------------|-----------|--------------------| -| `transducer` | Conformer | LSTM | -| `transducer_stateless` | Conformer | Embedding + Conv1d | -| `transducer_lstm ` | LSTM | LSTM | +| | Encoder | Decoder | Comment | +|---------------------------------------|-----------|--------------------|---------------------------------------------------| +| `transducer` | Conformer | LSTM | | +| `transducer_stateless` | Conformer | Embedding + Conv1d | | +| `transducer_lstm` | LSTM | LSTM | | +| `transducer_stateless_multi_datasets` | Conformer | Embedding + Conv1d | Using data from GigaSpeech as extra training data | The decoder in `transducer_stateless` is modified from the paper [Rnn-Transducer with Stateless Prediction Network](https://ieeexplore.ieee.org/document/9054419/). diff --git a/egs/librispeech/ASR/RESULTS-100hours.md b/egs/librispeech/ASR/RESULTS-100hours.md new file mode 100644 index 000000000..40245c917 --- /dev/null +++ b/egs/librispeech/ASR/RESULTS-100hours.md @@ -0,0 +1,75 @@ +# Results for train-clean-100 + +This page shows the WERs for test-clean/test-other using only +train-clean-100 subset as training data. + +## Conformer encoder + embedding decoder + +### 2022-02-21 + +| | test-clean | test-other | comment | +|-------------------------------------|------------|------------|------------------------------------------| +| greedy search (max sym per frame 1) | 6.34 | 16.7 | --epoch 57, --avg 17, --max-duration 100 | +| greedy search (max sym per frame 2) | 6.34 | 16.7 | --epoch 57, --avg 17, --max-duration 100 | +| greedy search (max sym per frame 3) | 6.34 | 16.7 | --epoch 57, --avg 17, --max-duration 100 | +| modified beam search (beam size 4) | 6.31 | 16.3 | --epoch 57, --avg 17, --max-duration 100 | + + +The training command for reproducing is given below: + +```bash +cd egs/librispeech/ASR/ +./prepare.sh +./prepare_giga_speech.sh + +export CUDA_VISIBLE_DEVICES="0,1" + +./transducer_stateless_multi_datasets/train.py \ + --world-size 2 \ + --num-epochs 60 \ + --start-epoch 0 \ + --exp-dir transducer_stateless_multi_datasets/exp-100-2 \ + --full-libri 0 \ + --max-duration 300 \ + --lr-factor 1 \ + --bpe-model data/lang_bpe_500/bpe.model \ + --modified-transducer-prob 0.25 + --giga-prob 0.2 +``` + +The decoding command is given below: + +```bash +for epoch in 57; do + for avg in 17; do + for sym in 1 2 3; do + ./transducer_stateless_multi_datasets/decode.py \ + --epoch $epoch \ + --avg $avg \ + --exp-dir transducer_stateless_multi_datasets/exp-100-2 \ + --bpe-model ./data/lang_bpe_500/bpe.model \ + --max-duration 100 \ + --context-size 2 \ + --max-sym-per-frame $sym + done + done +done + +epoch=57 +avg=17 +./transducer_stateless_multi_datasets/decode.py \ + --epoch $epoch \ + --avg $avg \ + --exp-dir transducer_stateless_multi_datasets/exp-100-2 \ + --bpe-model ./data/lang_bpe_500/bpe.model \ + --max-duration 100 \ + --context-size 2 \ + --decoding-method modified_beam_search \ + --beam-size 4 +``` + +The tensorboard log is available at + + +A pre-trained model and decoding logs can be found at + diff --git a/egs/librispeech/ASR/local/compute_fbank_librispeech.py b/egs/librispeech/ASR/local/compute_fbank_librispeech.py index b26034eb2..5c33ff8be 100755 --- a/egs/librispeech/ASR/local/compute_fbank_librispeech.py +++ b/egs/librispeech/ASR/local/compute_fbank_librispeech.py @@ -28,7 +28,7 @@ import os from pathlib import Path import torch -from lhotse import CutSet, Fbank, FbankConfig, LilcomHdf5Writer +from lhotse import ChunkedLilcomHdf5Writer, CutSet, Fbank, FbankConfig from lhotse.recipes.utils import read_manifests_if_cached from icefall.utils import get_executor @@ -85,7 +85,7 @@ def compute_fbank_librispeech(): # when an executor is specified, make more partitions num_jobs=num_jobs if ex is None else 80, executor=ex, - storage_type=LilcomHdf5Writer, + storage_type=ChunkedLilcomHdf5Writer, ) cut_set.to_json(output_dir / f"cuts_{partition}.json.gz") diff --git a/egs/librispeech/ASR/local/compute_fbank_musan.py b/egs/librispeech/ASR/local/compute_fbank_musan.py index d44524e70..f5911746b 100755 --- a/egs/librispeech/ASR/local/compute_fbank_musan.py +++ b/egs/librispeech/ASR/local/compute_fbank_musan.py @@ -28,7 +28,7 @@ import os from pathlib import Path import torch -from lhotse import CutSet, Fbank, FbankConfig, LilcomHdf5Writer, combine +from lhotse import ChunkedLilcomHdf5Writer, CutSet, Fbank, FbankConfig, combine from lhotse.recipes.utils import read_manifests_if_cached from icefall.utils import get_executor @@ -82,7 +82,7 @@ def compute_fbank_musan(): storage_path=f"{output_dir}/feats_musan", num_jobs=num_jobs if ex is None else 80, executor=ex, - storage_type=LilcomHdf5Writer, + storage_type=ChunkedLilcomHdf5Writer, ) ) musan_cuts.to_json(musan_cuts_path) diff --git a/egs/librispeech/ASR/local/preprocess_gigaspeech.py b/egs/librispeech/ASR/local/preprocess_gigaspeech.py new file mode 100644 index 000000000..4168a7185 --- /dev/null +++ b/egs/librispeech/ASR/local/preprocess_gigaspeech.py @@ -0,0 +1,123 @@ +#!/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/GigaSpeech/blob/main/toolkits/kaldi/gigaspeech_data_prep.sh + + +def normalize_text( + utt: str, + punct_pattern=re.compile(r"<(COMMA|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_giga_speech(): + src_dir = Path("data/manifests") + output_dir = Path("data/fbank") + output_dir.mkdir(exist_ok=True) + + dataset_parts = ( + "DEV", + "TEST", + "XS", + "S", + "M", + "L", + "XL", + ) + + logging.info("Loading manifest (may take 4 minutes)") + manifests = read_manifests_if_cached( + dataset_parts=dataset_parts, + output_dir=src_dir, + prefix="gigaspeech", + 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"]: + sup.text = normalize_text(sup.text) + sup.custom = {"origin": "giga"} + + # 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"]: + 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("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 {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_giga_speech() + + +if __name__ == "__main__": + main() diff --git a/egs/librispeech/ASR/prepare_giga_speech.sh b/egs/librispeech/ASR/prepare_giga_speech.sh new file mode 100755 index 000000000..49124c4d7 --- /dev/null +++ b/egs/librispeech/ASR/prepare_giga_speech.sh @@ -0,0 +1,109 @@ +#!/usr/bin/env bash + +set -eou pipefail + +nj=15 +stage=-1 +stop_stage=100 + +# We assume dl_dir (download dir) contains the following +# directories and files. If not, they will be downloaded +# by this script automatically. +# +# - $dl_dir/GigaSpeech +# You can find audio, dict, GigaSpeech.json inside it. +# You can apply for the download credentials by following +# https://github.com/SpeechColab/GigaSpeech#download + +# Number of hours for GigaSpeech subsets +# XL 10k hours +# L 2.5k hours +# M 1k hours +# S 250 hours +# XS 10 hours +# DEV 12 hours +# Test 40 hours + +dl_dir=$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/GigaSpeech ] && mkdir -p $dl_dir/GigaSpeech + + # If you have pre-downloaded it to /path/to/GigaSpeech, + # you can create a symlink + # + # ln -sfv /path/to/GigaSpeech $dl_dir/GigaSpeech + # + if [ ! -d $dl_dir/GigaSpeech/audio ] && [ ! -f $dl_dir/GigaSpeech.json ]; then + # Check credentials. + if [ ! -f $dl_dir/password ]; then + echo -n "$0: Please apply for the download credentials by following" + echo -n "https://github.com/SpeechColab/GigaSpeech#dataset-download" + echo " and save it to $dl_dir/password." + exit 1; + fi + PASSWORD=`cat $dl_dir/password 2>/dev/null` + if [ -z "$PASSWORD" ]; then + echo "$0: Error, $dl_dir/password is empty." + exit 1; + fi + PASSWORD_MD5=`echo $PASSWORD | md5sum | cut -d ' ' -f 1` + if [[ $PASSWORD_MD5 != "dfbf0cde1a3ce23749d8d81e492741b8" ]]; then + echo "$0: Error, invalid $dl_dir/password." + exit 1; + fi + # Download XL, DEV and TEST sets by default. + lhotse download gigaspeech \ + --subset XL \ + --subset L \ + --subset M \ + --subset S \ + --subset XS \ + --subset DEV \ + --subset TEST \ + --host tsinghua \ + $dl_dir/password $dl_dir/GigaSpeech + fi +fi + +if [ $stage -le 1 ] && [ $stop_stage -ge 1 ]; then + log "Stage 1: Prepare GigaSpeech manifest (may take 30 minutes)" + # We assume that you have downloaded the GigaSpeech corpus + # to $dl_dir/GigaSpeech + mkdir -p data/manifests + lhotse prepare gigaspeech \ + --subset XL \ + --subset L \ + --subset M \ + --subset S \ + --subset XS \ + --subset DEV \ + --subset TEST \ + -j $nj \ + $dl_dir/GigaSpeech data/manifests +fi + +if [ $stage -le 2 ] && [ $stop_stage -ge 2 ]; then + log "Stage 2: Preprocess GigaSpeech manifest" + if [ ! -f data/fbank/.preprocess_complete ]; then + log "It may take 2 hours for this stage" + python3 ./local/preprocess_gigaspeech.py + touch data/fbank/.preprocess_complete + fi +fi diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/README.md b/egs/librispeech/ASR/transducer_stateless_multi_datasets/README.md new file mode 100644 index 000000000..574fbf78e --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/README.md @@ -0,0 +1,27 @@ +## Introduction + +The decoder, i.e., the prediction network, is from +https://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=9054419 +(Rnn-Transducer with Stateless Prediction Network) + +You can use the following command to start the training: + +```bash +cd egs/librispeech/ASR +./prepare.sh +./prepare_giga_speech.sh + +export CUDA_VISIBLE_DEVICES="0,1" + +./transducer_stateless_multi_datasets/train.py \ + --world-size 2 \ + --num-epochs 60 \ + --start-epoch 0 \ + --exp-dir transducer_stateless_multi_datasets/exp-100 \ + --full-libri 0 \ + --max-duration 300 \ + --lr-factor 1 \ + --bpe-model data/lang_bpe_500/bpe.model \ + --modified-transducer-prob 0.25 + --giga-prob 0.2 +``` diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/__init__.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/__init__.py new file mode 100644 index 000000000..e69de29bb diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/asr_datamodule.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/asr_datamodule.py new file mode 100644 index 000000000..fe0d0a872 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/asr_datamodule.py @@ -0,0 +1,304 @@ +# Copyright 2021 Piotr Żelasko +# 2022 Xiaomi Corp. (authors: 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 pathlib import Path +from typing import Optional + +from lhotse import CutSet, Fbank, FbankConfig +from lhotse.dataset import ( + BucketingSampler, + CutMix, + DynamicBucketingSampler, + K2SpeechRecognitionDataset, + SpecAugment, +) +from lhotse.dataset.input_strategies import ( + OnTheFlyFeatures, + PrecomputedFeatures, +) +from torch.utils.data import DataLoader + +from icefall.utils import str2bool + + +class AsrDataModule: + 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( + "--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=30, + help="The number of buckets for the BucketingSampler " + "and DynamicBucketingSampler." + "(you might want to increase it for larger datasets).", + ) + + 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( + "--manifest-dir", + type=Path, + default=Path("data/fbank"), + help="Path to directory with train/valid/test cuts.", + ) + + 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. Used only in dev/test CutSet", + ) + + def train_dataloaders( + self, + cuts_train: CutSet, + dynamic_bucketing: bool, + on_the_fly_feats: bool, + cuts_musan: Optional[CutSet] = None, + ) -> DataLoader: + """ + Args: + cuts_train: + Cuts for training. + cuts_musan: + If not None, it is the cuts for mixing. + dynamic_bucketing: + True to use DynamicBucketingSampler; + False to use BucketingSampler. + on_the_fly_feats: + True to use OnTheFlyFeatures; + False to use PrecomputedFeatures. + """ + transforms = [] + if cuts_musan is not None: + logging.info("Enable MUSAN") + transforms.append( + CutMix( + cuts=cuts_musan, prob=0.5, snr=(10, 20), preserve_id=True + ) + ) + else: + logging.info("Disable MUSAN") + + 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}" + ) + input_transforms.append( + SpecAugment( + time_warp_factor=self.args.spec_aug_time_warp_factor, + num_frame_masks=2, + 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, + ) + + # 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))) + if on_the_fly_feats + else PrecomputedFeatures() + ), + input_transforms=input_transforms, + return_cuts=self.args.return_cuts, + ) + + if dynamic_bucketing: + 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, + drop_last=True, + ) + else: + logging.info("Using BucketingSampler.") + train_sampler = BucketingSampler( + cuts_train, + max_duration=self.args.max_duration, + shuffle=self.args.shuffle, + num_buckets=self.args.num_buckets, + bucket_method="equal_duration", + drop_last=True, + ) + + logging.info("About to create train dataloader") + train_dl = DataLoader( + train, + sampler=train_sampler, + batch_size=None, + num_workers=self.args.num_workers, + persistent_workers=False, + ) + return train_dl + + def valid_dataloaders(self, cuts_valid: CutSet) -> DataLoader: + 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 = BucketingSampler( + cuts_valid, + max_duration=self.args.max_duration, + shuffle=False, + ) + logging.info("About to create dev dataloader") + valid_dl = DataLoader( + validate, + sampler=valid_sampler, + batch_size=None, + num_workers=2, + 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 = BucketingSampler( + cuts, max_duration=self.args.max_duration, shuffle=False + ) + logging.debug("About to create test dataloader") + test_dl = DataLoader( + test, + batch_size=None, + sampler=sampler, + num_workers=self.args.num_workers, + ) + return test_dl diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/beam_search.