diff --git a/.pre-commit-config.yaml b/.pre-commit-config.yaml index 5cb2133270..70068f9cf3 100644 --- a/.pre-commit-config.yaml +++ b/.pre-commit-config.yaml @@ -26,7 +26,7 @@ repos: # E121,E123,E126,E226,E24,E704,W503,W504 - repo: https://github.com/pycqa/isort - rev: 5.10.1 + rev: 5.12.0 hooks: - id: isort args: ["--profile=black"] diff --git a/egs/audioset/AT/README.md b/egs/audioset/AT/README.md new file mode 100644 index 0000000000..2506d41e57 --- /dev/null +++ b/egs/audioset/AT/README.md @@ -0,0 +1,12 @@ +# Introduction + +This is an audio tagging recipe for [Audioset](https://research.google.com/audioset/#/). It aims at predicting the sound events of an audio clip. + +[./RESULTS.md](./RESULTS.md) contains the latest results. + + +# Zipformer + +| Encoder | Feature type | +| --------| -------------| +| Zipformer | Frame level fbank| diff --git a/egs/audioset/AT/RESULTS.md b/egs/audioset/AT/RESULTS.md new file mode 100644 index 0000000000..0c75dfe4e3 --- /dev/null +++ b/egs/audioset/AT/RESULTS.md @@ -0,0 +1,44 @@ +## Results + +### zipformer +See for more details + +[zipformer](./zipformer) + +You can find a pretrained model, training logs, decoding logs, and decoding results at: + + +The model achieves the following mean averaged precision on AudioSet: + +| Model | mAP | +| ------ | ------- | +| Zipformer-AT | 45.1 | + +The training command is: + +```bash +export CUDA_VISIBLE_DEVICES="4,5,6,7" +subset=full + +python zipformer/train.py \ + --world-size 4 \ + --num-epochs 50 \ + --exp-dir zipformer/exp_at_as_${subset} \ + --start-epoch 1 \ + --use-fp16 1 \ + --num-events 527 \ + --audioset-subset $subset \ + --max-duration 1000 \ + --enable-musan True \ + --master-port 13455 +``` + +The evaluation command is: + +```bash +python zipformer/evaluate.py \ + --epoch 32 \ + --avg 8 \ + --exp-dir zipformer/exp_at_as_full \ + --max-duration 500 +``` diff --git a/egs/audioset/AT/local/compute_fbank_musan.py b/egs/audioset/AT/local/compute_fbank_musan.py new file mode 120000 index 0000000000..5833f2484e --- /dev/null +++ b/egs/audioset/AT/local/compute_fbank_musan.py @@ -0,0 +1 @@ +../../../librispeech/ASR/local/compute_fbank_musan.py \ No newline at end of file diff --git a/egs/audioset/AT/local/generate_audioset_manifest.py b/egs/audioset/AT/local/generate_audioset_manifest.py new file mode 100644 index 0000000000..1c5b3457cb --- /dev/null +++ b/egs/audioset/AT/local/generate_audioset_manifest.py @@ -0,0 +1,177 @@ +#!/usr/bin/env python3 +# Copyright 2023 Xiaomi Corp. (authors: Xiaoyu Yang) +# +# See ../../../../LICENSE for clarification regarding multiple authors +# +# Licensed under the Apache License, Version 2.0 (the "License"); +# you may not use this file except in compliance with the License. +# You may obtain a copy of the License at +# +# http://www.apache.org/licenses/LICENSE-2.0 +# +# Unless required by applicable law or agreed to in writing, software +# distributed under the License is distributed on an "AS IS" BASIS, +# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +# See the License for the specific language governing permissions and +# limitations under the License. + +""" +This file generates the manifest and computes the fbank features for AudioSet +dataset. The generated manifests and features are stored in data/fbank. +""" + +import argparse +import csv +import glob +import logging +import os +from typing import Dict + +import torch +from lhotse import CutSet, Fbank, FbankConfig, LilcomChunkyWriter +from lhotse.audio import Recording +from lhotse.cut import MonoCut +from lhotse.supervision import SupervisionSegment + +from icefall.utils import get_executor + +torch.set_num_threads(1) +torch.set_num_interop_threads(1) + + +def get_ID_mapping(csv_file): + # get a mapping between class ID and class name + mapping = {} + with open(csv_file, "r") as fin: + reader = csv.reader(fin, delimiter=",") + for i, row in enumerate(reader): + if i == 0: + continue + mapping[row[1]] = row[0] + return mapping + + +def parse_csv(csv_file: str, id_mapping: Dict): + # The content of the csv file shoud be something like this + # ------------------------------------------------------ + # filename label + # dataset/AudioSet/balanced/xxxx.wav 0;451 + # dataset/AudioSet/balanced/xxxy.wav 375 + # ------------------------------------------------------ + + def name2id(names): + ids = [id_mapping[name] for name in names.split(",")] + return ";".join(ids) + + mapping = {} + with open(csv_file, "r") as fin: + reader = csv.reader(fin, delimiter=" ") + for i, row in enumerate(reader): + if i <= 2: + continue + key = row[0].replace(",", "") + mapping[key] = name2id(row[-1]) + return mapping + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument("--dataset-dir", type=str, default="downloads/audioset") + + parser.add_argument( + "--split", + type=str, + default="balanced", + choices=["balanced", "unbalanced", "eval"], + ) + + parser.add_argument( + "--feat-output-dir", + type=str, + default="data/fbank", + ) + + return parser + + +def main(): + parser = get_parser() + args = parser.parse_args() + + dataset_dir = args.dataset_dir + split = args.split + feat_output_dir = args.feat_output_dir + + num_jobs = min(15, os.cpu_count()) + num_mel_bins = 80 + + if split in ["balanced", "unbalanced"]: + csv_file = f"{dataset_dir}/{split}_train_segments.csv" + elif split == "eval": + csv_file = f"{dataset_dir}/eval_segments.csv" + else: + raise ValueError() + + class_indices_csv = f"{dataset_dir}/class_labels_indices.csv" + id_mapping = get_ID_mapping(class_indices_csv) + labels = parse_csv(csv_file, id_mapping) + + audio_files = glob.glob(f"{dataset_dir}/{split}/*.wav") + + new_cuts = [] + for i, audio in enumerate(audio_files): + cut_id = audio.split("/")[-1].split("_")[0] + recording = Recording.from_file(audio, cut_id) + cut = MonoCut( + id=cut_id, + start=0.0, + duration=recording.duration, + channel=0, + recording=recording, + ) + supervision = SupervisionSegment( + id=cut_id, + recording_id=cut.recording.id, + start=0.0, + channel=0, + duration=cut.duration, + ) + try: + supervision.audio_event = labels[cut_id] + except KeyError: + logging.info(f"No labels found for {cut_id}.") + continue + cut.supervisions = [supervision] + new_cuts.append(cut) + + if i % 100 == 0 and i: + logging.info(f"Processed {i} cuts until now.") + + cuts = CutSet.from_cuts(new_cuts) + + extractor = Fbank(FbankConfig(num_mel_bins=num_mel_bins)) + + logging.info(f"Computing fbank features for {split}") + with get_executor() as ex: + cuts = cuts.compute_and_store_features( + extractor=extractor, + storage_path=f"{feat_output_dir}/{split}_feats", + num_jobs=num_jobs if ex is None else 80, + executor=ex, + storage_type=LilcomChunkyWriter, + ) + + manifest_output_dir = feat_output_dir + "/" + f"cuts_audioset_{split}.jsonl.gz" + + logging.info(f"Storing the manifest to {manifest_output_dir}") + cuts.to_jsonl(manifest_output_dir) + + +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/audioset/AT/prepare.sh b/egs/audioset/AT/prepare.sh new file mode 100755 index 0000000000..f7f73a008c --- /dev/null +++ b/egs/audioset/AT/prepare.sh @@ -0,0 +1,104 @@ +#!