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/beam_search.py new file mode 100644 index 000000000..c5efb733d --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/beam_search.py @@ -0,0 +1,541 @@ +# Copyright 2021 Xiaomi Corp. (authors: 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. + +from dataclasses import dataclass +from typing import Dict, List, Optional + +import torch +from model import Transducer + + +def greedy_search( + model: Transducer, encoder_out: torch.Tensor, max_sym_per_frame: int +) -> List[int]: + """ + Args: + model: + An instance of `Transducer`. + encoder_out: + A tensor of shape (N, T, C) from the encoder. Support only N==1 for now. + max_sym_per_frame: + Maximum number of symbols per frame. If it is set to 0, the WER + would be 100%. + Returns: + Return the decoded result. + """ + assert encoder_out.ndim == 3 + + # support only batch_size == 1 for now + assert encoder_out.size(0) == 1, encoder_out.size(0) + + blank_id = model.decoder.blank_id + context_size = model.decoder.context_size + + device = model.device + + decoder_input = torch.tensor( + [blank_id] * context_size, device=device, dtype=torch.int64 + ).reshape(1, context_size) + + decoder_out = model.decoder(decoder_input, need_pad=False) + + T = encoder_out.size(1) + t = 0 + hyp = [blank_id] * context_size + + # Maximum symbols per utterance. + max_sym_per_utt = 1000 + + # symbols per frame + sym_per_frame = 0 + + # symbols per utterance decoded so far + sym_per_utt = 0 + + encoder_out_len = torch.tensor([1]) + decoder_out_len = torch.tensor([1]) + + while t < T and sym_per_utt < max_sym_per_utt: + if sym_per_frame >= max_sym_per_frame: + sym_per_frame = 0 + t += 1 + continue + + # fmt: off + current_encoder_out = encoder_out[:, t:t+1, :] + # fmt: on + logits = model.joiner( + current_encoder_out, decoder_out, encoder_out_len, decoder_out_len + ) + # logits is (1, 1, 1, vocab_size) + + y = logits.argmax().item() + if y != blank_id: + hyp.append(y) + decoder_input = torch.tensor( + [hyp[-context_size:]], device=device + ).reshape(1, context_size) + + decoder_out = model.decoder(decoder_input, need_pad=False) + + sym_per_utt += 1 + sym_per_frame += 1 + else: + sym_per_frame = 0 + t += 1 + hyp = hyp[context_size:] # remove blanks + + return hyp + + +@dataclass +class Hypothesis: + # The predicted tokens so far. + # Newly predicted tokens are appended to `ys`. + ys: List[int] + + # The log prob of ys. + # It contains only one entry. + log_prob: torch.Tensor + + @property + def key(self) -> str: + """Return a string representation of self.ys""" + return "_".join(map(str, self.ys)) + + +class HypothesisList(object): + def __init__(self, data: Optional[Dict[str, Hypothesis]] = None) -> None: + """ + Args: + data: + A dict of Hypotheses. Its key is its `value.key`. + """ + if data is None: + self._data = {} + else: + self._data = data + + @property + def data(self) -> Dict[str, Hypothesis]: + return self._data + + def add(self, hyp: Hypothesis) -> None: + """Add a Hypothesis to `self`. + + If `hyp` already exists in `self`, its probability is updated using + `log-sum-exp` with the existed one. + + Args: + hyp: + The hypothesis to be added. + """ + key = hyp.key + if key in self: + old_hyp = self._data[key] # shallow copy + torch.logaddexp( + old_hyp.log_prob, hyp.log_prob, out=old_hyp.log_prob + ) + else: + self._data[key] = hyp + + def get_most_probable(self, length_norm: bool = False) -> Hypothesis: + """Get the most probable hypothesis, i.e., the one with + the largest `log_prob`. + + Args: + length_norm: + If True, the `log_prob` of a hypothesis is normalized by the + number of tokens in it. + Returns: + Return the hypothesis that has the largest `log_prob`. + """ + if length_norm: + return max( + self._data.values(), key=lambda hyp: hyp.log_prob / len(hyp.ys) + ) + else: + return max(self._data.values(), key=lambda hyp: hyp.log_prob) + + def remove(self, hyp: Hypothesis) -> None: + """Remove a given hypothesis. + + Caution: + `self` is modified **in-place**. + + Args: + hyp: + The hypothesis to be removed from `self`. + Note: It must be contained in `self`. Otherwise, + an exception is raised. + """ + key = hyp.key + assert key in self, f"{key} does not exist" + del self._data[key] + + def filter(self, threshold: torch.Tensor) -> "HypothesisList": + """Remove all Hypotheses whose log_prob is less than threshold. + + Caution: + `self` is not modified. Instead, a new HypothesisList is returned. + + Returns: + Return a new HypothesisList containing all hypotheses from `self` + with `log_prob` being greater than the given `threshold`. + """ + ans = HypothesisList() + for _, hyp in self._data.items(): + if hyp.log_prob > threshold: + ans.add(hyp) # shallow copy + return ans + + def topk(self, k: int) -> "HypothesisList": + """Return the top-k hypothesis.""" + hyps = list(self._data.items()) + + hyps = sorted(hyps, key=lambda h: h[1].log_prob, reverse=True)[:k] + + ans = HypothesisList(dict(hyps)) + return ans + + def __contains__(self, key: str): + return key in self._data + + def __iter__(self): + return iter(self._data.values()) + + def __len__(self) -> int: + return len(self._data) + + def __str__(self) -> str: + s = [] + for key in self: + s.append(key) + return ", ".join(s) + + +def run_decoder( + ys: List[int], + model: Transducer, + decoder_cache: Dict[str, torch.Tensor], +) -> torch.Tensor: + """Run the neural decoder model for a given hypothesis. + + Args: + ys: + The current hypothesis. + model: + The transducer model. + decoder_cache: + Cache to save computations. + Returns: + Return a 1-D tensor of shape (decoder_out_dim,) containing + output of `model.decoder`. + """ + context_size = model.decoder.context_size + key = "_".join(map(str, ys[-context_size:])) + if key in decoder_cache: + return decoder_cache[key] + + device = model.device + + decoder_input = torch.tensor([ys[-context_size:]], device=device).reshape( + 1, context_size + ) + + decoder_out = model.decoder(decoder_input, need_pad=False) + decoder_cache[key] = decoder_out + + return decoder_out + + +def run_joiner( + key: str, + model: Transducer, + encoder_out: torch.Tensor, + decoder_out: torch.Tensor, + encoder_out_len: torch.Tensor, + decoder_out_len: torch.Tensor, + joint_cache: Dict[str, torch.Tensor], +): + """Run the joint network given outputs from the encoder and decoder. + + Args: + key: + A key into the `joint_cache`. + model: + The transducer model. + encoder_out: + A tensor of shape (1, 1, encoder_out_dim). + decoder_out: + A tensor of shape (1, 1, decoder_out_dim). + encoder_out_len: + A tensor with value [1]. + decoder_out_len: + A tensor with value [1]. + joint_cache: + A dict to save computations. + Returns: + Return a tensor from the output of log-softmax. + Its shape is (vocab_size,). + """ + if key in joint_cache: + return joint_cache[key] + + logits = model.joiner( + encoder_out, + decoder_out, + encoder_out_len, + decoder_out_len, + ) + + # TODO(fangjun): Scale the blank posterior + log_prob = logits.log_softmax(dim=-1) + # log_prob is (1, 1, 1, vocab_size) + + log_prob = log_prob.squeeze() + # Now log_prob is (vocab_size,) + + joint_cache[key] = log_prob + + return log_prob + + +def modified_beam_search( + model: Transducer, + encoder_out: torch.Tensor, + beam: int = 4, +) -> List[int]: + """It limits the maximum number of symbols per frame to 1. + + Args: + model: + An instance of `Transducer`. + encoder_out: + A tensor of shape (N, T, C) from the encoder. Support only N==1 for now. + beam: + Beam size. + Returns: + Return the decoded result. + """ + + assert encoder_out.ndim == 3 + + # support only batch_size == 1 for now + assert encoder_out.size(0) == 1, encoder_out.size(0) + blank_id = model.decoder.blank_id + context_size = model.decoder.context_size + + device = model.device + + decoder_input = torch.tensor( + [blank_id] * context_size, device=device + ).reshape(1, context_size) + + decoder_out = model.decoder(decoder_input, need_pad=False) + + T = encoder_out.size(1) + + B = HypothesisList() + B.add( + Hypothesis( + ys=[blank_id] * context_size, + log_prob=torch.zeros(1, dtype=torch.float32, device=device), + ) + ) + + encoder_out_len = torch.tensor([1]) + decoder_out_len = torch.tensor([1]) + + for t in range(T): + # fmt: off + current_encoder_out = encoder_out[:, t:t+1, :] + # current_encoder_out is of shape (1, 1, encoder_out_dim) + # fmt: on + A = list(B) + B = HypothesisList() + + ys_log_probs = torch.cat([hyp.log_prob.reshape(1, 1) for hyp in A]) + # ys_log_probs is of shape (num_hyps, 1) + + decoder_input = torch.tensor( + [hyp.ys[-context_size:] for hyp in A], + device=device, + ) + # decoder_input is of shape (num_hyps, context_size) + + decoder_out = model.decoder(decoder_input, need_pad=False) + # decoder_output is of shape (num_hyps, 1, decoder_output_dim) + + current_encoder_out = current_encoder_out.expand( + decoder_out.size(0), 1, -1 + ) + + logits = model.joiner( + current_encoder_out, + decoder_out, + encoder_out_len.expand(decoder_out.size(0)), + decoder_out_len.expand(decoder_out.size(0)), + ) + # logits is of shape (num_hyps, vocab_size) + log_probs = logits.log_softmax(dim=-1) + + log_probs.add_(ys_log_probs) + + log_probs = log_probs.reshape(-1) + topk_log_probs, topk_indexes = log_probs.topk(beam) + + # topk_hyp_indexes are indexes into `A` + topk_hyp_indexes = topk_indexes // logits.size(-1) + topk_token_indexes = topk_indexes % logits.size(-1) + + topk_hyp_indexes = topk_hyp_indexes.tolist() + topk_token_indexes = topk_token_indexes.tolist() + + for i in range(len(topk_hyp_indexes)): + hyp = A[topk_hyp_indexes[i]] + new_ys = hyp.ys[:] + new_token = topk_token_indexes[i] + if new_token != blank_id: + new_ys.append(new_token) + new_log_prob = topk_log_probs[i] + new_hyp = Hypothesis(ys=new_ys, log_prob=new_log_prob) + B.add(new_hyp) + + best_hyp = B.get_most_probable(length_norm=True) + ys = best_hyp.ys[context_size:] # [context_size:] to remove blanks + + return ys + + +def beam_search( + model: Transducer, + encoder_out: torch.Tensor, + beam: int = 4, +) -> List[int]: + """ + It implements Algorithm 1 in https://arxiv.org/pdf/1211.3711.pdf + + espnet/nets/beam_search_transducer.py#L247 is used as a reference. + + Args: + model: + An instance of `Transducer`. + encoder_out: + A tensor of shape (N, T, C) from the encoder. Support only N==1 for now. + beam: + Beam size. + Returns: + Return the decoded result. + """ + assert encoder_out.ndim == 3 + + # support only batch_size == 1 for now + assert encoder_out.size(0) == 1, encoder_out.size(0) + blank_id = model.decoder.blank_id + context_size = model.decoder.context_size + + device = model.device + + decoder_input = torch.tensor( + [blank_id] * context_size, device=device + ).reshape(1, context_size) + + decoder_out = model.decoder(decoder_input, need_pad=False) + + T = encoder_out.size(1) + t = 0 + + B = HypothesisList() + B.add( + Hypothesis( + ys=[blank_id] * context_size, + log_prob=torch.zeros(1, dtype=torch.float32, device=device), + ) + ) + + max_sym_per_utt = 20000 + + sym_per_utt = 0 + + encoder_out_len = torch.tensor([1]) + decoder_out_len = torch.tensor([1]) + + decoder_cache: Dict[str, torch.Tensor] = {} + + while t < T and sym_per_utt < max_sym_per_utt: + # fmt: off + current_encoder_out = encoder_out[:, t:t+1, :] + # fmt: on + A = B + B = HypothesisList() + + joint_cache: Dict[str, torch.Tensor] = {} + + while True: + y_star = A.get_most_probable() + A.remove(y_star) + + decoder_out = run_decoder( + ys=y_star.ys, model=model, decoder_cache=decoder_cache + ) + + key = "_".join(map(str, y_star.ys[-context_size:])) + key += f"-t-{t}" + log_prob = run_joiner( + key=key, + model=model, + encoder_out=current_encoder_out, + decoder_out=decoder_out, + encoder_out_len=encoder_out_len, + decoder_out_len=decoder_out_len, + joint_cache=joint_cache, + ) + + # First, process the blank symbol + skip_log_prob = log_prob[blank_id] + new_y_star_log_prob = y_star.log_prob + skip_log_prob + + # ys[:] returns a copy of ys + B.add(Hypothesis(ys=y_star.ys[:], log_prob=new_y_star_log_prob)) + + # Second, process other non-blank labels + values, indices = log_prob.topk(beam + 1) + for idx in range(values.size(0)): + i = indices[idx].item() + if i == blank_id: + continue + + new_ys = y_star.ys + [i] + + new_log_prob = y_star.log_prob + values[idx] + A.add(Hypothesis(ys=new_ys, log_prob=new_log_prob)) + + # Check whether B contains more than "beam" elements more probable + # than the most probable in A + A_most_probable = A.get_most_probable() + + kept_B = B.filter(A_most_probable.log_prob) + + if len(kept_B) >= beam: + B = kept_B.topk(beam) + break + + t += 1 + + best_hyp = B.get_most_probable(length_norm=True) + ys = best_hyp.ys[context_size:] # [context_size:] to remove blanks + return ys diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/conformer.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/conformer.py new file mode 100644 index 000000000..81d7708f9 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/conformer.py @@ -0,0 +1,920 @@ +#!/usr/bin/env python3 +# Copyright (c) 2021 University of Chinese Academy of Sciences (author: Han Zhu) +# +# 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 math +import warnings +from typing import Optional, Tuple + +import torch +from torch import Tensor, nn +from transformer import Transformer + +from icefall.utils import make_pad_mask + + +class Conformer(Transformer): + """ + Args: + num_features (int): Number of input features + output_dim (int): Number of output dimension + subsampling_factor (int): subsampling factor of encoder (the convolution layers before transformers) + d_model (int): attention dimension + nhead (int): number of head + dim_feedforward (int): feedforward dimention + num_encoder_layers (int): number of encoder layers + dropout (float): dropout rate + cnn_module_kernel (int): Kernel size of convolution module + normalize_before (bool): whether to use layer_norm before the first block. + vgg_frontend (bool): whether to use vgg frontend. + """ + + def __init__( + self, + num_features: int, + output_dim: int, + subsampling_factor: int = 4, + d_model: int = 256, + nhead: int = 4, + dim_feedforward: int = 2048, + num_encoder_layers: int = 12, + dropout: float = 0.1, + cnn_module_kernel: int = 31, + normalize_before: bool = True, + vgg_frontend: bool = False, + ) -> None: + super(Conformer, self).