/usr/bin/env bash + +# fix segmentation fault reported in https://github.com/k2-fsa/icefall/issues/674 +export PROTOCOL_BUFFERS_PYTHON_IMPLEMENTATION=python + +set -eou pipefail + +# run step 0 to step 5 by default +stage=-1 +stop_stage=4 + +dl_dir=$PWD/download + +# we assume that you have your downloaded the AudioSet and placed +# it under $dl_dir/audioset, the folder structure should look like +# this: +# - $dl_dir/audioset +# - balanced +# - eval +# - unbalanced +# If you haven't downloaded the AudioSet, please refer to +# https://github.com/RicherMans/SAT/blob/main/datasets/audioset/1_download_audioset.sh. + +. shared/parse_options.sh || exit 1 + +# All files generated by this script are saved in "data". +# You can safely remove "data" and rerun this script to regenerate it. +mkdir -p data + +log() { + # This function is from espnet + local fname=${BASH_SOURCE[1]##*/} + echo -e "$(date '+%Y-%m-%d %H:%M:%S') (${fname}:${BASH_LINENO[0]}:${FUNCNAME[1]}) $*" +} + +log "Running prepare.sh" + +log "dl_dir: $dl_dir" + +if [ $stage -le -1 ] && [ $stop_stage -ge -1 ]; then + log "Stage 0: Download the necessary csv files" + if [ ! -e $dl_dir/audioset/.csv.done]; then + wget --continue "http://storage.googleapis.com/us_audioset/youtube_corpus/v1/csv/class_labels_indices.csv" -O "${dl_dir}/audioset/class_labels_indices.csv" + wget --continue http://storage.googleapis.com/us_audioset/youtube_corpus/v1/csv/balanced_train_segments.csv -O "${dl_dir}/audioset/balanced_train_segments.csv" + wget --continue http://storage.googleapis.com/us_audioset/youtube_corpus/v1/csv/eval_segments.csv -O "${dl_dir}/audioset/eval_segments.csv" + touch $dl_dir/audioset/.csv.done + fi +fi + +if [ $stage -le 0 ] && [ $stop_stage -ge 0 ]; then + log "Stage 0: Construct the audioset manifest and compute the fbank features for balanced set" + fbank_dir=data/fbank + if [! -e $fbank_dir/.balanced.done]; then + python local/generate_audioset_manifest.py \ + --dataset-dir $dl_dir/audioset \ + --split balanced \ + --feat-output-dir $fbank_dir + touch $fbank_dir/.balanced.done + fi +fi + +if [ $stage -le 1 ] && [ $stop_stage -ge 1 ]; then + log "Stage 1: Construct the audioset manifest and compute the fbank features for unbalanced set" + fbank_dir=data/fbank + if [! -e $fbank_dir/.unbalanced.done]; then + python local/generate_audioset_manifest.py \ + --dataset-dir $dl_dir/audioset \ + --split unbalanced \ + --feat-output-dir $fbank_dir + touch $fbank_dir/.unbalanced.done + fi +fi + +if [ $stage -le 2 ] && [ $stop_stage -ge 2 ]; then + log "Stage 2: Construct the audioset manifest and compute the fbank features for eval set" + fbank_dir=data/fbank + if [! -e $fbank_dir/.eval.done]; then + python local/generate_audioset_manifest.py \ + --dataset-dir $dl_dir/audioset \ + --split eval \ + --feat-output-dir $fbank_dir + touch $fbank_dir/.eval.done + fi +fi + +if [ $stage -le 3 ] && [ $stop_stage -ge 3 ]; then + log "Stage 3: Prepare musan manifest" + # We assume that you have downloaded the musan corpus + # to $dl_dir/musan + mkdir -p data/manifests + if [ ! -e data/manifests/.musan.done ]; then + lhotse prepare musan $dl_dir/musan data/manifests + touch data/manifests/.musan.done + fi +fi + +if [ $stage -le 4 ] && [ $stop_stage -ge 4 ]; then + log "Stage 4: Compute fbank for musan" + mkdir -p data/fbank + if [ ! -e data/fbank/.musan.done ]; then + ./local/compute_fbank_musan.py + touch data/fbank/.musan.done + fi +fi diff --git a/egs/audioset/AT/shared b/egs/audioset/AT/shared new file mode 120000 index 0000000000..4cbd91a7e9 --- /dev/null +++ b/egs/audioset/AT/shared @@ -0,0 +1 @@ +../../../icefall/shared \ No newline at end of file diff --git a/egs/audioset/AT/zipformer/at_datamodule.py b/egs/audioset/AT/zipformer/at_datamodule.py new file mode 100644 index 0000000000..66497c1ca6 --- /dev/null +++ b/egs/audioset/AT/zipformer/at_datamodule.py @@ -0,0 +1,420 @@ +# Copyright 2023 Xiaomi Corp. (authors: Xiaoyu Yang) +# +# See ../LICENSE for clarification regarding multiple authors +# +# Licensed under the Apache License, Version 2.0 (the "License"); +# you may not use this file except in compliance with the License. +# You may obtain a copy of the License at +# +# http://www.apache.org/licenses/LICENSE-2.0 +# +# Unless required by applicable law or agreed to in writing, software +# distributed under the License is distributed on an "AS IS" BASIS, +# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +# See the License for the specific language governing permissions and +# limitations under the License. + +import argparse +import inspect +import logging +from functools import lru_cache +from pathlib import Path +from typing import Any, Dict, Optional + +import torch +from lhotse import CutSet, Fbank, FbankConfig, load_manifest, load_manifest_lazy +from lhotse.dataset import ( # noqa F401 for PrecomputedFeatures + AudioTaggingDataset, + CutConcatenate, + CutMix, + DynamicBucketingSampler, + PrecomputedFeatures, + SimpleCutSampler, + SpecAugment, +) +from lhotse.dataset.input_strategies import ( # noqa F401 For AudioSamples + AudioSamples, + OnTheFlyFeatures, +) +from lhotse.utils import fix_random_seed +from torch.utils.data import DataLoader + +from icefall.utils import str2bool + + +class _SeedWorkers: + def __init__(self, seed: int): + self.seed = seed + + def __call__(self, worker_id: int): + fix_random_seed(self.seed + worker_id) + + +class AudioSetATDatamodule: + """ + DataModule for k2 audio tagging (AT) experiments. + + + It contains all the common data pipeline modules used in AT + experiments, e.g.: + - dynamic batch size, + - bucketing samplers, + - cut concatenation, + - augmentation, + - on-the-fly feature extraction + + This class should be derived for specific corpora used in AT tasks. + """ + + def __init__(self, args: argparse.Namespace): + self.args = args + + @classmethod + def add_arguments(cls, parser: argparse.ArgumentParser): + group = parser.add_argument_group( + title="AT 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( + "--audioset-subset", + type=str, + default="balanced", + choices=["balanced", "full"], + ) + + group.add_argument( + "--manifest-dir", + type=Path, + default=Path("data/fbank"), + help="Path to directory with audioset train/test cuts.", + ) + group.add_argument( + "--max-duration", + type=int, + default=200.0, + help="Maximum pooled recordings duration (seconds) in a " + "single batch. You can reduce it if it causes CUDA OOM.", + ) + group.add_argument( + "--bucketing-sampler", + type=str2bool, + default=True, + help="When enabled, the batches will come from buckets of " + "similar duration (saves padding frames).", + ) + group.add_argument( + "--num-buckets", + type=int, + default=30, + help="The number of buckets for the DynamicBucketingSampler" + "(you might want to increase it for larger datasets).", + ) + group.add_argument( + "--concatenate-cuts", + type=str2bool, + default=False, + help="When enabled, utterances (cuts) will be concatenated " + "to minimize the amount of padding.", + ) + group.add_argument( + "--duration-factor", + type=float, + default=1.0, + help="Determines the maximum duration of a concatenated cut " + "relative to the duration of the longest cut in a batch.", + ) + group.add_argument( + "--gap", + type=float, + default=1.0, + help="The amount of padding (in seconds) inserted between " + "concatenated cuts. This padding is filled with noise when " + "noise augmentation is used.", + ) + group.add_argument( + "--on-the-fly-feats", + type=str2bool, + default=False, + help="When enabled, use on-the-fly cut mixing and feature " + "extraction. Will drop existing precomputed feature manifests " + "if available.", + ) + group.add_argument( + "--shuffle", + type=str2bool, + default=True, + help="When enabled (=default), the examples will be " + "shuffled for each epoch.", + ) + group.add_argument( + "--drop-last", + type=str2bool, + default=True, + help="Whether to drop last batch. Used by sampler.", + ) + 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( + "--input-strategy", + type=str, + default="PrecomputedFeatures", + help="AudioSamples or PrecomputedFeatures", + ) + + def train_dataloaders( + self, + cuts_train: CutSet, + sampler_state_dict: Optional[Dict[str, Any]] = None, + ): + """ + Args: + cuts_train: + CutSet for training. + sampler_state_dict: + The state dict for the training sampler. + """ + transforms = [] + if self.args.enable_musan: + logging.info("Enable MUSAN") + logging.info("About to get Musan cuts") + cuts_musan = load_manifest(self.args.manifest_dir / "musan_cuts.jsonl.gz") + transforms.append( + CutMix(cuts=cuts_musan, p=0.5, snr=(10, 20), preserve_id=True) + ) + else: + logging.info("Disable MUSAN") + + if self.args.concatenate_cuts: + logging.info( + f"Using cut concatenation with duration factor " + f"{self.args.duration_factor} and gap {self.args.gap}." + ) + # Cut concatenation should be the first transform in the list, + # so that if we e.g. mix noise in, it will fill the gaps between + # different utterances. + transforms = [ + CutConcatenate( + duration_factor=self.args.duration_factor, gap=self.args.gap + ) + ] + transforms + + input_transforms = [] + if self.args.enable_spec_aug: + logging.info("Enable SpecAugment") + logging.info(f"Time warp factor: {self.args.spec_aug_time_warp_factor}") + # Set the value of num_frame_masks according to Lhotse's version. + # In different Lhotse's versions, the default of num_frame_masks is + # different. + num_frame_masks = 10 + num_frame_masks_parameter = inspect.signature( + SpecAugment.__init__ + ).parameters["num_frame_masks"] + if num_frame_masks_parameter.default == 1: + num_frame_masks = 2 + logging.info(f"Num frame mask: {num_frame_masks}") + input_transforms.append( + SpecAugment( + time_warp_factor=self.args.spec_aug_time_warp_factor, + num_frame_masks=num_frame_masks, + features_mask_size=27, + num_feature_masks=2, + frames_mask_size=100, + ) + ) + else: + logging.info("Disable SpecAugment") + + logging.info("About to create train dataset") + train = AudioTaggingDataset( + input_strategy=eval(self.args.input_strategy)(), + cut_transforms=transforms, + input_transforms=input_transforms, + return_cuts=self.args.return_cuts, + ) + + if self.args.on_the_fly_feats: + # NOTE: the PerturbSpeed transform should be added only if we + # remove it from data prep stage. + # Add on-the-fly speed perturbation; since originally it would + # have increased epoch size by 3, we will apply prob 2/3 and use + # 3x more epochs. + # Speed perturbation probably should come first before + # concatenation, but in principle the transforms order doesn't have + # to be strict (e.g. could be randomized) + # transforms = [PerturbSpeed(factors=[0.9, 1.1], p=2/3)] + transforms # noqa + # Drop feats to be on the safe side. + train = AudioTaggingDataset( + cut_transforms=transforms, + input_strategy=OnTheFlyFeatures(Fbank(FbankConfig(num_mel_bins=80))), + input_transforms=input_transforms, + return_cuts=self.args.return_cuts, + ) + + if self.args.bucketing_sampler: + logging.info("Using DynamicBucketingSampler.") + train_sampler = DynamicBucketingSampler( + cuts_train, + max_duration=self.args.max_duration, + shuffle=self.args.shuffle, + num_buckets=self.args.num_buckets, + drop_last=self.args.drop_last, + ) + else: + logging.info("Using SimpleCutSampler.") + train_sampler = SimpleCutSampler( + cuts_train, + max_duration=self.args.max_duration, + shuffle=self.args.shuffle, + drop_last=self.args.drop_last, + ) + logging.info("About to create train dataloader") + + if sampler_state_dict is not None: + logging.info("Loading sampler state dict") + train_sampler.load_state_dict(sampler_state_dict) + + # 'seed' is derived from the current random state, which will have + # previously been set in the main process. + seed = torch.randint(0, 100000, ()).item() + worker_init_fn = _SeedWorkers(seed) + + train_dl = DataLoader( + train, + sampler=train_sampler, + batch_size=None, + num_workers=self.args.num_workers, + persistent_workers=False, + worker_init_fn=worker_init_fn, + ) + + return train_dl + + def valid_dataloaders(self, cuts_valid: CutSet) -> DataLoader: + transforms = [] + if self.args.concatenate_cuts: + transforms = [ + CutConcatenate( + duration_factor=self.args.duration_factor, gap=self.args.gap + ) + ] + transforms + + logging.info("About to create dev dataset") + if self.args.on_the_fly_feats: + validate = AudioTaggingDataset( + cut_transforms=transforms, + input_strategy=OnTheFlyFeatures(Fbank(FbankConfig(num_mel_bins=80))), + return_cuts=self.args.return_cuts, + ) + else: + validate = AudioTaggingDataset( + cut_transforms=transforms, + return_cuts=self.args.return_cuts, + ) + valid_sampler = DynamicBucketingSampler( + 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 = AudioTaggingDataset( + input_strategy=OnTheFlyFeatures(Fbank(FbankConfig(num_mel_bins=80))) + if self.args.on_the_fly_feats + else eval(self.args.input_strategy)(), + return_cuts=self.args.return_cuts, + ) + sampler = DynamicBucketingSampler( + 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 + + @lru_cache() + def audioset_train_cuts(self) -> CutSet: + logging.info("About to get the audioset training cuts.") + balanced_cuts = load_manifest_lazy( + self.args.manifest_dir / "cuts_audioset_balanced.jsonl.gz" + ) + if self.args.audioset_subset == "full": + unbalanced_cuts = load_manifest_lazy( + self.args.manifest_dir / "cuts_audioset_unbalanced.jsonl.gz" + ) + cuts = CutSet.mux( + balanced_cuts, + unbalanced_cuts, + weights=[20000, 2000000], + stop_early=True, + ) + else: + cuts = balanced_cuts + return cuts + + @lru_cache() + def audioset_eval_cuts(self) -> CutSet: + logging.info("About to get audioset eval cuts") + return load_manifest_lazy( + self.args.manifest_dir / "cuts_audioset_eval.jsonl.gz" + ) diff --git a/egs/audioset/AT/zipformer/encoder_interface.py b/egs/audioset/AT/zipformer/encoder_interface.py new file mode 120000 index 0000000000..653c5b09af --- /dev/null +++ b/egs/audioset/AT/zipformer/encoder_interface.py @@ -0,0 +1 @@ +../../../librispeech/ASR/transducer_stateless/encoder_interface.py \ No newline at end of file diff --git a/egs/audioset/AT/zipformer/evaluate.py b/egs/audioset/AT/zipformer/evaluate.py new file mode 100644 index 0000000000..b52a284d04 --- /dev/null +++ b/egs/audioset/AT/zipformer/evaluate.py @@ -0,0 +1,344 @@ +#!