__init__( + num_features=num_features, + output_dim=output_dim, + subsampling_factor=subsampling_factor, + d_model=d_model, + nhead=nhead, + dim_feedforward=dim_feedforward, + num_encoder_layers=num_encoder_layers, + dropout=dropout, + normalize_before=normalize_before, + vgg_frontend=vgg_frontend, + ) + + self.encoder_pos = RelPositionalEncoding(d_model, dropout) + + encoder_layer = ConformerEncoderLayer( + d_model, + nhead, + dim_feedforward, + dropout, + cnn_module_kernel, + normalize_before, + ) + self.encoder = ConformerEncoder(encoder_layer, num_encoder_layers) + self.normalize_before = normalize_before + if self.normalize_before: + self.after_norm = nn.LayerNorm(d_model) + else: + # Note: TorchScript detects that self.after_norm could be used inside forward() + # and throws an error without this change. + self.after_norm = identity + + def forward( + self, x: torch.Tensor, x_lens: torch.Tensor + ) -> Tuple[torch.Tensor, torch.Tensor]: + """ + Args: + x: + The input tensor. Its shape is (batch_size, seq_len, feature_dim). + x_lens: + A tensor of shape (batch_size,) containing the number of frames in + `x` before padding. + Returns: + Return a tuple containing 2 tensors: + - logits, its shape is (batch_size, output_seq_len, output_dim) + - logit_lens, a tensor of shape (batch_size,) containing the number + of frames in `logits` before padding. + """ + x = self.encoder_embed(x) + x, pos_emb = self.encoder_pos(x) + x = x.permute(1, 0, 2) # (N, T, C) -> (T, N, C) + + # Caution: We assume the subsampling factor is 4! + lengths = ((x_lens - 1) // 2 - 1) // 2 + assert x.size(0) == lengths.max().item() + mask = make_pad_mask(lengths) + + x = self.encoder(x, pos_emb, src_key_padding_mask=mask) # (T, N, C) + + if self.normalize_before: + x = self.after_norm(x) + + logits = self.encoder_output_layer(x) + logits = logits.permute(1, 0, 2) # (T, N, C) ->(N, T, C) + + return logits, lengths + + +class ConformerEncoderLayer(nn.Module): + """ + ConformerEncoderLayer is made up of self-attn, feedforward and convolution networks. + See: "Conformer: Convolution-augmented Transformer for Speech Recognition" + + Args: + d_model: the number of expected features in the input (required). + nhead: the number of heads in the multiheadattention models (required). + dim_feedforward: the dimension of the feedforward network model (default=2048). + dropout: the dropout value (default=0.1). + cnn_module_kernel (int): Kernel size of convolution module. + normalize_before: whether to use layer_norm before the first block. + + Examples:: + >>> encoder_layer = ConformerEncoderLayer(d_model=512, nhead=8) + >>> src = torch.rand(10, 32, 512) + >>> pos_emb = torch.rand(32, 19, 512) + >>> out = encoder_layer(src, pos_emb) + """ + + def __init__( + self, + d_model: int, + nhead: int, + dim_feedforward: int = 2048, + dropout: float = 0.1, + cnn_module_kernel: int = 31, + normalize_before: bool = True, + ) -> None: + super(ConformerEncoderLayer, self).__init__() + self.self_attn = RelPositionMultiheadAttention( + d_model, nhead, dropout=0.0 + ) + + self.feed_forward = nn.Sequential( + nn.Linear(d_model, dim_feedforward), + Swish(), + nn.Dropout(dropout), + nn.Linear(dim_feedforward, d_model), + ) + + self.feed_forward_macaron = nn.Sequential( + nn.Linear(d_model, dim_feedforward), + Swish(), + nn.Dropout(dropout), + nn.Linear(dim_feedforward, d_model), + ) + + self.conv_module = ConvolutionModule(d_model, cnn_module_kernel) + + self.norm_ff_macaron = nn.LayerNorm( + d_model + ) # for the macaron style FNN module + self.norm_ff = nn.LayerNorm(d_model) # for the FNN module + self.norm_mha = nn.LayerNorm(d_model) # for the MHA module + + self.ff_scale = 0.5 + + self.norm_conv = nn.LayerNorm(d_model) # for the CNN module + self.norm_final = nn.LayerNorm( + d_model + ) # for the final output of the block + + self.dropout = nn.Dropout(dropout) + + self.normalize_before = normalize_before + + def forward( + self, + src: Tensor, + pos_emb: Tensor, + src_mask: Optional[Tensor] = None, + src_key_padding_mask: Optional[Tensor] = None, + ) -> Tensor: + """ + Pass the input through the encoder layer. + + Args: + src: the sequence to the encoder layer (required). + pos_emb: Positional embedding tensor (required). + src_mask: the mask for the src sequence (optional). + src_key_padding_mask: the mask for the src keys per batch (optional). + + Shape: + src: (S, N, E). + pos_emb: (N, 2*S-1, E) + src_mask: (S, S). + src_key_padding_mask: (N, S). + S is the source sequence length, N is the batch size, E is the feature number + """ + + # macaron style feed forward module + residual = src + if self.normalize_before: + src = self.norm_ff_macaron(src) + src = residual + self.ff_scale * self.dropout( + self.feed_forward_macaron(src) + ) + if not self.normalize_before: + src = self.norm_ff_macaron(src) + + # multi-headed self-attention module + residual = src + if self.normalize_before: + src = self.norm_mha(src) + src_att = self.self_attn( + src, + src, + src, + pos_emb=pos_emb, + attn_mask=src_mask, + key_padding_mask=src_key_padding_mask, + )[0] + src = residual + self.dropout(src_att) + if not self.normalize_before: + src = self.norm_mha(src) + + # convolution module + residual = src + if self.normalize_before: + src = self.norm_conv(src) + src = residual + self.dropout(self.conv_module(src)) + if not self.normalize_before: + src = self.norm_conv(src) + + # feed forward module + residual = src + if self.normalize_before: + src = self.norm_ff(src) + src = residual + self.ff_scale * self.dropout(self.feed_forward(src)) + if not self.normalize_before: + src = self.norm_ff(src) + + if self.normalize_before: + src = self.norm_final(src) + + return src + + +class ConformerEncoder(nn.TransformerEncoder): + r"""ConformerEncoder is a stack of N encoder layers + + Args: + encoder_layer: an instance of the ConformerEncoderLayer() class (required). + num_layers: the number of sub-encoder-layers in the encoder (required). + norm: the layer normalization component (optional). + + Examples:: + >>> encoder_layer = ConformerEncoderLayer(d_model=512, nhead=8) + >>> conformer_encoder = ConformerEncoder(encoder_layer, num_layers=6) + >>> src = torch.rand(10, 32, 512) + >>> pos_emb = torch.rand(32, 19, 512) + >>> out = conformer_encoder(src, pos_emb) + """ + + def __init__( + self, encoder_layer: nn.Module, num_layers: int, norm: nn.Module = None + ) -> None: + super(ConformerEncoder, self).__init__( + encoder_layer=encoder_layer, num_layers=num_layers, norm=norm + ) + + def forward( + self, + src: Tensor, + pos_emb: Tensor, + mask: Optional[Tensor] = None, + src_key_padding_mask: Optional[Tensor] = None, + ) -> Tensor: + r"""Pass the input through the encoder layers in turn. + + Args: + src: the sequence to the encoder (required). + pos_emb: Positional embedding tensor (required). + mask: the mask for the src sequence (optional). + src_key_padding_mask: the mask for the src keys per batch (optional). + + Shape: + src: (S, N, E). + pos_emb: (N, 2*S-1, E) + mask: (S, S). + src_key_padding_mask: (N, S). + S is the source sequence length, T is the target sequence length, N is the batch size, E is the feature number + + """ + output = src + + for mod in self.layers: + output = mod( + output, + pos_emb, + src_mask=mask, + src_key_padding_mask=src_key_padding_mask, + ) + + if self.norm is not None: + output = self.norm(output) + + return output + + +class RelPositionalEncoding(torch.nn.Module): + """Relative positional encoding module. + + See : Appendix B in "Transformer-XL: Attentive Language Models Beyond a Fixed-Length Context" + Modified from https://github.com/espnet/espnet/blob/master/espnet/nets/pytorch_backend/transformer/embedding.py + + Args: + d_model: Embedding dimension. + dropout_rate: Dropout rate. + max_len: Maximum input length. + + """ + + def __init__( + self, d_model: int, dropout_rate: float, max_len: int = 5000 + ) -> None: + """Construct an PositionalEncoding object.""" + super(RelPositionalEncoding, self).__init__() + self.d_model = d_model + self.xscale = math.sqrt(self.d_model) + self.dropout = torch.nn.Dropout(p=dropout_rate) + self.pe = None + self.extend_pe(torch.tensor(0.0).expand(1, max_len)) + + def extend_pe(self, x: Tensor) -> None: + """Reset the positional encodings.""" + if self.pe is not None: + # self.pe contains both positive and negative parts + # the length of self.pe is 2 * input_len - 1 + if self.pe.size(1) >= x.size(1) * 2 - 1: + # Note: TorchScript doesn't implement operator== for torch.Device + if self.pe.dtype != x.dtype or str(self.pe.device) != str( + x.device + ): + self.pe = self.pe.to(dtype=x.dtype, device=x.device) + return + # Suppose `i` means to the position of query vecotr and `j` means the + # position of key vector. We use position relative positions when keys + # are to the left (i>j) and negative relative positions otherwise (i Tuple[Tensor, Tensor]: + """Add positional encoding. + + Args: + x (torch.Tensor): Input tensor (batch, time, `*`). + + Returns: + torch.Tensor: Encoded tensor (batch, time, `*`). + torch.Tensor: Encoded tensor (batch, 2*time-1, `*`). + + """ + self.extend_pe(x) + x = x * self.xscale + pos_emb = self.pe[ + :, + self.pe.size(1) // 2 + - x.size(1) + + 1 : self.pe.size(1) // 2 # noqa E203 + + x.size(1), + ] + return self.dropout(x), self.dropout(pos_emb) + + +class RelPositionMultiheadAttention(nn.Module): + r"""Multi-Head Attention layer with relative position encoding + + See reference: "Transformer-XL: Attentive Language Models Beyond a Fixed-Length Context" + + Args: + embed_dim: total dimension of the model. + num_heads: parallel attention heads. + dropout: a Dropout layer on attn_output_weights. Default: 0.0. + + Examples:: + + >>> rel_pos_multihead_attn = RelPositionMultiheadAttention(embed_dim, num_heads) + >>> attn_output, attn_output_weights = multihead_attn(query, key, value, pos_emb) + """ + + def __init__( + self, + embed_dim: int, + num_heads: int, + dropout: float = 0.0, + ) -> None: + super(RelPositionMultiheadAttention, self).__init__() + self.embed_dim = embed_dim + self.num_heads = num_heads + self.dropout = dropout + self.head_dim = embed_dim // num_heads + assert ( + self.head_dim * num_heads == self.embed_dim + ), "embed_dim must be divisible by num_heads" + + self.in_proj = nn.Linear(embed_dim, 3 * embed_dim, bias=True) + self.out_proj = nn.Linear(embed_dim, embed_dim, bias=True) + + # linear transformation for positional encoding. + self.linear_pos = nn.Linear(embed_dim, embed_dim, bias=False) + # these two learnable bias are used in matrix c and matrix d + # as described in "Transformer-XL: Attentive Language Models Beyond a Fixed-Length Context" Section 3.3 + self.pos_bias_u = nn.Parameter(torch.Tensor(num_heads, self.head_dim)) + self.pos_bias_v = nn.Parameter(torch.Tensor(num_heads, self.head_dim)) + + self._reset_parameters() + + def _reset_parameters(self) -> None: + nn.init.xavier_uniform_(self.in_proj.weight) + nn.init.constant_(self.in_proj.bias, 0.0) + nn.init.constant_(self.out_proj.bias, 0.0) + + nn.init.xavier_uniform_(self.pos_bias_u) + nn.init.xavier_uniform_(self.pos_bias_v) + + def forward( + self, + query: Tensor, + key: Tensor, + value: Tensor, + pos_emb: Tensor, + key_padding_mask: Optional[Tensor] = None, + need_weights: bool = True, + attn_mask: Optional[Tensor] = None, + ) -> Tuple[Tensor, Optional[Tensor]]: + r""" + Args: + query, key, value: map a query and a set of key-value pairs to an output. + pos_emb: Positional embedding tensor + key_padding_mask: if provided, specified padding elements in the key will + be ignored by the attention. When given a binary mask and a value is True, + the corresponding value on the attention layer will be ignored. When given + a byte mask and a value is non-zero, the corresponding value on the attention + layer will be ignored + need_weights: output attn_output_weights. + attn_mask: 2D or 3D mask that prevents attention to certain positions. A 2D mask will be broadcasted for all + the batches while a 3D mask allows to specify a different mask for the entries of each batch. + + Shape: + - Inputs: + - query: :math:`(L, N, E)` where L is the target sequence length, N is the batch size, E is + the embedding dimension. + - key: :math:`(S, N, E)`, where S is the source sequence length, N is the batch size, E is + the embedding dimension. + - value: :math:`(S, N, E)` where S is the source sequence length, N is the batch size, E is + the embedding dimension. + - pos_emb: :math:`(N, 2*L-1, E)` where L is the target sequence length, N is the batch size, E is + the embedding dimension. + - key_padding_mask: :math:`(N, S)` where N is the batch size, S is the source sequence length. + If a ByteTensor is provided, the non-zero positions will be ignored while the position + with the zero positions will be unchanged. If a BoolTensor is provided, the positions with the + value of ``True`` will be ignored while the position with the value of ``False`` will be unchanged. + - attn_mask: 2D mask :math:`(L, S)` where L is the target sequence length, S is the source sequence length. + 3D mask :math:`(N*num_heads, L, S)` where N is the batch size, L is the target sequence length, + S is the source sequence length. attn_mask ensure that position i is allowed to attend the unmasked + positions. If a ByteTensor is provided, the non-zero positions are not allowed to attend + while the zero positions will be unchanged. If a BoolTensor is provided, positions with ``True`` + is not allowed to attend while ``False`` values will be unchanged. If a FloatTensor + is provided, it will be added to the attention weight. + + - Outputs: + - attn_output: :math:`(L, N, E)` where L is the target sequence length, N is the batch size, + E is the embedding dimension. + - attn_output_weights: :math:`(N, L, S)` where N is the batch size, + L is the target sequence length, S is the source sequence length. + """ + return self.multi_head_attention_forward( + query, + key, + value, + pos_emb, + self.embed_dim, + self.num_heads, + self.in_proj.weight, + self.in_proj.bias, + self.dropout, + self.out_proj.weight, + self.out_proj.bias, + training=self.training, + key_padding_mask=key_padding_mask, + need_weights=need_weights, + attn_mask=attn_mask, + ) + + def rel_shift(self, x: Tensor) -> Tensor: + """Compute relative positional encoding. + + Args: + x: Input tensor (batch, head, time1, 2*time1-1). + time1 means the length of query vector. + + Returns: + Tensor: tensor of shape (batch, head, time1, time2) + (note: time2 has the same value as time1, but it is for + the key, while time1 is for the query). + """ + (batch_size, num_heads, time1, n) = x.shape + assert n == 2 * time1 - 1 + # Note: TorchScript requires explicit arg for stride() + batch_stride = x.stride(0) + head_stride = x.stride(1) + time1_stride = x.stride(2) + n_stride = x.stride(3) + return x.as_strided( + (batch_size, num_heads, time1, time1), + (batch_stride, head_stride, time1_stride - n_stride, n_stride), + storage_offset=n_stride * (time1 - 1), + ) + + def multi_head_attention_forward( + self, + query: Tensor, + key: Tensor, + value: Tensor, + pos_emb: Tensor, + embed_dim_to_check: int, + num_heads: int, + in_proj_weight: Tensor, + in_proj_bias: Tensor, + dropout_p: float, + out_proj_weight: Tensor, + out_proj_bias: Tensor, + training: bool = True, + key_padding_mask: Optional[Tensor] = None, + need_weights: bool = True, + attn_mask: Optional[Tensor] = None, + ) -> Tuple[Tensor, Optional[Tensor]]: + r""" + Args: + query, key, value: map a query and a set of key-value pairs to an output. + pos_emb: Positional embedding tensor + embed_dim_to_check: total dimension of the model. + num_heads: parallel attention heads. + in_proj_weight, in_proj_bias: input projection weight and bias. + dropout_p: probability of an element to be zeroed. + out_proj_weight, out_proj_bias: the output projection weight and bias. + training: apply dropout if is ``True``. + key_padding_mask: if provided, specified padding elements in the key will + be ignored by the attention. This is an binary mask. When the value is True, + the corresponding value on the attention layer will be filled with -inf. + need_weights: output attn_output_weights. + attn_mask: 2D or 3D mask that prevents attention to certain positions. A 2D mask will be broadcasted for all + the batches while a 3D mask allows to specify a different mask for the entries of each batch. + + Shape: + Inputs: + - query: :math:`(L, N, E)` where L is the target sequence length, N is the batch size, E is + the embedding dimension. + - key: :math:`(S, N, E)`, where S is the source sequence length, N is the batch size, E is + the embedding dimension. + - value: :math:`(S, N, E)` where S is the source sequence length, N is the batch size, E is + the embedding dimension. + - pos_emb: :math:`(N, 2*L-1, E)` or :math:`(1, 2*L-1, E)` where L is the target sequence + length, N is the batch size, E is the embedding dimension. + - key_padding_mask: :math:`(N, S)` where N is the batch size, S is the source sequence length. + If a ByteTensor is provided, the non-zero positions will be ignored while the zero positions + will be unchanged. If a BoolTensor is provided, the positions with the + value of ``True`` will be ignored while the position with the value of ``False`` will be unchanged. + - attn_mask: 2D mask :math:`(L, S)` where L is the target sequence length, S is the source sequence length. + 3D mask :math:`(N*num_heads, L, S)` where N is the batch size, L is the target sequence length, + S is the source sequence length. attn_mask ensures that position i is allowed to attend the unmasked + positions. If a ByteTensor is provided, the non-zero positions are not allowed to attend + while the zero positions will be unchanged. If a BoolTensor is provided, positions with ``True`` + are not allowed to attend while ``False`` values will be unchanged. If a FloatTensor + is provided, it will be added to the attention weight. + + Outputs: + - attn_output: :math:`(L, N, E)` where L is the target sequence length, N is the batch size, + E is the embedding dimension. + - attn_output_weights: :math:`(N, L, S)` where N is the batch size, + L is the target sequence length, S is the source sequence length. + """ + + tgt_len, bsz, embed_dim = query.size() + assert embed_dim == embed_dim_to_check + assert key.size(0) == value.size(0) and key.size(1) == value.size(1) + + head_dim = embed_dim // num_heads + assert ( + head_dim * num_heads == embed_dim + ), "embed_dim must be divisible by num_heads" + scaling = float(head_dim) ** -0.5 + + if torch.equal(query, key) and torch.equal(key, value): + # self-attention + q, k, v = nn.functional.linear( + query, in_proj_weight, in_proj_bias + ).chunk(3, dim=-1) + + elif torch.equal(key, value): + # encoder-decoder attention + # This is inline in_proj function with in_proj_weight and in_proj_bias + _b = in_proj_bias + _start = 0 + _end = embed_dim + _w = in_proj_weight[_start:_end, :] + if _b is not None: + _b = _b[_start:_end] + q = nn.functional.linear(query, _w, _b) + # This is inline in_proj function with in_proj_weight and in_proj_bias + _b = in_proj_bias + _start = embed_dim + _end = None + _w = in_proj_weight[_start:, :] + if _b is not None: + _b = _b[_start:] + k, v = nn.functional.linear(key, _w, _b).chunk(2, dim=-1) + + else: + # This is inline in_proj function with in_proj_weight and in_proj_bias + _b = in_proj_bias + _start = 0 + _end = embed_dim + _w = in_proj_weight[_start:_end, :] + if _b is not None: + _b = _b[_start:_end] + q = nn.functional.linear(query, _w, _b) + + # This is inline in_proj function with in_proj_weight and in_proj_bias + _b = in_proj_bias + _start = embed_dim + _end = embed_dim * 2 + _w = in_proj_weight[_start:_end, :] + if _b is not None: + _b = _b[_start:_end] + k = nn.functional.linear(key, _w, _b) + + # This is inline in_proj function with in_proj_weight and in_proj_bias + _b = in_proj_bias + _start = embed_dim * 2 + _end = None + _w = in_proj_weight[_start:, :] + if _b is not None: + _b = _b[_start:] + v = nn.functional.linear(value, _w, _b) + + if attn_mask is not None: + assert ( + attn_mask.dtype == torch.float32 + or attn_mask.dtype == torch.float64 + or attn_mask.dtype == torch.float16 + or attn_mask.dtype == torch.uint8 + or attn_mask.dtype == torch.bool + ), "Only float, byte, and bool types are supported for attn_mask, not {}".format( + attn_mask.dtype + ) + if attn_mask.dtype == torch.uint8: + warnings.warn( + "Byte tensor for attn_mask is deprecated. Use bool tensor instead." + ) + attn_mask = attn_mask.to(torch.bool) + + if attn_mask.dim() == 2: + attn_mask = attn_mask.unsqueeze(0) + if list(attn_mask.size()) != [1, query.size(0), key.size(0)]: + raise RuntimeError( + "The size of the 2D attn_mask is not correct." + ) + elif attn_mask.dim() == 3: + if list(attn_mask.size()) != [ + bsz * num_heads, + query.size(0), + key.size(0), + ]: + raise RuntimeError( + "The size of the 3D attn_mask is not correct." + ) + else: + raise RuntimeError( + "attn_mask's dimension {} is not supported".format( + attn_mask.dim() + ) + ) + # attn_mask's dim is 3 now. + + # convert ByteTensor key_padding_mask to bool + if ( + key_padding_mask is not None + and key_padding_mask.dtype == torch.uint8 + ): + warnings.warn( + "Byte tensor for key_padding_mask is deprecated. Use bool tensor instead." + ) + key_padding_mask = key_padding_mask.to(torch.bool) + + q = q.contiguous().view(tgt_len, bsz, num_heads, head_dim) + k = k.contiguous().view(-1, bsz, num_heads, head_dim) + v = v.contiguous().view(-1, bsz * num_heads, head_dim).transpose(0, 1) + + src_len = k.size(0) + + if key_padding_mask is not None: + assert key_padding_mask.size(0) == bsz, "{} == {}".format( + key_padding_mask.size(0), bsz + ) + assert key_padding_mask.size(1) == src_len, "{} == {}".format( + key_padding_mask.size(1), src_len + ) + + q = q.transpose(0, 1) # (batch, time1, head, d_k) + + pos_emb_bsz = pos_emb.size(0) + assert pos_emb_bsz in (1, bsz) # actually it is 1 + p = self.linear_pos(pos_emb).view(pos_emb_bsz, -1, num_heads, head_dim) + p = p.transpose(1, 2) # (batch, head, 2*time1-1, d_k) + + q_with_bias_u = (q + self.pos_bias_u).transpose( + 1, 2 + ) # (batch, head, time1, d_k) + + q_with_bias_v = (q + self.pos_bias_v).transpose( + 1, 2 + ) # (batch, head, time1, d_k) + + # compute attention score + # first compute matrix a and matrix c + # as described in "Transformer-XL: Attentive Language Models Beyond a Fixed-Length Context" Section 3.3 + k = k.permute(1, 2, 3, 0) # (batch, head, d_k, time2) + matrix_ac = torch.matmul( + q_with_bias_u, k + ) # (batch, head, time1, time2) + + # compute matrix b and matrix d + matrix_bd = torch.matmul( + q_with_bias_v, p.transpose(-2, -1) + ) # (batch, head, time1, 2*time1-1) + matrix_bd = self.rel_shift(matrix_bd) + + attn_output_weights = ( + matrix_ac + matrix_bd + ) * scaling # (batch, head, time1, time2) + + attn_output_weights = attn_output_weights.view( + bsz * num_heads, tgt_len, -1 + ) + + assert list(attn_output_weights.size()) == [ + bsz * num_heads, + tgt_len, + src_len, + ] + + if attn_mask is not None: + if attn_mask.dtype == torch.bool: + attn_output_weights.masked_fill_(attn_mask, float("-inf")) + else: + attn_output_weights += attn_mask + + if key_padding_mask is not None: + attn_output_weights = attn_output_weights.view( + bsz, num_heads, tgt_len, src_len + ) + attn_output_weights = attn_output_weights.masked_fill( + key_padding_mask.unsqueeze(1).unsqueeze(2), + float("-inf"), + ) + attn_output_weights = attn_output_weights.view( + bsz * num_heads, tgt_len, src_len + ) + + attn_output_weights = nn.functional.softmax(attn_output_weights, dim=-1) + attn_output_weights = nn.functional.dropout( + attn_output_weights, p=dropout_p, training=training + ) + + attn_output = torch.bmm(attn_output_weights, v) + assert list(attn_output.size()) == [bsz * num_heads, tgt_len, head_dim] + attn_output = ( + attn_output.transpose(0, 1) + .contiguous() + .view(tgt_len, bsz, embed_dim) + ) + attn_output = nn.functional.linear( + attn_output, out_proj_weight, out_proj_bias + ) + + if need_weights: + # average attention weights over heads + attn_output_weights = attn_output_weights.view( + bsz, num_heads, tgt_len, src_len + ) + return attn_output, attn_output_weights.sum(dim=1) / num_heads + else: + return attn_output, None + + +class ConvolutionModule(nn.Module): + """ConvolutionModule in Conformer model. + Modified from https://github.com/espnet/espnet/blob/master/espnet/nets/pytorch_backend/conformer/convolution.py + + Args: + channels (int): The number of channels of conv layers. + kernel_size (int): Kernerl size of conv layers. + bias (bool): Whether to use bias in conv layers (default=True). + + """ + + def __init__( + self, channels: int, kernel_size: int, bias: bool = True + ) -> None: + """Construct an ConvolutionModule object.""" + super(ConvolutionModule, self).__init__() + # kernerl_size should be a odd number for 'SAME' padding + assert (kernel_size - 1) % 2 == 0 + + self.pointwise_conv1 = nn.Conv1d( + channels, + 2 * channels, + kernel_size=1, + stride=1, + padding=0, + bias=bias, + ) + self.depthwise_conv = nn.Conv1d( + channels, + channels, + kernel_size, + stride=1, + padding=(kernel_size - 1) // 2, + groups=channels, + bias=bias, + ) + self.norm = nn.LayerNorm(channels) + self.pointwise_conv2 = nn.Conv1d( + channels, + channels, + kernel_size=1, + stride=1, + padding=0, + bias=bias, + ) + self.activation = Swish() + + def forward(self, x: Tensor) -> Tensor: + """Compute convolution module. + + Args: + x: Input tensor (#time, batch, channels). + + Returns: + Tensor: Output tensor (#time, batch, channels). + + """ + # exchange the temporal dimension and the feature dimension + x = x.permute(1, 2, 0) # (#batch, channels, time). + + # GLU mechanism + x = self.pointwise_conv1(x) # (batch, 2*channels, time) + x = nn.functional.glu(x, dim=1) # (batch, channels, time) + + # 1D Depthwise Conv + x = self.depthwise_conv(x) + # x is (batch, channels, time) + x = x.permute(0, 2, 1) + x = self.norm(x) + x = x.permute(0, 2, 1) + + x = self.activation(x) + + x = self.pointwise_conv2(x) # (batch, channel, time) + + return x.permute(2, 0, 1) + + +class Swish(torch.nn.Module): + """Construct an Swish object.""" + + def forward(self, x: Tensor) -> Tensor: + """Return Swich activation function.""" + return x * torch.sigmoid(x) + + +def identity(x): + return x diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/decode.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/decode.py new file mode 100755 index 000000000..136afe9c0 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/decode.py @@ -0,0 +1,490 @@ +#!/usr/bin/env python3 +# +# 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. +""" +Usage: +(1) greedy search +./transducer_stateless_multi_datasets/decode.py \ + --epoch 14 \ + --avg 7 \ + --exp-dir ./transducer_stateless_multi_datasets/exp \ + --max-duration 100 \ + --decoding-method greedy_search + +(2) beam search +./transducer_stateless_multi_datasets/decode.py \ + --epoch 14 \ + --avg 7 \ + --exp-dir ./transducer_stateless_multi_datasets/exp \ + --max-duration 100 \ + --decoding-method beam_search \ + --beam-size 4 +""" + + +import argparse +import logging +from collections import defaultdict +from pathlib import Path +from typing import Dict, List, Tuple + +import sentencepiece as spm +import torch +import torch.nn as nn +from asr_datamodule import AsrDataModule +from beam_search import beam_search, greedy_search, modified_beam_search +from conformer import Conformer +from decoder import Decoder +from joiner import Joiner +from librispeech import LibriSpeech +from model import Transducer + +from icefall.checkpoint import average_checkpoints, load_checkpoint +from icefall.env import get_env_info +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=29, + help="It specifies the checkpoint to use for decoding." + "Note: Epoch counts from 0.", + ) + parser.add_argument( + "--avg", + type=int, + default=13, + help="Number of checkpoints to average. Automatically select " + "consecutive checkpoints before the checkpoint specified by " + "'--epoch'. ", + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="transducer_stateless_multi_datasets/exp", + help="The experiment dir", + ) + + parser.add_argument( + "--bpe-model", + type=str, + default="data/lang_bpe_500/bpe.model", + help="Path to the BPE model", + ) + + parser.add_argument( + "--decoding-method", + type=str, + default="greedy_search", + help="""Possible values are: + - greedy_search + - beam_search + - modified_beam_search + """, + ) + + parser.add_argument( + "--beam-size", + type=int, + default=4, + help="""Used only when --decoding-method is + beam_search or modified_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=3, + help="""Maximum number of symbols per frame. + Used only when --decoding_method is greedy_search""", + ) + + return