/usr/bin/env python3 +# Copyright 2023 Xiaomi Corp. (authors: Xiaoyu Yang) +# +# 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" + +./zipformer/evaluate.py \ + --epoch 50 \ + --avg 10 \ + --exp-dir zipformer/exp \ + --max-duration 1000 + + +""" + +import argparse +import csv +import logging +import math +import os +from collections import defaultdict +from pathlib import Path +from typing import Dict, List, Optional, Tuple + +import k2 +import numpy as np +import sentencepiece as spm +import torch +import torch.nn as nn +import torch.nn.functional as F +from at_datamodule import AudioSetATDatamodule +from lhotse import load_manifest + +try: + from sklearn.metrics import average_precision_score +except Exception as ex: + raise RuntimeError(f"{ex}\nPlease run\n" "pip3 install -U scikit-learn") +from train import add_model_arguments, get_model, get_params, str2multihot + +from icefall.checkpoint import ( + average_checkpoints, + average_checkpoints_with_averaged_model, + find_checkpoints, + load_checkpoint, +) +from icefall.lexicon import Lexicon +from icefall.utils import ( + AttributeDict, + make_pad_mask, + setup_logger, + store_transcripts, + str2bool, + write_error_stats, +) + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--epoch", + type=int, + default=30, + help="""It specifies the checkpoint to use for decoding. + Note: Epoch counts from 1. + You can specify --avg to use more checkpoints for model averaging.""", + ) + + parser.add_argument( + "--iter", + type=int, + default=0, + help="""If positive, --epoch is ignored and it + will use the checkpoint exp_dir/checkpoint-iter.pt. + You can specify --avg to use more checkpoints for model averaging. + """, + ) + + parser.add_argument( + "--avg", + type=int, + default=15, + help="Number of checkpoints to average. Automatically select " + "consecutive checkpoints before the checkpoint specified by " + "'--epoch' and '--iter'", + ) + + parser.add_argument( + "--use-averaged-model", + type=str2bool, + default=True, + help="Whether to load averaged model. Currently it only supports " + "using --epoch. If True, it would decode with the averaged model " + "over the epoch range from `epoch-avg` (excluded) to `epoch`." + "Actually only the models with epoch number of `epoch-avg` and " + "`epoch` are loaded for averaging. ", + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="zipformer/exp", + help="The experiment dir", + ) + + add_model_arguments(parser) + + return parser + + +def inference_one_batch( + params: AttributeDict, + model: nn.Module, + batch: dict, +): + device = next(model.parameters()).device + feature = batch["inputs"] + assert feature.ndim == 3, feature.shape + + feature = feature.to(device) + # at entry, feature is (N, T, C) + + supervisions = batch["supervisions"] + audio_event = supervisions["audio_event"] + + label, _ = str2multihot(audio_event) + label = label.detach().cpu() + + feature_lens = supervisions["num_frames"].to(device) + + encoder_out, encoder_out_lens = model.forward_encoder(feature, feature_lens) + + audio_logits = model.forward_audio_tagging(encoder_out, encoder_out_lens) + # convert to probabilities between 0-1 + audio_logits = audio_logits.sigmoid().detach().cpu() + + return audio_logits, label + + +def decode_dataset( + dl: torch.utils.data.DataLoader, + params: AttributeDict, + model: nn.Module, +) -> Dict: + num_cuts = 0 + + try: + num_batches = len(dl) + except TypeError: + num_batches = "?" + + all_logits = [] + all_labels = [] + + for batch_idx, batch in enumerate(dl): + cut_ids = [cut.id for cut in batch["supervisions"]["cut"]] + num_cuts += len(cut_ids) + + audio_logits, labels = inference_one_batch( + params=params, + model=model, + batch=batch, + ) + + all_logits.append(audio_logits) + all_labels.append(labels) + + if batch_idx % 20 == 1: + logging.info(f"Processed {num_cuts} cuts already.") + logging.info("Finish collecting audio logits") + + return all_logits, all_labels + + +@torch.no_grad() +def main(): + parser = get_parser() + AudioSetATDatamodule.add_arguments(parser) + args = parser.parse_args() + args.exp_dir = Path(args.exp_dir) + + params = get_params() + params.update(vars(args)) + + params.res_dir = params.exp_dir / "inference_audio_tagging" + + if params.iter > 0: + params.suffix = f"iter-{params.iter}-avg-{params.avg}" + else: + params.suffix = f"epoch-{params.epoch}-avg-{params.avg}" + + if params.use_averaged_model: + params.suffix += "-use-averaged-model" + + setup_logger(f"{params.res_dir}/log-decode-{params.suffix}") + logging.info("Evaluation started") + + logging.info(params) + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", 0) + + logging.info("About to create model") + + model = get_model(params) + + if not params.use_averaged_model: + if params.iter > 0: + filenames = find_checkpoints(params.exp_dir, iteration=-params.iter)[ + : params.avg + ] + if len(filenames) == 0: + raise ValueError( + f"No checkpoints found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + elif len(filenames) < params.avg: + raise ValueError( + f"Not enough checkpoints ({len(filenames)}) found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + logging.info(f"averaging {filenames}") + model.to(device) + model.load_state_dict( + average_checkpoints(filenames, device=device), strict=False + ) + elif params.avg == 1: + load_checkpoint(f"{params.exp_dir}/epoch-{params.epoch}.pt", model) + else: + start = params.epoch - params.avg + 1 + filenames = [] + for i in range(start, params.epoch + 1): + if i >= 1: + 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 + ) + else: + if params.iter > 0: + filenames = find_checkpoints(params.exp_dir, iteration=-params.iter)[ + : params.avg + 1 + ] + if len(filenames) == 0: + raise ValueError( + f"No checkpoints found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + elif len(filenames) < params.avg + 1: + raise ValueError( + f"Not enough checkpoints ({len(filenames)}) found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + filename_start = filenames[-1] + filename_end = filenames[0] + logging.info( + "Calculating