parser + + +def get_params() -> AttributeDict: + params = AttributeDict( + { + # parameters for conformer + "feature_dim": 80, + "encoder_out_dim": 512, + "subsampling_factor": 4, + "attention_dim": 512, + "nhead": 8, + "dim_feedforward": 2048, + "num_encoder_layers": 12, + "vgg_frontend": False, + "env_info": get_env_info(), + } + ) + return params + + +def get_encoder_model(params: AttributeDict): + # TODO: We can add an option to switch between Conformer and Transformer + encoder = Conformer( + num_features=params.feature_dim, + output_dim=params.encoder_out_dim, + subsampling_factor=params.subsampling_factor, + d_model=params.attention_dim, + nhead=params.nhead, + dim_feedforward=params.dim_feedforward, + num_encoder_layers=params.num_encoder_layers, + vgg_frontend=params.vgg_frontend, + ) + return encoder + + +def get_decoder_model(params: AttributeDict): + decoder = Decoder( + vocab_size=params.vocab_size, + embedding_dim=params.encoder_out_dim, + blank_id=params.blank_id, + context_size=params.context_size, + ) + return decoder + + +def get_joiner_model(params: AttributeDict): + joiner = Joiner( + input_dim=params.encoder_out_dim, + output_dim=params.vocab_size, + ) + return joiner + + +def get_transducer_model(params: AttributeDict): + encoder = get_encoder_model(params) + decoder = get_decoder_model(params) + joiner = get_joiner_model(params) + + model = Transducer( + encoder=encoder, + decoder=decoder, + joiner=joiner, + ) + + return model + + +def decode_one_batch( + params: AttributeDict, + model: nn.Module, + sp: spm.SentencePieceProcessor, + batch: dict, +) -> 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. + sp: + The BPE model. + batch: + It is the return value from iterating + `lhotse.dataset.K2SpeechRecognitionDataset`. See its documentation + for the format of the `batch`. + 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 = [] + 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 + ) + elif params.decoding_method == "modified_beam_search": + hyp = modified_beam_search( + model=model, encoder_out=encoder_out_i, beam=params.beam_size + ) + else: + raise ValueError( + f"Unsupported decoding method: {params.decoding_method}" + ) + hyps.append(sp.decode(hyp).split()) + + if params.decoding_method == "greedy_search": + return {"greedy_search": hyps} + else: + return {f"beam_{params.beam_size}": hyps} + + +def decode_dataset( + dl: torch.utils.data.DataLoader, + params: AttributeDict, + model: nn.Module, + sp: spm.SentencePieceProcessor, +) -> 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. + sp: + The BPE model. + 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"] + + hyps_dict = decode_one_batch( + params=params, + model=model, + sp=sp, + 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): + ref_words = ref_text.split() + this_batch.append((ref_words, 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() + AsrDataModule.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", + "modified_beam_search", + ) + params.res_dir = params.exp_dir / params.decoding_method + + params.suffix = f"epoch-{params.epoch}-avg-{params.avg}" + if "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}") + + sp = spm.SentencePieceProcessor() + sp.load(params.bpe_model) + + # is defined in local/train_bpe_model.py + params.blank_id = sp.piece_to_id("") + params.vocab_size = sp.get_piece_size() + + logging.info(params) + + logging.info("About to create model") + model = get_transducer_model(params) + + 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), strict=False + ) + + model.to(device) + model.eval() + model.device = device + + num_param = sum([p.numel() for p in model.parameters()]) + logging.info(f"Number of model parameters: {num_param}") + + asr_datamodule = AsrDataModule(args) + librispeech = LibriSpeech(manifest_dir=args.manifest_dir) + + test_clean_cuts = librispeech.test_clean_cuts() + test_other_cuts = librispeech.test_other_cuts() + + test_clean_dl = asr_datamodule.test_dataloaders(test_clean_cuts) + test_other_dl = asr_datamodule.test_dataloaders(test_other_cuts) + + test_sets = ["test-clean", "test-other"] + test_dl = [test_clean_dl, test_other_dl] + + for test_set, test_dl in zip(test_sets, test_dl): + results_dict = decode_dataset( + dl=test_dl, + params=params, + model=model, + sp=sp, + ) + + save_results( + params=params, + test_set_name=test_set, + results_dict=results_dict, + ) + + logging.info("Done!") + + +torch.set_num_threads(1) +torch.set_num_interop_threads(1) + +if __name__ == "__main__": + main() diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/decoder.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/decoder.py new file mode 100644 index 000000000..b82fed37b --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/decoder.py @@ -0,0 +1,98 @@ +# Copyright 2021 Xiaomi Corp. (authors: 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 torch +import torch.nn as nn +import torch.nn.functional as F + + +class Decoder(nn.Module): + """This class modifies the stateless decoder from the following paper: + + RNN-transducer with stateless prediction network + https://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=9054419 + + It removes the recurrent connection from the decoder, i.e., the prediction + network. Different from the above paper, it adds an extra Conv1d + right after the embedding layer. + + TODO: Implement https://arxiv.org/pdf/2109.07513.pdf + """ + + def __init__( + self, + vocab_size: int, + embedding_dim: int, + blank_id: int, + context_size: int, + ): + """ + Args: + vocab_size: + Number of tokens of the modeling unit including blank. + embedding_dim: + Dimension of the input embedding. + blank_id: + The ID of the blank symbol. + context_size: + Number of previous words to use to predict the next word. + 1 means bigram; 2 means trigram. n means (n+1)-gram. + """ + super().__init__() + self.embedding = nn.Embedding( + num_embeddings=vocab_size, + embedding_dim=embedding_dim, + padding_idx=blank_id, + ) + self.blank_id = blank_id + + assert context_size >= 1, context_size + self.context_size = context_size + if context_size > 1: + self.conv = nn.Conv1d( + in_channels=embedding_dim, + out_channels=embedding_dim, + kernel_size=context_size, + padding=0, + groups=embedding_dim, + bias=False, + ) + + def forward(self, y: torch.Tensor, need_pad: bool = True) -> torch.Tensor: + """ + Args: + y: + A 2-D tensor of shape (N, U). + need_pad: + True to left pad the input. Should be True during training. + False to not pad the input. Should be False during inference. + Returns: + Return a tensor of shape (N, U, embedding_dim). + """ + embedding_out = self.embedding(y) + if self.context_size > 1: + embedding_out = embedding_out.permute(0, 2, 1) + if need_pad is True: + embedding_out = F.pad( + embedding_out, pad=(self.context_size - 1, 0) + ) + else: + # During inference time, there is no need to do extra padding + # as we only need one output + assert embedding_out.size(-1) == self.context_size + embedding_out = self.conv(embedding_out) + embedding_out = embedding_out.permute(0, 2, 1) + return embedding_out diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/encoder_interface.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/encoder_interface.py new file mode 100644 index 000000000..257facce4 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/encoder_interface.py @@ -0,0 +1,43 @@ +# Copyright 2021 Xiaomi Corp. (authors: 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. + +from typing import Tuple + +import torch +import torch.nn as nn + + +class EncoderInterface(nn.Module): + def forward( + self, x: torch.Tensor, x_lens: torch.Tensor + ) -> Tuple[torch.Tensor, torch.Tensor]: + """ + Args: + x: + A tensor of shape (batch_size, input_seq_len, num_features) + containing the input features. + x_lens: + A tensor of shape (batch_size,) containing the number of frames + in `x` before padding. + Returns: + Return a tuple containing two tensors: + - encoder_out, a tensor of (batch_size, out_seq_len, output_dim) + containing unnormalized probabilities, i.e., the output of a + linear layer. + - encoder_out_lens, a tensor of shape (batch_size,) containing + the number of frames in `encoder_out` before padding. + """ + raise NotImplementedError("Please implement it in a subclass") diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/export.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/export.py new file mode 100755 index 000000000..7d14d011d --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/export.py @@ -0,0 +1,252 @@ +#!/usr/bin/env python3 +# +# 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: +./transducer_stateless_multi_datasets/export.py \ + --exp-dir ./transducer_stateless_multi_datasets/exp \ + --bpe-model data/lang_bpe_500/bpe.model \ + --epoch 20 \ + --avg 10 + +It will generate a file exp_dir/pretrained.pt + +To use the generated file with `transducer_stateless_multi_datasets/decode.py`, +you can do:: + + cd /path/to/exp_dir + ln -s pretrained.pt epoch-9999.pt + + cd /path/to/egs/librispeech/ASR + ./transducer_stateless_multi_datasets/decode.py \ + --exp-dir ./transducer_stateless_multi_datasets/exp \ + --epoch 9999 \ + --avg 1 \ + --max-duration 1 \ + --bpe-model data/lang_bpe_500/bpe.model +""" + +import argparse +import logging +from pathlib import Path + +import sentencepiece as spm +import torch +import torch.nn as nn +from conformer import Conformer +from decoder import Decoder +from joiner import Joiner +from model import Transducer + +from icefall.checkpoint import average_checkpoints, load_checkpoint +from icefall.env import get_env_info +from icefall.utils import AttributeDict, str2bool + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--epoch", + type=int, + default=20, + help="It specifies the checkpoint to use for decoding." + "Note: Epoch counts from 0.", + ) + + parser.add_argument( + "--avg", + type=int, + default=10, + help="Number of checkpoints to average. Automatically select " + "consecutive checkpoints before the checkpoint specified by " + "'--epoch'. ", + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="transducer_stateless_multi_datasets/exp", + help="""It specifies the directory where all training related + files, e.g., checkpoints, log, etc, are saved + """, + ) + + parser.add_argument( + "--bpe-model", + type=str, + default="data/lang_bpe_500/bpe.model", + help="Path to the BPE model", + ) + + 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 get_params() -> AttributeDict: + params = AttributeDict( + { + # parameters for conformer + "feature_dim": 80, + "encoder_out_dim": 512, + "subsampling_factor": 4, + "attention_dim": 512, + "nhead": 8, + "dim_feedforward": 2048, + "num_encoder_layers": 12, + "vgg_frontend": False, + "env_info": get_env_info(), + } + ) + return params + + +def get_encoder_model(params: AttributeDict) -> nn.Module: + encoder = Conformer( + num_features=params.feature_dim, + output_dim=params.encoder_out_dim, + subsampling_factor=params.subsampling_factor, + d_model=params.attention_dim, + nhead=params.nhead, + dim_feedforward=params.dim_feedforward, + num_encoder_layers=params.num_encoder_layers, + vgg_frontend=params.vgg_frontend, + ) + return encoder + + +def get_decoder_model(params: AttributeDict) -> nn.Module: + decoder = Decoder( + vocab_size=params.vocab_size, + embedding_dim=params.encoder_out_dim, + blank_id=params.blank_id, + context_size=params.context_size, + ) + return decoder + + +def get_joiner_model(params: AttributeDict) -> nn.Module: + joiner = Joiner( + input_dim=params.encoder_out_dim, + output_dim=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, + ) + return model + + +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}") + + sp = spm.SentencePieceProcessor() + sp.load(params.bpe_model) + + # is defined in local/train_bpe_model.py + params.blank_id = sp.piece_to_id("") + params.vocab_size = sp.get_piece_size() + + 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), strict=False + ) + + 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/librispeech/ASR/transducer_stateless_multi_datasets/gigaspeech.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/gigaspeech.py new file mode 100644 index 000000000..286771d7d --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/gigaspeech.py @@ -0,0 +1,75 @@ +# Copyright 2021 Piotr Żelasko +# 2022 Xiaomi Corp. (authors: 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 + +from lhotse import CutSet, load_manifest + + +class GigaSpeech: + def __init__(self, manifest_dir: str): + """ + Args: + manifest_dir: + It is expected to contain the following files:: + + - cuts_XL_raw.jsonl.gz + - cuts_L_raw.jsonl.gz + - cuts_M_raw.jsonl.gz + - cuts_S_raw.jsonl.gz + - cuts_XS_raw.jsonl.gz + - cuts_DEV_raw.jsonl.gz + - cuts_TEST_raw.jsonl.gz + """ + self.manifest_dir = Path(manifest_dir) + + def train_XL_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_XL_raw.jsonl.gz" + logging.info(f"About to get train-XL cuts from {f}") + return CutSet.from_jsonl_lazy(f) + + def train_L_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_L_raw.jsonl.gz" + logging.info(f"About to get train-L cuts from {f}") + return CutSet.from_jsonl_lazy(f) + + def train_M_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_M_raw.jsonl.gz" + logging.info(f"About to get train-M cuts from {f}") + return CutSet.from_jsonl_lazy(f) + + def train_S_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_S_raw.jsonl.gz" + logging.info(f"About to get train-S cuts from {f}") + return CutSet.from_jsonl_lazy(f) + + def train_XS_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_XS_raw.jsonl.gz" + logging.info(f"About to get train-XS cuts from {f}") + return CutSet.from_jsonl_lazy(f) + + def test_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_TEST.jsonl.gz" + logging.info(f"About to get TEST cuts from {f}") + return load_manifest(f) + + def dev_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_DEV.jsonl.gz" + logging.info(f"About to get DEV cuts from {f}") + return load_manifest(f) diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/joiner.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/joiner.py new file mode 100644 index 000000000..9fd9da4f1 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/joiner.py @@ -0,0 +1,72 @@ +# Copyright 2021 Xiaomi Corp. (authors: 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 torch +import torch.nn as nn + + +class Joiner(nn.Module): + def __init__(self, input_dim: int, output_dim: int): + super().__init__() + + self.input_dim = input_dim + self.output_dim = output_dim + self.output_linear = nn.Linear(input_dim, output_dim) + + def forward( + self, + encoder_out: torch.Tensor, + decoder_out: torch.Tensor, + encoder_out_len: torch.Tensor, + decoder_out_len: torch.Tensor, + ) -> torch.Tensor: + """ + Args: + encoder_out: + Output from the encoder. Its shape is (N, T, self.input_dim). + decoder_out: + Output from the decoder. Its shape is (N, U, self.input_dim). + Returns: + Return a tensor of shape (sum_all_TU, self.output_dim). + """ + assert encoder_out.ndim == decoder_out.ndim == 3 + assert encoder_out.size(0) == decoder_out.size(0) + assert encoder_out.size(2) == self.input_dim + assert decoder_out.size(2) == self.input_dim + + N = encoder_out.size(0) + + encoder_out_list = [ + encoder_out[i, : encoder_out_len[i], :] for i in range(N) + ] + + decoder_out_list = [ + decoder_out[i, : decoder_out_len[i], :] for i in range(N) + ] + + x = [ + e.unsqueeze(1) + d.unsqueeze(0) + for e, d in zip(encoder_out_list, decoder_out_list) + ] + + x = [p.reshape(-1, self.input_dim) for p in x] + x = torch.cat(x) + + activations = torch.tanh(x) + + logits = self.output_linear(activations) + + return logits diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/librispeech.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/librispeech.py new file mode 100644 index 000000000..00b7c8334 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/librispeech.py @@ -0,0 +1,74 @@ +# Copyright 2021 Piotr Żelasko +# 2022 Xiaomi Corp. (authors: 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 + +from lhotse import CutSet, load_manifest + + +class LibriSpeech: + def __init__(self, manifest_dir: str): + """ + Args: + manifest_dir: + It is expected to contain the following files:: + + - cuts_dev-clean.json.gz + - cuts_dev-other.json.gz + - cuts_test-clean.json.gz + - cuts_test-other.json.gz + - cuts_train-clean-100.json.gz + - cuts_train-clean-360.json.gz + - cuts_train-other-500.json.gz + """ + self.manifest_dir = Path(manifest_dir) + + def train_clean_100_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_train-clean-100.json.gz" + logging.info(f"About to get train-clean-100 cuts from {f}") + return load_manifest(f) + + def train_clean_360_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_train-clean-360.json.gz" + logging.info(f"About to get train-clean-360 cuts from {f}") + return load_manifest(f) + + def train_other_500_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_train-other-500.json.gz" + logging.info(f"About to get train-other-500 cuts from {f}") + return load_manifest(f) + + def test_clean_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_test-clean.json.gz" + logging.info(f"About to get test-clean cuts from {f}") + return load_manifest(f) + + def test_other_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_test-other.json.gz" + logging.info(f"About to get test-other cuts from {f}") + return load_manifest(f) + + def dev_clean_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_dev-clean.json.gz" + logging.info(f"About to get dev-clean cuts from {f}") + return load_manifest(f) + + def dev_other_cuts(self) -> CutSet: + f = self.manifest_dir / "cuts_dev-other.json.gz" + logging.info(f"About to get dev-other cuts from {f}") + return load_manifest(f) diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/model.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/model.py new file mode 100644 index 000000000..8141f9a83 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/model.py @@ -0,0 +1,168 @@ +# Copyright 2021 Xiaomi Corp. (authors: 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 random +from typing import Optional + +import k2 +import torch +import torch.nn as nn +from encoder_interface import EncoderInterface + +from icefall.utils import add_sos + + +class Transducer(nn.Module): + """It implements https://arxiv.org/pdf/1211.3711.pdf + "Sequence Transduction with Recurrent Neural Networks" + """ + + def __init__( + self, + encoder: EncoderInterface, + decoder: nn.Module, + joiner: nn.Module, + decoder_giga: Optional[nn.Module] = None, + joiner_giga: Optional[nn.Module] = None, + ): + """ + Args: + encoder: + It is the transcription network in the paper. Its accepts + two inputs: `x` of (N, T, C) and `x_lens` of shape (N,). + It returns two tensors: `logits` of shape (N, T, C) and + `logit_lens` of shape (N,). + decoder: + It is the prediction network in the paper. Its input shape + is (N, U) and its output shape is (N, U, C). It should contain + one attribute: `blank_id`. + joiner: + It has two inputs with shapes: (N, T, C) and (N, U, C). Its + output shape is (N, T, U, C). Note that its output contains + unnormalized probs, i.e., not processed by log-softmax. + decoder_giga: + The decoder for the GigaSpeech dataset. + joiner_giga: + The joiner for the GigaSpeech dataset. + """ + super().__init__() + assert isinstance(encoder, EncoderInterface), type(encoder) + assert hasattr(decoder, "blank_id") + + if decoder_giga is not None: + assert hasattr(decoder_giga, "blank_id") + + self.encoder = encoder + + self.decoder = decoder + self.joiner = joiner + + self.decoder_giga = decoder_giga + self.joiner_giga = joiner_giga + + def forward( + self, + x: torch.Tensor, + x_lens: torch.Tensor, + y: k2.RaggedTensor, + libri: bool = True, + modified_transducer_prob: float = 0.0, + ) -> torch.Tensor: + """ + Args: + x: + A 3-D tensor of shape (N, T, C). + x_lens: + A 1-D tensor of shape (N,). It contains the number of frames in `x` + before padding. + y: + A ragged tensor with 2 axes [utt][label]. It contains labels of each + utterance. + libri: + True to use the decoder and joiner for the LibriSpeech dataset. + False to use the decoder and joiner for the GigaSpeech dataset. + modified_transducer_prob: + The probability to use modified transducer loss. + Returns: + Return the transducer loss. + """ + assert x.ndim == 3, x.shape + assert x_lens.ndim == 1, x_lens.shape + assert y.num_axes == 2, y.num_axes + + assert x.size(0) == x_lens.size(0) == y.dim0 + + encoder_out, x_lens = self.encoder(x, x_lens) + assert torch.all(x_lens > 0) + + # Now for the decoder, i.e., the prediction network + row_splits = y.shape.row_splits(1) + y_lens = row_splits[1:] - row_splits[:-1] + + blank_id = self.decoder.blank_id + sos_y = add_sos(y, sos_id=blank_id) + + sos_y_padded = sos_y.pad(mode="constant", padding_value=blank_id) + sos_y_padded = sos_y_padded.to(torch.int64) + + if libri: + decoder = self.decoder + joiner = self.joiner + else: + decoder = self.decoder_giga + joiner = self.joiner_giga + + decoder_out = decoder(sos_y_padded) + + # +1 here since a blank is prepended to each utterance. + logits = joiner( + encoder_out=encoder_out, + decoder_out=decoder_out, + encoder_out_len=x_lens, + decoder_out_len=y_lens + 1, + ) + + # rnnt_loss requires 0 padded targets + # Note: y does not start with SOS + y_padded = y.pad(mode="constant", padding_value=0) + + # We don't put this `import` at the beginning of the file + # as it is required only in the training, not during the + # reference stage + import optimized_transducer + + assert 0 <= modified_transducer_prob <= 1 + + if modified_transducer_prob == 0: + one_sym_per_frame = False + elif random.random() < modified_transducer_prob: + # random.random() returns a float in the range [0, 1) + one_sym_per_frame = True + else: + one_sym_per_frame = False + + loss = optimized_transducer.transducer_loss( + logits=logits, + targets=y_padded, + logit_lengths=x_lens, + target_lengths=y_lens, + blank=blank_id, + reduction="sum", + one_sym_per_frame=one_sym_per_frame, + from_log_softmax=False, + ) + + return loss diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/pretrained.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/pretrained.py new file mode 100755 index 000000000..5ba3acea1 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/pretrained.py @@ -0,0 +1,340 @@ +#!/usr/bin/env python3 +# Copyright 2021 Xiaomi Corp. (authors: 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. +""" +Usage: + +(1) greedy search +./transducer_stateless_multi_datasets/pretrained.py \ + --checkpoint ./transducer_stateless_multi_datasets/exp/pretrained.pt \ + --bpe-model ./data/lang_bpe_500/bpe.model \ + --method greedy_search \ + --max-sym-per-frame 1 \ + /path/to/foo.wav \ + /path/to/bar.wav + +(2) beam search +./transducer_stateless_multi_datasets/pretrained.py \ + --checkpoint ./transducer_stateless_multi_datasets/exp/pretrained.pt \ + --bpe-model ./data/lang_bpe_500/bpe.model \ + --method beam_search \ + --beam-size 4 \ + /path/to/foo.wav \ + /path/to/bar.wav + +(3) modified beam search +./transducer_stateless_multi_datasets/pretrained.py \ + --checkpoint ./transducer_stateless_multi_datasets/exp/pretrained.pt \ + --bpe-model ./data/lang_bpe_500/bpe.model \ + --method modified_beam_search \ + --beam-size 4 \ + /path/to/foo.wav \ + /path/to/bar.wav + +You can also use `./transducer_stateless_multi_datasets/exp/epoch-xx.pt`. + +Note: ./transducer_stateless_multi_datasets/exp/pretrained.pt is generated by +./transducer_stateless_multi_datasets/export.py +""" + + +import argparse +import logging +import math +from typing import List + +import kaldifeat +import sentencepiece as spm +import torch +import torch.nn as nn +import torchaudio +from beam_search import beam_search, greedy_search, modified_beam_search +from conformer import Conformer +from decoder import Decoder +from joiner import Joiner +from model import Transducer +from torch.nn.utils.rnn import pad_sequence + +from icefall.env import get_env_info +from icefall.utils import AttributeDict + + +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( + "--bpe-model", + type=str, + help="""Path to bpe.model. + Used only when method is ctc-decoding. + """, + ) + + parser.add_argument( + "--method", + type=str, + default="greedy_search", + help="""Possible values are: + - greedy_search + - beam_search + - modified_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( + "--beam-size", + type=int, + default=4, + help="Used only when --method is beam_search and modified_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=3, + help="""Maximum number of symbols per frame. Used only when + --method is greedy_search. + """, + ) + + return parser + + +def get_params() -> AttributeDict: + params = AttributeDict( + { + "sample_rate": 16000, + # parameters for conformer + "feature_dim": 80, + "encoder_out_dim": 512, + "subsampling_factor": 4, + "attention_dim": 512, + "nhead": 8, + "dim_feedforward": 2048, + "num_encoder_layers": 12, + "vgg_frontend": False, + "env_info": get_env_info(), + } + ) + return params + + +def get_encoder_model(params: AttributeDict) -> nn.Module: + encoder = Conformer( + num_features=params.feature_dim, + output_dim=params.encoder_out_dim, + subsampling_factor=params.subsampling_factor, + d_model=params.attention_dim, + nhead=params.nhead, + dim_feedforward=params.dim_feedforward, + num_encoder_layers=params.num_encoder_layers, + vgg_frontend=params.vgg_frontend, + ) + return encoder + + +def get_decoder_model(params: AttributeDict) -> nn.Module: + decoder = Decoder( + vocab_size=params.vocab_size, + embedding_dim=params.encoder_out_dim, + blank_id=params.blank_id, + context_size=params.context_size, + ) + return decoder + + +def get_joiner_model(params: AttributeDict) -> nn.Module: + joiner = Joiner( + input_dim=params.encoder_out_dim, + output_dim=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, + ) + return model + + +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)) + + sp = spm.SentencePieceProcessor() + sp.load(params.bpe_model) + + # is defined in local/train_bpe_model.py + params.blank_id = sp.piece_to_id("") + params.vocab_size = sp.get_piece_size() + + 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 + + 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 + ) + + num_waves = encoder_out.size(0) + hyps = [] + msg = f"Using {params.method}" + if params.method == "beam_search": + msg += f" with beam size {params.beam_size}" + logging.info(msg) + for i in range(num_waves): + # fmt: off + encoder_out_i = encoder_out[i:i+1, :encoder_out_lens[i]] + # fmt: on + if params.method == "greedy_search": + hyp = greedy_search( + model=model, + encoder_out=encoder_out_i, + max_sym_per_frame=params.max_sym_per_frame, + ) + elif params.method == "beam_search": + hyp = beam_search( + model=model, encoder_out=encoder_out_i, beam=params.beam_size + ) + elif params.method == "modified_beam_search": + hyp = modified_beam_search( + model=model, encoder_out=encoder_out_i, beam=params.beam_size + ) + else: + raise ValueError(f"Unsupported method: {params.method}") + + hyps.append(sp.decode(hyp).split()) + + s = "\n" + for filename, hyp in zip(params.sound_files, hyps): + words = " ".join(hyp) + s += f"{filename}:\n{words}\n\n" + logging.info(s) + + logging.info("Decoding Done") + + +if __name__ == "__main__": + formatter = ( + "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s" + ) + + logging.basicConfig(format=formatter, level=logging.INFO) + main() diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/subsampling.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/subsampling.py new file mode 120000 index 000000000..73068da26 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/subsampling.py @@ -0,0 +1 @@ +../transducer/subsampling.py \ No newline at end of file diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/test_asr_datamodule.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/test_asr_datamodule.py new file mode 100755 index 000000000..e1833b841 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/test_asr_datamodule.py @@ -0,0 +1,102 @@ +#!