the averaged model over iteration checkpoints" + f" from {filename_start} (excluded) to {filename_end}" + ) + model.to(device) + model.load_state_dict( + average_checkpoints_with_averaged_model( + filename_start=filename_start, + filename_end=filename_end, + device=device, + ), + strict=False, + ) + else: + assert params.avg > 0, params.avg + start = params.epoch - params.avg + assert start >= 1, start + filename_start = f"{params.exp_dir}/epoch-{start}.pt" + filename_end = f"{params.exp_dir}/epoch-{params.epoch}.pt" + logging.info( + f"Calculating the averaged model over epoch range from " + f"{start} (excluded) to {params.epoch}" + ) + model.to(device) + model.load_state_dict( + average_checkpoints_with_averaged_model( + filename_start=filename_start, + filename_end=filename_end, + device=device, + ), + strict=False, + ) + + model.to(device) + model.eval() + + num_param = sum([p.numel() for p in model.parameters()]) + logging.info(f"Number of model parameters: {num_param}") + + args.return_cuts = True + audioset = AudioSetATDatamodule(args) + + audioset_cuts = audioset.audioset_eval_cuts() + + audioset_dl = audioset.valid_dataloaders(audioset_cuts) + + test_sets = ["audioset_eval"] + + logits, labels = decode_dataset( + dl=audioset_dl, + params=params, + model=model, + ) + + logits = torch.cat(logits, dim=0).squeeze(dim=1).detach().numpy() + labels = torch.cat(labels, dim=0).long().detach().numpy() + + # compute the metric + mAP = average_precision_score( + y_true=labels, + y_score=logits, + ) + + logging.info(f"mAP for audioset eval is: {mAP}") + + logging.info("Done") + + +if __name__ == "__main__": + main() diff --git a/egs/audioset/AT/zipformer/export-onnx.py b/egs/audioset/AT/zipformer/export-onnx.py new file mode 100755 index 0000000000..af83c0e9c6 --- /dev/null +++ b/egs/audioset/AT/zipformer/export-onnx.py @@ -0,0 +1,411 @@ +#!/usr/bin/env python3 +# +# Copyright 2023 Xiaomi Corporation (Author: Fangjun Kuang, Wei Kang) +# Copyright 2023 Danqing Fu (danqing.fu@gmail.com) + +""" +This script exports a transducer model from PyTorch to ONNX. + +We use the pre-trained model from +https://huggingface.co/Zengwei/icefall-asr-librispeech-zipformer-2023-05-15 +as an example to show how to use this file. + +1. Download the pre-trained model + +cd egs/librispeech/ASR + +repo_url=https://huggingface.co/marcoyang/icefall-audio-tagging-audioset-zipformer-2024-03-12#/ +GIT_LFS_SKIP_SMUDGE=1 git clone $repo_url +repo=$(basename $repo_url) + +pushd $repo +git lfs pull --include "exp/pretrained.pt" + +cd exp +ln -s pretrained.pt epoch-99.pt +popd + +2. Export the model to ONNX + +./zipformer/export-onnx.py \ + --use-averaged-model 0 \ + --epoch 99 \ + --avg 1 \ + --exp-dir $repo/exp \ + --num-encoder-layers "2,2,3,4,3,2" \ + --downsampling-factor "1,2,4,8,4,2" \ + --feedforward-dim "512,768,1024,1536,1024,768" \ + --num-heads "4,4,4,8,4,4" \ + --encoder-dim "192,256,384,512,384,256" \ + --query-head-dim 32 \ + --value-head-dim 12 \ + --pos-head-dim 4 \ + --pos-dim 48 \ + --encoder-unmasked-dim "192,192,256,256,256,192" \ + --cnn-module-kernel "31,31,15,15,15,31" \ + --decoder-dim 512 \ + --joiner-dim 512 \ + --causal False \ + --chunk-size "16,32,64,-1" \ + --left-context-frames "64,128,256,-1" + +It will generate the following 3 files inside $repo/exp: + + - encoder-epoch-99-avg-1.onnx + - decoder-epoch-99-avg-1.onnx + - joiner-epoch-99-avg-1.onnx + +See ./onnx_pretrained.py and ./onnx_check.py for how to +use the exported ONNX models. +""" + +import argparse +import logging +from pathlib import Path +from typing import Dict + +import k2 +import onnx +import torch +import torch.nn as nn +from onnxruntime.quantization import QuantType, quantize_dynamic +from scaling_converter import convert_scaled_to_non_scaled +from train import add_model_arguments, get_model, get_params +from zipformer import Zipformer2 + +from icefall.checkpoint import ( + average_checkpoints, + average_checkpoints_with_averaged_model, + find_checkpoints, + load_checkpoint, +) +from icefall.utils import make_pad_mask, num_tokens, str2bool + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--epoch", + type=int, + default=28, + help="""It specifies the checkpoint to use for averaging. + Note: Epoch counts from 0. + You can specify --avg to use more checkpoints for model averaging.""", + ) + + parser.add_argument( + "--iter", + type=int, + default=0, + help="""If positive, --epoch is ignored and it + will use the checkpoint exp_dir/checkpoint-iter.pt. + You can specify --avg to use more checkpoints for model averaging. + """, + ) + + parser.add_argument( + "--avg", + type=int, + default=15, + help="Number of checkpoints to average. Automatically select " + "consecutive checkpoints before the checkpoint specified by " + "'--epoch' and '--iter'", + ) + + parser.add_argument( + "--use-averaged-model", + type=str2bool, + default=True, + help="Whether to load averaged model. Currently it only supports " + "using --epoch. If True, it would decode with the averaged model " + "over the epoch range from `epoch-avg` (excluded) to `epoch`." + "Actually only the models with epoch number of `epoch-avg` and " + "`epoch` are loaded for averaging. ", + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="zipformer/exp", + help="""It specifies the directory where all training related + files, e.g., checkpoints, log, etc, are saved + """, + ) + + add_model_arguments(parser) + + return parser + + +def add_meta_data(filename: str, meta_data: Dict[str, str]): + """Add meta data to an ONNX model. It is changed in-place. + + Args: + filename: + Filename of the ONNX model to be changed. + meta_data: + Key-value pairs. + """ + model = onnx.load(filename) + for key, value in meta_data.items(): + meta = model.metadata_props.add() + meta.key = key + meta.value = value + + onnx.save(model, filename) + + +class OnnxAudioTagger(nn.Module): + """A wrapper for Zipformer audio tagger""" + + def __init__( + self, encoder: Zipformer2, encoder_embed: nn.Module, classifier: nn.Linear + ): + """ + Args: + encoder: + A Zipformer encoder. + encoder_proj: + The projection layer for encoder from the joiner. + """ + super().