/usr/bin/env python3 +# Copyright 2022 Xiaomi Corp. (authors: 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. + +""" +To run this file, do: + + cd icefall/egs/librispeech/ASR + python ./transducer_stateless_multi_datasets/test_asr_datamodule.py +""" + +import argparse +import random +from pathlib import Path + +from asr_datamodule import AsrDataModule +from gigaspeech import GigaSpeech +from lhotse import load_manifest +from librispeech import LibriSpeech + + +def test_dataset(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + AsrDataModule.add_arguments(parser) + args = parser.parse_args() + print(args) + + if args.enable_musan: + cuts_musan = load_manifest( + Path(args.manifest_dir) / "cuts_musan.json.gz" + ) + else: + cuts_musan = None + + librispeech = LibriSpeech(manifest_dir=args.manifest_dir) + gigaspeech = GigaSpeech(manifest_dir=args.manifest_dir) + + train_clean_100 = librispeech.train_clean_100_cuts() + train_S = gigaspeech.train_S_cuts() + + asr_datamodule = AsrDataModule(args) + + libri_train_dl = asr_datamodule.train_dataloaders( + train_clean_100, + dynamic_bucketing=False, + on_the_fly_feats=False, + cuts_musan=cuts_musan, + ) + + giga_train_dl = asr_datamodule.train_dataloaders( + train_S, + dynamic_bucketing=True, + on_the_fly_feats=True, + cuts_musan=cuts_musan, + ) + + seed = 20220216 + rng = random.Random(seed) + + for epoch in range(2): + print("epoch", epoch) + batch_idx = 0 + libri_train_dl.sampler.set_epoch(epoch) + giga_train_dl.sampler.set_epoch(epoch) + + iter_libri = iter(libri_train_dl) + iter_giga = iter(giga_train_dl) + while True: + idx = rng.choices((0, 1), weights=[0.8, 0.2], k=1)[0] + dl = iter_libri if idx == 0 else iter_giga + batch_idx += 1 + + print("dl idx", idx, "batch_idx", batch_idx) + try: + _ = next(dl) + except StopIteration: + print("dl idx", idx) + print("Go to the next epoch") + break + + +def main(): + test_dataset() + + +if __name__ == "__main__": + main() diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/test_decoder.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/test_decoder.py new file mode 100755 index 000000000..9ee197ee8 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/test_decoder.py @@ -0,0 +1,58 @@ +#!/usr/bin/env python3 +# Copyright 2021 Xiaomi Corp. (authors: 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. + +""" +To run this file, do: + + cd icefall/egs/librispeech/ASR + python ./transducer_stateless_multi_datasets/test_decoder.py +""" + +import torch +from decoder import Decoder + + +def test_decoder(): + vocab_size = 3 + blank_id = 0 + embedding_dim = 128 + context_size = 4 + + decoder = Decoder( + vocab_size=vocab_size, + embedding_dim=embedding_dim, + blank_id=blank_id, + context_size=context_size, + ) + N = 100 + U = 20 + x = torch.randint(low=0, high=vocab_size, size=(N, U)) + y = decoder(x) + assert y.shape == (N, U, embedding_dim) + + # for inference + x = torch.randint(low=0, high=vocab_size, size=(N, context_size)) + y = decoder(x, need_pad=False) + assert y.shape == (N, 1, embedding_dim) + + +def main(): + test_decoder() + + +if __name__ == "__main__": + main() diff --git a/egs/librispeech/ASR/transducer_stateless_multi_datasets/train.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/train.py new file mode 100755 index 000000000..720151ea0 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/train.py @@ -0,0 +1,890 @@ +#!/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: + +export CUDA_VISIBLE_DEVICES="0,1,2,3" + +./transducer_stateless_multi_datasets/train.py \ + --world-size 4 \ + --num-epochs 30 \ + --start-epoch 0 \ + --exp-dir transducer_stateless_multi_datasets/exp \ + --full-libri 1 \ + --max-duration 250 \ + --lr-factor 2.5 +""" + + +import argparse +import logging +import random +from pathlib import Path +from shutil import copyfile +from typing import Optional, Tuple + +import k2 +import sentencepiece as spm +import torch +import torch.multiprocessing as mp +import torch.nn as nn +from asr_datamodule import AsrDataModule +from conformer import Conformer +from decoder import Decoder +from gigaspeech import GigaSpeech +from joiner import Joiner +from lhotse import CutSet, load_manifest +from lhotse.cut import Cut +from lhotse.utils import fix_random_seed +from librispeech import LibriSpeech +from model import Transducer +from torch import Tensor +from torch.nn.parallel import DistributedDataParallel as DDP +from torch.nn.utils import clip_grad_norm_ +from torch.utils.tensorboard import SummaryWriter +from transformer import Noam + +from icefall.checkpoint import load_checkpoint +from icefall.checkpoint import save_checkpoint as save_checkpoint_impl +from icefall.dist import cleanup_dist, setup_dist +from icefall.env import get_env_info +from icefall.utils import AttributeDict, MetricsTracker, setup_logger, str2bool + + +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( + "--full-libri", + type=str2bool, + default=True, + help="When enabled, use 960h LibriSpeech. " + "Otherwise, use 100h subset.", + ) + + 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_stateless/exp/epoch-{start_epoch-1}.pt + """, + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="transducer_stateless_multi_datasets/exp", + help="""The experiment dir. + It specifies the directory where all training related + files, e.g., checkpoints, log, etc, are saved + """, + ) + + parser.add_argument( + "--bpe-model", + type=str, + default="data/lang_bpe_500/bpe.model", + help="Path to the BPE model", + ) + + parser.add_argument( + "--lr-factor", + type=float, + default=5.0, + help="The lr_factor for Noam optimizer", + ) + + 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( + "--modified-transducer-prob", + type=float, + default=0.25, + help="""The probability to use modified transducer loss. + In modified transduer, it limits the maximum number of symbols + per frame to 1. See also the option --max-sym-per-frame in + transducer_stateless/decode.py + """, + ) + + parser.add_argument( + "--giga-prob", + type=float, + default=0.2, + help="The probability to select a batch from the GigaSpeech dataset", + ) + + 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. + + - attention_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": 0, + "log_interval": 50, + "reset_interval": 200, + "valid_interval": 3000, # For the 100h subset, use 800 + # parameters for conformer + "feature_dim": 80, + "encoder_out_dim": 512, + "subsampling_factor": 4, + "attention_dim": 512, + "nhead": 8, + "dim_feedforward": 2048, + "num_encoder_layers": 12, + "vgg_frontend": False, + # parameters for Noam + "warm_step": 80000, # For the 100h subset, use 8k + "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, + output_dim=params.encoder_out_dim, + subsampling_factor=params.subsampling_factor, + d_model=params.attention_dim, + nhead=params.nhead, + dim_feedforward=params.dim_feedforward, + num_encoder_layers=params.num_encoder_layers, + vgg_frontend=params.vgg_frontend, + ) + return encoder + + +def get_decoder_model(params: AttributeDict) -> nn.Module: + decoder = Decoder( + vocab_size=params.vocab_size, + embedding_dim=params.encoder_out_dim, + blank_id=params.blank_id, + context_size=params.context_size, + ) + return decoder + + +def get_joiner_model(params: AttributeDict) -> nn.Module: + joiner = Joiner( + input_dim=params.encoder_out_dim, + output_dim=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) + + decoder_giga = get_decoder_model(params) + joiner_giga = get_joiner_model(params) + + model = Transducer( + encoder=encoder, + decoder=decoder, + joiner=joiner, + decoder_giga=decoder_giga, + joiner_giga=joiner_giga, + ) + return model + + +def load_checkpoint_if_available( + params: AttributeDict, + model: nn.Module, + optimizer: Optional[torch.optim.Optimizer] = None, + scheduler: Optional[torch.optim.lr_scheduler._LRScheduler] = None, +) -> None: + """Load checkpoint from file. + + If params.start_epoch is positive, it will load the checkpoint from + `params.start_epoch - 1`. Otherwise, this function does nothing. + + Apart from loading state dict for `model`, `optimizer` and `scheduler`, + 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 learning rate scheduler we are using. + Returns: + Return None. + """ + if params.start_epoch <= 0: + return + + filename = params.exp_dir / f"epoch-{params.start_epoch-1}.pt" + 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] + + return saved_params + + +def save_checkpoint( + params: AttributeDict, + model: nn.Module, + optimizer: Optional[torch.optim.Optimizer] = None, + scheduler: Optional[torch.optim.lr_scheduler._LRScheduler] = 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. + """ + 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, + 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 is_libri(c: Cut) -> bool: + """Return True if this cut is from the LibriSpeech dataset. + + Note: + During data preparation, we set the custom field in + the supervision segment of GigaSpeech to dict(origin='giga') + See ../local/preprocess_gigaspeech.py. + """ + return c.supervisions[0].custom is None + + +def compute_loss( + params: AttributeDict, + model: nn.Module, + sp: spm.SentencePieceProcessor, + batch: dict, + is_training: bool, +) -> 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. + """ + 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) + + libri = is_libri(supervisions["cut"][0]) + + texts = batch["supervisions"]["text"] + y = sp.encode(texts, out_type=int) + y = k2.RaggedTensor(y).to(device) + + with torch.set_grad_enabled(is_training): + loss = model( + x=feature, + x_lens=feature_lens, + y=y, + libri=libri, + modified_transducer_prob=params.modified_transducer_prob, + ) + + assert loss.requires_grad == is_training + + info = MetricsTracker() + info["frames"] = (feature_lens // params.subsampling_factor).sum().item() + + # Note: We use reduction=sum while computing the loss. + info["loss"] = loss.detach().cpu().item() + + return loss, info + + +def compute_validation_loss( + params: AttributeDict, + model: nn.Module, + sp: spm.SentencePieceProcessor, + 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, + sp=sp, + 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, + sp: spm.SentencePieceProcessor, + train_dl: torch.utils.data.DataLoader, + giga_train_dl: torch.utils.data.DataLoader, + valid_dl: torch.utils.data.DataLoader, + rng: random.Random, + tb_writer: Optional[SummaryWriter] = None, + world_size: int = 1, +) -> 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. + train_dl: + Dataloader for the training dataset. + valid_dl: + Dataloader for the validation dataset. + rng: + For select which dataset to use. + tb_writer: + Writer to write log messages to tensorboard. + world_size: + Number of nodes in DDP training. If it is 1, DDP is disabled. + """ + model.train() + + libri_tot_loss = MetricsTracker() + giga_tot_loss = MetricsTracker() + tot_loss = MetricsTracker() + + # index 0: for LibriSpeech + # index 1: for GigaSpeech + # This sets the probabilities for choosing which datasets + dl_weights = [1 - params.giga_prob, params.giga_prob] + + iter_libri = iter(train_dl) + iter_giga = iter(giga_train_dl) + + batch_idx = 0 + + while True: + idx = rng.choices((0, 1), weights=dl_weights, k=1)[0] + dl = iter_libri if idx == 0 else iter_giga + + try: + batch = next(dl) + except StopIteration: + break + + batch_idx += 1 + + params.batch_idx_train += 1 + batch_size = len(batch["supervisions"]["text"]) + + libri = is_libri(batch["supervisions"]["cut"][0]) + + loss, loss_info = compute_loss( + params=params, + model=model, + sp=sp, + batch=batch, + is_training=True, + ) + # summary stats + tot_loss = (tot_loss * (1 - 1 / params.reset_interval)) + loss_info + if libri: + libri_tot_loss = ( + libri_tot_loss * (1 - 1 / params.reset_interval) + ) + loss_info + prefix = "libri" # for logging only + else: + giga_tot_loss = ( + giga_tot_loss * (1 - 1 / params.reset_interval) + ) + loss_info + prefix = "giga" + + # NOTE: We use reduction==sum and loss is computed over utterances + # in the batch and there is no normalization to it so far. + + optimizer.zero_grad() + loss.backward() + clip_grad_norm_(model.parameters(), 5.0, 2.0) + optimizer.step() + + if batch_idx % params.log_interval == 0: + logging.info( + f"Epoch {params.cur_epoch}, " + f"batch {batch_idx}, {prefix}_loss[{loss_info}], " + f"tot_loss[{tot_loss}], " + f"libri_tot_loss[{libri_tot_loss}], " + f"giga_tot_loss[{giga_tot_loss}], " + f"batch size: {batch_size}" + ) + + if batch_idx % params.log_interval == 0: + if tb_writer is not None: + loss_info.write_summary( + tb_writer, + f"train/current_{prefix}_", + params.batch_idx_train, + ) + tot_loss.write_summary( + tb_writer, "train/tot_", params.batch_idx_train + ) + libri_tot_loss.write_summary( + tb_writer, "train/libri_tot_", params.batch_idx_train + ) + giga_tot_loss.write_summary( + tb_writer, "train/giga_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, + sp=sp, + 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 filter_short_and_long_utterances(cuts: CutSet) -> CutSet: + def remove_short_and_long_utt(c: Cut): + # Keep only utterances with duration between 1 second and 20 seconds + return 1.0 <= c.duration <= 20.0 + + num_in_total = len(cuts) + cuts = cuts.filter(remove_short_and_long_utt) + + num_left = len(cuts) + num_removed = num_in_total - num_left + removed_percent = num_removed / num_in_total * 100 + + logging.info(f"Before removing short and long utterances: {num_in_total}") + logging.info(f"After removing short and long utterances: {num_left}") + logging.info(f"Removed {num_removed} utterances ({removed_percent:.5f}%)") + + return cuts + + +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)) + if params.full_libri is False: + params.valid_interval = 800 + params.warm_step = 8000 + + seed = 42 + fix_random_seed(seed) + rng = random.Random(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}") + + sp = spm.SentencePieceProcessor() + sp.load(params.bpe_model) + + # is defined in local/train_bpe_model.py + params.blank_id = sp.piece_to_id("") + params.vocab_size = sp.get_piece_size() + + 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], find_unused_parameters=True) + model.device = device + + optimizer = Noam( + model.parameters(), + model_size=params.attention_dim, + factor=params.lr_factor, + warm_step=params.warm_step, + ) + + if checkpoints and "optimizer" in checkpoints: + logging.info("Loading optimizer state dict") + optimizer.load_state_dict(checkpoints["optimizer"]) + + librispeech = LibriSpeech(manifest_dir=args.manifest_dir) + + train_cuts = librispeech.train_clean_100_cuts() + if params.full_libri: + train_cuts += librispeech.train_clean_360_cuts() + train_cuts += librispeech.train_other_500_cuts() + + train_cuts = filter_short_and_long_utterances(train_cuts) + + gigaspeech = GigaSpeech(manifest_dir=args.manifest_dir) + # XL 10k hours + # L 2.5k hours + # M 1k hours + # S 250 hours + # XS 10 hours + # DEV 12 hours + # Test 40 hours + if params.full_libri: + logging.info("Using the L subset of GigaSpeech (2.5k hours)") + train_giga_cuts = gigaspeech.train_L_cuts() + else: + logging.info("Using the S subset of GigaSpeech (250 hours)") + train_giga_cuts = gigaspeech.train_S_cuts() + + train_giga_cuts = filter_short_and_long_utterances(train_giga_cuts) + + if args.enable_musan: + cuts_musan = load_manifest( + Path(args.manifest_dir) / "cuts_musan.json.gz" + ) + else: + cuts_musan = None + + asr_datamodule = AsrDataModule(args) + + train_dl = asr_datamodule.train_dataloaders( + train_cuts, + dynamic_bucketing=False, + on_the_fly_feats=False, + cuts_musan=cuts_musan, + ) + + giga_train_dl = asr_datamodule.train_dataloaders( + train_giga_cuts, + dynamic_bucketing=True, + on_the_fly_feats=True, + cuts_musan=cuts_musan, + ) + + valid_cuts = librispeech.dev_clean_cuts() + valid_cuts += librispeech.dev_other_cuts() + valid_dl = asr_datamodule.valid_dataloaders(valid_cuts) + + # It's time consuming to include `giga_train_dl` here + # for dl in [train_dl, giga_train_dl]: + for dl in [train_dl]: + scan_pessimistic_batches_for_oom( + model=model, + train_dl=dl, + optimizer=optimizer, + sp=sp, + params=params, + ) + + for epoch in range(params.start_epoch, params.num_epochs): + train_dl.sampler.set_epoch(epoch) + giga_train_dl.sampler.set_epoch(epoch) + + cur_lr = optimizer._rate + if tb_writer is not None: + tb_writer.add_scalar( + "train/learning_rate", cur_lr, params.batch_idx_train + ) + tb_writer.add_scalar("train/epoch", epoch, params.batch_idx_train) + + if rank == 0: + logging.info("epoch {}, learning rate {}".format(epoch, cur_lr)) + + params.cur_epoch = epoch + + train_one_epoch( + params=params, + model=model, + optimizer=optimizer, + sp=sp, + train_dl=train_dl, + giga_train_dl=giga_train_dl, + valid_dl=valid_dl, + rng=rng, + tb_writer=tb_writer, + world_size=world_size, + ) + + save_checkpoint( + params=params, + model=model, + optimizer=optimizer, + 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, + sp: spm.SentencePieceProcessor, + 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: + optimizer.zero_grad() + loss, _ = compute_loss( + params=params, + model=model, + sp=sp, + batch=batch, + is_training=True, + ) + loss.backward() + clip_grad_norm_(model.parameters(), 5.0, 2.0) + optimizer.step() + 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() + AsrDataModule.add_arguments(parser) + args = parser.parse_args() + args.exp_dir = Path(args.exp_dir) + + assert 0 <= args.giga_prob < 1, args.giga_prob + + 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/librispeech/ASR/transducer_stateless_multi_datasets/transformer.py b/egs/librispeech/ASR/transducer_stateless_multi_datasets/transformer.py new file mode 100644 index 000000000..e851dcc32 --- /dev/null +++ b/egs/librispeech/ASR/transducer_stateless_multi_datasets/transformer.py @@ -0,0 +1,418 @@ +# Copyright 2021 University of Chinese Academy of Sciences (author: Han Zhu) +# +# 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 math +from typing import Optional, Tuple + +import torch +import torch.nn as nn +from encoder_interface import EncoderInterface +from subsampling import Conv2dSubsampling, VggSubsampling + +from icefall.utils import make_pad_mask + + +class Transformer(EncoderInterface): + def __init__( + self, + num_features: int, + output_dim: int, + subsampling_factor: int = 4, + d_model: int = 256, + nhead: int = 4, + dim_feedforward: int = 2048, + num_encoder_layers: int = 12, + dropout: float = 0.1, + normalize_before: bool = True, + vgg_frontend: bool = False, + ) -> None: + """ + Args: + num_features: + The input dimension of the model. + output_dim: + The output dimension of the model. + subsampling_factor: + Number of output frames is num_in_frames // subsampling_factor. + Currently, subsampling_factor MUST be 4. + d_model: + Attention dimension. + nhead: + Number of heads in multi-head attention. + Must satisfy d_model // nhead == 0. + dim_feedforward: + The output dimension of the feedforward layers in encoder. + num_encoder_layers: + Number of encoder layers. + dropout: + Dropout in encoder. + normalize_before: + If True, use pre-layer norm; False to use post-layer norm. + vgg_frontend: + True to use vgg style frontend for subsampling. + """ + super().__init__() + + self.num_features = num_features + self.output_dim = output_dim + self.subsampling_factor = subsampling_factor + if subsampling_factor != 4: + raise NotImplementedError("Support only 'subsampling_factor=4'.") + + # self.encoder_embed converts the input of shape (N, T, num_features) + # to the shape (N, T//subsampling_factor, d_model). + # That is, it does two things simultaneously: + # (1) subsampling: T -> T//subsampling_factor + # (2) embedding: num_features -> d_model + if vgg_frontend: + self.encoder_embed = VggSubsampling(num_features, d_model) + else: + self.encoder_embed = Conv2dSubsampling(num_features, d_model) + + self.encoder_pos = PositionalEncoding(d_model, dropout) + + encoder_layer = TransformerEncoderLayer( + d_model=d_model, + nhead=nhead, + dim_feedforward=dim_feedforward, + dropout=dropout, + normalize_before=normalize_before, + ) + + if normalize_before: + encoder_norm = nn.LayerNorm(d_model) + else: + encoder_norm = None + + self.encoder = nn.TransformerEncoder( + encoder_layer=encoder_layer, + num_layers=num_encoder_layers, + norm=encoder_norm, + ) + + # TODO(fangjun): remove dropout + self.encoder_output_layer = nn.Sequential( + nn.Dropout(p=dropout), nn.Linear(d_model, output_dim) + ) + + def forward( + self, x: torch.Tensor, x_lens: torch.Tensor + ) -> Tuple[torch.Tensor, torch.Tensor]: + """ + Args: + x: + The input tensor. Its shape is (batch_size, seq_len, feature_dim). + x_lens: + A tensor of shape (batch_size,) containing the number of frames in + `x` before padding. + Returns: + Return a tuple containing 2 tensors: + - logits, its shape is (batch_size, output_seq_len, output_dim) + - logit_lens, a tensor of shape (batch_size,) containing the number + of frames in `logits` before padding. + """ + x = self.encoder_embed(x) + x = self.encoder_pos(x) + x = x.permute(1, 0, 2) # (N, T, C) -> (T, N, C) + + # Caution: We assume the subsampling factor is 4! + lengths = ((x_lens - 1) // 2 - 1) // 2 + assert x.size(0) == lengths.max().item() + + mask = make_pad_mask(lengths) + x = self.encoder(x, src_key_padding_mask=mask) # (T, N, C) + + logits = self.encoder_output_layer(x) + logits = logits.permute(1, 0, 2) # (T, N, C) ->(N, T, C) + + return logits, lengths + + +class TransformerEncoderLayer(nn.Module): + """ + Modified from torch.nn.TransformerEncoderLayer. + Add support of normalize_before, + i.e., use layer_norm before the first block. + + Args: + d_model: + the number of expected features in the input (required). + nhead: + the number of heads in the multiheadattention models (required). + dim_feedforward: + the dimension of the feedforward network model (default=2048). + dropout: + the dropout value (default=0.1). + activation: + the activation function of intermediate layer, relu or + gelu (default=relu). + normalize_before: + whether to use layer_norm before the first block. + + Examples:: + >>> encoder_layer = TransformerEncoderLayer(d_model=512, nhead=8) + >>> src = torch.rand(10, 32, 512) + >>> out = encoder_layer(src) + """ + + def __init__( + self, + d_model: int, + nhead: int, + dim_feedforward: int = 2048, + dropout: float = 0.1, + activation: str = "relu", + normalize_before: bool = True, + ) -> None: + super(TransformerEncoderLayer, self).__init__() + self.self_attn = nn.MultiheadAttention(d_model, nhead, dropout=0.0) + # Implementation of Feedforward model + self.linear1 = nn.Linear(d_model, dim_feedforward) + self.dropout = nn.Dropout(dropout) + self.linear2 = nn.Linear(dim_feedforward, d_model) + + self.norm1 = nn.LayerNorm(d_model) + self.norm2 = nn.LayerNorm(d_model) + self.dropout1 = nn.Dropout(dropout) + self.dropout2 = nn.Dropout(dropout) + + self.activation = _get_activation_fn(activation) + + self.normalize_before = normalize_before + + def __setstate__(self, state): + if "activation" not in state: + state["activation"] = nn.functional.relu + super(TransformerEncoderLayer, self).__setstate__(state) + + def forward( + self, + src: torch.Tensor, + src_mask: Optional[torch.Tensor] = None, + src_key_padding_mask: Optional[torch.Tensor] = None, + ) -> torch.Tensor: + """ + Pass the input through the encoder layer. + + Args: + src: the sequence to the encoder layer (required). + src_mask: the mask for the src sequence (optional). + src_key_padding_mask: the mask for the src keys per batch (optional) + + Shape: + src: (S, N, E). + src_mask: (S, S). + src_key_padding_mask: (N, S). + S is the source sequence length, T is the target sequence length, + N is the batch size, E is the feature number + """ + residual = src + if self.normalize_before: + src = self.norm1(src) + src2 = self.self_attn( + src, + src, + src, + attn_mask=src_mask, + key_padding_mask=src_key_padding_mask, + )[0] + src = residual + self.dropout1(src2) + if not self.normalize_before: + src = self.norm1(src) + + residual = src + if self.normalize_before: + src = self.norm2(src) + src2 = self.linear2(self.dropout(self.activation(self.linear1(src)))) + src = residual + self.dropout2(src2) + if not self.normalize_before: + src = self.norm2(src) + return src + + +def _get_activation_fn(activation: str): + if activation == "relu": + return nn.functional.relu + elif activation == "gelu": + return nn.functional.gelu + + raise RuntimeError( + "activation should be relu/gelu, not {}".format(activation) + ) + + +class PositionalEncoding(nn.Module): + """This class implements the positional encoding + proposed in the following paper: + + - Attention Is All You Need: https://arxiv.org/pdf/1706.03762.pdf + + PE(pos, 2i) = sin(pos / (10000^(2i/d_modle)) + PE(pos, 2i+1) = cos(pos / (10000^(2i/d_modle)) + + Note:: + + 1 / (10000^(2i/d_model)) = exp(-log(10000^(2i/d_model))) + = exp(-1* 2i / d_model * log(100000)) + = exp(2i * -(log(10000) / d_model)) + """ + + def __init__(self, d_model: int, dropout: float = 0.1) -> None: + """ + Args: + d_model: + Embedding dimension. + dropout: + Dropout probability to be applied to the output of this module. + """ + super().__init__() + self.d_model = d_model + self.xscale = math.sqrt(self.d_model) + self.dropout = nn.Dropout(p=dropout) + # not doing: self.pe = None because of errors thrown by torchscript + self.pe = torch.zeros(1, 0, self.d_model, dtype=torch.float32) + + def extend_pe(self, x: torch.Tensor) -> None: + """Extend the time t in the positional encoding if required. + + The shape of `self.pe` is (1, T1, d_model). The shape of the input x + is (N, T, d_model). If T > T1, then we change the shape of self.pe + to (N, T, d_model). Otherwise, nothing is done. + + Args: + x: + It is a tensor of shape (N, T, C). + Returns: + Return None. + """ + if self.pe is not None: + if self.pe.size(1) >= x.size(1): + self.pe = self.pe.to(dtype=x.dtype, device=x.device) + return + pe = torch.zeros(x.size(1), self.d_model, dtype=torch.float32) + position = torch.arange(0, x.size(1), dtype=torch.float32).unsqueeze(1) + div_term = torch.exp( + torch.arange(0, self.d_model, 2, dtype=torch.float32) + * -(math.log(10000.0) / self.d_model) + ) + pe[:, 0::2] = torch.sin(position * div_term) + pe[:, 1::2] = torch.cos(position * div_term) + pe = pe.unsqueeze(0) + # Now pe is of shape (1, T, d_model), where T is x.size(1) + self.pe = pe.to(device=x.device, dtype=x.dtype) + + def forward(self, x: torch.Tensor) -> torch.Tensor: + """ + Add positional encoding. + + Args: + x: + Its shape is (N, T, C) + + Returns: + Return a tensor of shape (N, T, C) + """ + self.extend_pe(x) + x = x * self.xscale + self.pe[:, : x.size(1), :] + return self.dropout(x) + + +class Noam(object): + """ + Implements Noam optimizer. + + Proposed in + "Attention Is All You Need", https://arxiv.org/pdf/1706.03762.pdf + + Modified from + https://github.com/espnet/espnet/blob/master/espnet/nets/pytorch_backend/transformer/optimizer.py # noqa + + Args: + params: + iterable of parameters to optimize or dicts defining parameter groups + model_size: + attention dimension of the transformer model + factor: + learning rate factor + warm_step: + warmup steps + """ + + def __init__( + self, + params, + model_size: int = 256, + factor: float = 10.0, + warm_step: int = 25000, + weight_decay=0, + ) -> None: + """Construct an Noam object.""" + self.optimizer = torch.optim.Adam( + params, lr=0, betas=(0.9, 0.98), eps=1e-9, weight_decay=weight_decay + ) + self._step = 0 + self.warmup = warm_step + self.factor = factor + self.model_size = model_size + self._rate = 0 + + @property + def param_groups(self): + """Return param_groups.""" + return self.optimizer.param_groups + + def step(self): + """Update parameters and rate.""" + self._step += 1 + rate = self.rate() + for p in self.optimizer.param_groups: + p["lr"] = rate + self._rate = rate + self.optimizer.step() + + def rate(self, step=None): + """Implement `lrate` above.""" + if step is None: + step = self._step + return ( + self.factor + * self.model_size ** (-0.5) + * min(step ** (-0.5), step * self.warmup ** (-1.5)) + ) + + def zero_grad(self): + """Reset gradient.""" + self.optimizer.zero_grad() + + def state_dict(self): + """Return state_dict.""" + return { + "_step": self._step, + "warmup": self.warmup, + "factor": self.factor, + "model_size": self.model_size, + "_rate": self._rate, + "optimizer": self.optimizer.state_dict(), + } + + def load_state_dict(self, state_dict): + """Load state_dict.""" + for key, value in state_dict.items(): + if key == "optimizer": + self.optimizer.load_state_dict(state_dict["optimizer"]) + else: + setattr(self, key, value)