__init__() + self.encoder = encoder + self.encoder_embed = encoder_embed + self.classifier = classifier + + def forward( + self, + x: torch.Tensor, + x_lens: torch.Tensor, + ) -> torch.Tensor: + """Please see the help information of Zipformer.forward + + Args: + x: + A 3-D tensor of shape (N, T, C) + x_lens: + A 1-D tensor of shape (N,). Its dtype is torch.int64 + Returns: + Return a tensor containing: + - logits, A 2-D tensor of shape (N, num_classes) + + """ + x, x_lens = self.encoder_embed(x, x_lens) + src_key_padding_mask = make_pad_mask(x_lens) + x = x.permute(1, 0, 2) + encoder_out, encoder_out_lens = self.encoder(x, x_lens, src_key_padding_mask) + encoder_out = encoder_out.permute(1, 0, 2) # (N,T,C) + + logits = self.classifier(encoder_out) # (N, T, num_classes) + # Note that this is slightly different from model.py for better + # support of onnx + logits = logits.mean(dim=1) + return logits + + +def export_audio_tagging_model_onnx( + model: OnnxAudioTagger, + filename: str, + opset_version: int = 11, +) -> None: + """Export the given encoder model to ONNX format. + The exported model has two inputs: + + - x, a tensor of shape (N, T, C); dtype is torch.float32 + - x_lens, a tensor of shape (N,); dtype is torch.int64 + + and it has two outputs: + + - encoder_out, a tensor of shape (N, T', joiner_dim) + - encoder_out_lens, a tensor of shape (N,) + + Args: + model: + The input encoder model + filename: + The filename to save the exported ONNX model. + opset_version: + The opset version to use. + """ + x = torch.zeros(1, 200, 80, dtype=torch.float32) + x_lens = torch.tensor([200], dtype=torch.int64) + + model = torch.jit.trace(model, (x, x_lens)) + + torch.onnx.export( + model, + (x, x_lens), + filename, + verbose=False, + opset_version=opset_version, + input_names=["x", "x_lens"], + output_names=["logits"], + dynamic_axes={ + "x": {0: "N", 1: "T"}, + "x_lens": {0: "N"}, + "logits": {0: "N"}, + }, + ) + + meta_data = { + "model_type": "zipformer2_at", + "version": "1", + "model_author": "k2-fsa", + "comment": "zipformer2 audio tagger", + } + logging.info(f"meta_data: {meta_data}") + + add_meta_data(filename=filename, meta_data=meta_data) + + +@torch.no_grad() +def main(): + args = get_parser().parse_args() + args.exp_dir = Path(args.exp_dir) + + 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}") + + logging.info(params) + + logging.info("About to create model") + model = get_model(params) + + model.to(device) + + if not params.use_averaged_model: + if params.iter > 0: + filenames = find_checkpoints(params.exp_dir, iteration=-params.iter)[ + : params.avg + ] + if len(filenames) == 0: + raise ValueError( + f"No checkpoints found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + elif len(filenames) < params.avg: + raise ValueError( + f"Not enough checkpoints ({len(filenames)}) found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + logging.info(f"averaging {filenames}") + model.to(device) + model.load_state_dict(average_checkpoints(filenames, device=device)) + elif params.avg == 1: + load_checkpoint(f"{params.exp_dir}/epoch-{params.epoch}.pt", model) + else: + start = params.epoch - params.avg + 1 + filenames = [] + for i in range(start, params.epoch + 1): + if i >= 1: + 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)) + else: + if params.iter > 0: + filenames = find_checkpoints(params.exp_dir, iteration=-params.iter)[ + : params.avg + 1 + ] + if len(filenames) == 0: + raise ValueError( + f"No checkpoints found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + elif len(filenames) < params.avg + 1: + raise ValueError( + f"Not enough checkpoints ({len(filenames)}) found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + filename_start = filenames[-1] + filename_end = filenames[0] + logging.info( + "Calculating the averaged model over iteration checkpoints" + f" from {filename_start} (excluded) to {filename_end}" + ) + model.to(device) + model.load_state_dict( + average_checkpoints_with_averaged_model( + filename_start=filename_start, + filename_end=filename_end, + device=device, + ) + ) + else: + assert params.avg > 0, params.avg + start = params.epoch - params.avg + assert start >= 1, start + filename_start = f"{params.exp_dir}/epoch-{start}.pt" + filename_end = f"{params.exp_dir}/epoch-{params.epoch}.pt" + logging.info( + f"Calculating the averaged model over epoch range from " + f"{start} (excluded) to {params.epoch}" + ) + model.to(device) + model.load_state_dict( + average_checkpoints_with_averaged_model( + filename_start=filename_start, + filename_end=filename_end, + device=device, + ) + ) + + model.to("cpu") + model.eval() + + convert_scaled_to_non_scaled(model, inplace=True, is_onnx=True) + + model = OnnxAudioTagger( + encoder=model.encoder, + encoder_embed=model.encoder_embed, + classifier=model.classifier, + ) + + model_num_param = sum([p.numel() for p in model.parameters()]) + logging.info(f"total parameters: {model_num_param}") + + if params.iter > 0: + suffix = f"iter-{params.iter}" + else: + suffix = f"epoch-{params.epoch}" + + suffix += f"-avg-{params.avg}" + + opset_version = 13 + + logging.info("Exporting audio tagging model") + model_filename = params.exp_dir / f"model-{suffix}.onnx" + export_audio_tagging_model_onnx( + model, + model_filename, + opset_version=opset_version, + ) + logging.info(f"Exported audio tagging model to {model_filename}") + + # Generate int8 quantization models + # See https://onnxruntime.ai/docs/performance/model-optimizations/quantization.html#data-type-selection + + logging.info("Generate int8 quantization models") + + model_filename_int8 = params.exp_dir / f"model-{suffix}.int8.onnx" + quantize_dynamic( + model_input=model_filename, + model_output=model_filename_int8, + op_types_to_quantize=["MatMul"], + weight_type=QuantType.QInt8, + ) + + +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/audioset/AT/zipformer/export.py b/egs/audioset/AT/zipformer/export.py new file mode 100755 index 0000000000..bdcf8b7dd9 --- /dev/null +++ b/egs/audioset/AT/zipformer/export.py @@ -0,0 +1,339 @@ +#!/usr/bin/env python3 +# +# Copyright 2021-2023 Xiaomi Corporation (Author: Fangjun Kuang, +# Zengwei Yao, +# Wei Kang, +# Xiaoyu Yang) +# +# 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: + +Note: This is a example for librispeech dataset, if you are using different +dataset, you should change the argument values according to your dataset. + +(1) Export to torchscript model using torch.jit.script() + +./zipformer/export.py \ + --exp-dir ./zipformer/exp \ + --epoch 30 \ + --avg 9 \ + --jit 1 + +It will generate a file `jit_script.pt` in the given `exp_dir`. You can later +load it by `torch.jit.load("jit_script.pt")`. + +Check ./jit_pretrained.py for its usage. + +Check https://github.com/k2-fsa/sherpa +for how to use the exported models outside of icefall. + +(2) Export `model.state_dict()` + +./zipformer/export.py \ + --exp-dir ./zipformer/exp \ + --epoch 30 \ + --avg 9 + + +It will generate a file `pretrained.pt` in the given `exp_dir`. You can later +load it by `icefall.checkpoint.load_checkpoint()`. + +To use the generated file with `zipformer/decode.py`, +you can do: + + cd /path/to/exp_dir + ln -s pretrained.pt epoch-9999.pt + + cd /path/to/egs/librispeech/ASR + ./zipformer/evaluate.py \ + --exp-dir ./zipformer/exp \ + --use-averaged-model False \ + --epoch 9999 \ + --avg 1 \ + --max-duration 600 + +Check ./pretrained.py for its usage. + +""" + +import argparse +import logging +from pathlib import Path +from typing import Tuple + +import torch +from scaling_converter import convert_scaled_to_non_scaled +from torch import Tensor, nn +from train import add_model_arguments, get_model, get_params + +from icefall.checkpoint import ( + average_checkpoints, + average_checkpoints_with_averaged_model, + find_checkpoints, + load_checkpoint, +) +from icefall.utils import make_pad_mask, str2bool + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--epoch", + type=int, + default=30, + help="""It specifies the checkpoint to use for decoding. + Note: Epoch counts from 1. + You can specify --avg to use more checkpoints for model averaging.""", + ) + + parser.add_argument( + "--iter", + type=int, + default=0, + help="""If positive, --epoch is ignored and it + will use the checkpoint exp_dir/checkpoint-iter.pt. + You can specify --avg to use more checkpoints for model averaging. + """, + ) + + parser.add_argument( + "--avg", + type=int, + default=9, + help="Number of checkpoints to average. Automatically select " + "consecutive checkpoints before the checkpoint specified by " + "'--epoch' and '--iter'", + ) + + parser.add_argument( + "--use-averaged-model", + type=str2bool, + default=True, + help="Whether to load averaged model. Currently it only supports " + "using --epoch. If True, it would decode with the averaged model " + "over the epoch range from `epoch-avg` (excluded) to `epoch`." + "Actually only the models with epoch number of `epoch-avg` and " + "`epoch` are loaded for averaging. ", + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="zipformer/exp", + help="""It specifies the directory where all training related + files, e.g., checkpoints, log, etc, are saved + """, + ) + + parser.add_argument( + "--jit", + type=str2bool, + default=False, + help="""True to save a model after applying torch.jit.script. + It will generate a file named jit_script.pt. + Check ./jit_pretrained.py for how to use it. + """, + ) + + add_model_arguments(parser) + + return parser + + +class EncoderModel(nn.Module): + """A wrapper for encoder and encoder_embed""" + + def __init__(self, encoder: nn.Module, encoder_embed: nn.Module) -> None: + super().__init__() + self.encoder = encoder + self.encoder_embed = encoder_embed + + def forward( + self, features: Tensor, feature_lengths: Tensor + ) -> Tuple[Tensor, Tensor]: + """ + Args: + features: (N, T, C) + feature_lengths: (N,) + """ + x, x_lens = self.encoder_embed(features, feature_lengths) + + src_key_padding_mask = make_pad_mask(x_lens) + x = x.permute(1, 0, 2) # (N, T, C) -> (T, N, C) + + encoder_out, encoder_out_lens = self.encoder(x, x_lens, src_key_padding_mask) + encoder_out = encoder_out.permute(1, 0, 2) # (T, N, C) ->(N, T, C) + + return encoder_out, encoder_out_lens + + +class Classifier(nn.Module): + """A wrapper for audio tagging classifier""" + + def __init__(self, classifier: nn.Module) -> None: + super().__init__() + self.classifier = classifier + + def forward(self, encoder_out: Tensor, encoder_out_lens: Tensor): + """ + Args: + encoder_out: + A 3-D tensor of shape (N, T, C). + encoder_out_lens: + A 1-D tensor of shape (N,). It contains the number of frames in `x` + before padding. + """ + logits = self.classifier(encoder_out) # (N, T, num_classes) + padding_mask = make_pad_mask(encoder_out_lens) + logits[padding_mask] = 0 + logits = logits.sum(dim=1) # mask the padding frames + logits = logits / (~padding_mask).sum(dim=1).unsqueeze(-1).expand_as( + logits + ) # normalize the logits + + return logits + + +@torch.no_grad() +def main(): + args = get_parser().parse_args() + args.exp_dir = Path(args.exp_dir) + + params = get_params() + params.update(vars(args)) + + device = torch.device("cpu") + + logging.info(f"device: {device}") + logging.info(params) + + logging.info("About to create model") + model = get_model(params) + + if not params.use_averaged_model: + if params.iter > 0: + filenames = find_checkpoints(params.exp_dir, iteration=-params.iter)[ + : params.avg + ] + if len(filenames) == 0: + raise ValueError( + f"No checkpoints found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + elif len(filenames) < params.avg: + raise ValueError( + f"Not enough checkpoints ({len(filenames)}) found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + logging.info(f"averaging {filenames}") + model.load_state_dict(average_checkpoints(filenames, device=device)) + elif params.avg == 1: + load_checkpoint(f"{params.exp_dir}/epoch-{params.epoch}.pt", model) + else: + start = params.epoch - params.avg + 1 + filenames = [] + for i in range(start, params.epoch + 1): + if i >= 1: + filenames.append(f"{params.exp_dir}/epoch-{i}.pt") + logging.info(f"averaging {filenames}") + model.load_state_dict(average_checkpoints(filenames, device=device)) + else: + if params.iter > 0: + filenames = find_checkpoints(params.exp_dir, iteration=-params.iter)[ + : params.avg + 1 + ] + if len(filenames) == 0: + raise ValueError( + f"No checkpoints found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + elif len(filenames) < params.avg + 1: + raise ValueError( + f"Not enough checkpoints ({len(filenames)}) found for" + f" --iter {params.iter}, --avg {params.avg}" + ) + filename_start = filenames[-1] + filename_end = filenames[0] + logging.info( + "Calculating the averaged model over iteration checkpoints" + f" from {filename_start} (excluded) to {filename_end}" + ) + model.load_state_dict( + average_checkpoints_with_averaged_model( + filename_start=filename_start, + filename_end=filename_end, + device=device, + ) + ) + elif params.avg == 1: + load_checkpoint(f"{params.exp_dir}/epoch-{params.epoch}.pt", model) + else: + assert params.avg > 0, params.avg + start = params.epoch - params.avg + assert start >= 1, start + filename_start = f"{params.exp_dir}/epoch-{start}.pt" + filename_end = f"{params.exp_dir}/epoch-{params.epoch}.pt" + logging.info( + f"Calculating the averaged model over epoch range from " + f"{start} (excluded) to {params.epoch}" + ) + model.load_state_dict( + average_checkpoints_with_averaged_model( + filename_start=filename_start, + filename_end=filename_end, + device=device, + ) + ) + + model.eval() + + if params.jit is True: + convert_scaled_to_non_scaled(model, inplace=True) + # We won't use the forward() method of the model in C++, so just ignore + # it here. + # Otherwise, one of its arguments is a ragged tensor and is not + # torch scriptabe. + model.__class__.forward = torch.jit.ignore(model.__class__.forward) + + model.encoder = EncoderModel(model.encoder, model.encoder_embed) + model.classifier = Classifier(model.classifier) + filename = "jit_script.pt" + + logging.info("Using torch.jit.script") + model = torch.jit.script(model) + model.save(str(params.exp_dir / filename)) + logging.info(f"Saved to {filename}") + else: + logging.info("Not using torchscript. Export model.state_dict()") + # 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/audioset/AT/zipformer/jit_pretrained.py b/egs/audioset/AT/zipformer/jit_pretrained.py new file mode 100755 index 0000000000..8e3afcb6fc --- /dev/null +++ b/egs/audioset/AT/zipformer/jit_pretrained.py @@ -0,0 +1,181 @@ +#!/usr/bin/env python3 +# Copyright 2021-2023 Xiaomi Corporation (Author: Fangjun Kuang, Zengwei Yao) +# 2024 Xiaoyu Yang +# +# 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 loads torchscript models, exported by `torch.jit.script()` +and uses them to decode waves. +You can use the following command to get the exported models: + +./zipformer/export.py \ + --exp-dir ./zipformer/exp \ + --epoch 30 \ + --avg 9 \ + --jit 1 + +Usage of this script: + +./zipformer/jit_pretrained.py \ + --nn-model-filename ./zipformer/exp/cpu_jit.pt \ + /path/to/foo.wav \ + /path/to/bar.wav +""" + +import argparse +import csv +import logging +import math +from typing import List + +import k2 +import kaldifeat +import torch +import torchaudio +from torch.nn.utils.rnn import pad_sequence + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--nn-model-filename", + type=str, + required=True, + help="Path to the torchscript model cpu_jit.pt", + ) + + parser.add_argument( + "--label-dict", + type=str, + help="""class_labels_indices.csv.""", + ) + + 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.", + ) + + return parser + + +def read_sound_files( + filenames: List[str], expected_sample_rate: float = 16000 +) -> 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}. Given: {sample_rate}" + # We use only the first channel + ans.append(wave[0].contiguous()) + return ans + + +@torch.no_grad() +def main(): + parser = get_parser() + args = parser.parse_args() + logging.info(vars(args)) + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", 0) + + logging.info(f"device: {device}") + + model = torch.jit.load(args.nn_model_filename) + + model.eval() + + model.to(device) + + # get the label dictionary + label_dict = {} + with open(args.label_dict, "r") as f: + reader = csv.reader(f, delimiter=",") + for i, row in enumerate(reader): + if i == 0: + continue + label_dict[int(row[0])] = row[2] + + 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 = 16000 + opts.mel_opts.num_bins = 80 + opts.mel_opts.high_freq = -400 + + fbank = kaldifeat.Fbank(opts) + + logging.info(f"Reading sound files: {args.sound_files}") + waves = read_sound_files( + filenames=args.sound_files, + ) + 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) + + encoder_out, encoder_out_lens = model.encoder( + features=features, + feature_lengths=feature_lengths, + ) + + logits = model.classifier(encoder_out, encoder_out_lens) + + for filename, logit in zip(args.sound_files, logits): + topk_prob, topk_index = logit.sigmoid().topk(5) + topk_labels = [label_dict[index.item()] for index in topk_index] + logging.info( + f"{filename}: Top 5 predicted labels are {topk_labels} with probability of {topk_prob.tolist()}" + ) + + logging.info("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/audioset/AT/zipformer/model.py b/egs/audioset/AT/zipformer/model.py new file mode 100644 index 0000000000..f189eac622 --- /dev/null +++ b/egs/audioset/AT/zipformer/model.py @@ -0,0 +1,157 @@ +# Copyright 2021-2023 Xiaomi Corp. (authors: Xiaoyu Yang, +# +# 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 random +from typing import List, Optional, Tuple + +import k2 +import torch +import torch.nn as nn +import torch.nn.functional as F +from encoder_interface import EncoderInterface + +from icefall.utils import AttributeDict, make_pad_mask + + +class AudioTaggingModel(nn.Module): + def __init__( + self, + encoder_embed: nn.Module, + encoder: EncoderInterface, + encoder_dim: int = 384, + num_events: int = 527, + ): + """An audio tagging model + + Args: + encoder_embed: + It is a Convolutional 2D subsampling module. It converts + an input of shape (N, T, idim) to an output of of shape + (N, T', odim), where T' = (T-3)//2-2 = (T-7)//2. + encoder: + It is the transcription network in the paper. Its accepts + two inputs: `x` of (N, T, encoder_dim) and `x_lens` of shape (N,). + It returns two tensors: `logits` of shape (N, T, encoder_dim) and + `logit_lens` of shape (N,). + encoder_dim: + Dimension of the encoder. + num_event: + The number of classes. + """ + super().__init__() + + assert isinstance(encoder, EncoderInterface), type(encoder) + + self.encoder_embed = encoder_embed + self.encoder = encoder + self.encoder_dim = encoder_dim + + self.classifier = nn.Sequential( + nn.Dropout(0.1), + nn.Linear(encoder_dim, num_events), + ) + + # for multi-class classification + self.criterion = torch.nn.BCEWithLogitsLoss(reduction="sum") + + def forward_encoder( + self, + x: torch.Tensor, + x_lens: torch.Tensor, + ) -> Tuple[torch.Tensor, torch.Tensor]: + """Compute encoder outputs. + 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. + + Returns: + encoder_out: + Encoder output, of shape (N, T, C). + encoder_out_lens: + Encoder output lengths, of shape (N,). + """ + # logging.info(f"Memory allocated at entry: {torch.cuda.memory_allocated() // 1000000}M") + x, x_lens = self.encoder_embed(x, x_lens) + # logging.info(f"Memory allocated after encoder_embed: {torch.cuda.memory_allocated() // 1000000}M") + + src_key_padding_mask = make_pad_mask(x_lens) + x = x.permute(1, 0, 2) # (N, T, C) -> (T, N, C) + + encoder_out, encoder_out_lens = self.encoder(x, x_lens, src_key_padding_mask) + + encoder_out = encoder_out.permute(1, 0, 2) # (T, N, C) ->(N, T, C) + assert torch.all(encoder_out_lens > 0), (x_lens, encoder_out_lens) + + return encoder_out, encoder_out_lens + + def forward( + self, + x: torch.Tensor, + x_lens: torch.Tensor, + target: torch.Tensor, + ) -> Tuple[torch.Tensor, torch.Tensor, 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. + target: + The ground truth label of audio events, could be many hot + Returns: + Return the binary crossentropy loss + """ + assert x.ndim == 3, x.shape + assert x_lens.ndim == 1, x_lens.shape + + # Compute encoder outputs + encoder_out, encoder_out_lens = self.forward_encoder(x, x_lens) + + # Forward the speaker module + logits = self.forward_audio_tagging( + encoder_out=encoder_out, encoder_out_lens=encoder_out_lens + ) # (N, num_classes) + + loss = self.criterion(logits, target) + + return loss + + def forward_audio_tagging(self, encoder_out, encoder_out_lens): + """ + Args: + encoder_out: + A 3-D tensor of shape (N, T, C). + encoder_out_lens: + A 1-D tensor of shape (N,). It contains the number of frames in `x` + before padding. + + Returns: + A 3-D tensor of shape (N, num_classes). + """ + logits = self.classifier(encoder_out) # (N, T, num_classes) + padding_mask = make_pad_mask(encoder_out_lens) + logits[padding_mask] = 0 + logits = logits.sum(dim=1) # mask the padding frames + logits = logits / (~padding_mask).sum(dim=1).unsqueeze(-1).expand_as( + logits + ) # normalize the logits + + return logits diff --git a/egs/audioset/AT/zipformer/onnx_pretrained.py b/egs/audioset/AT/zipformer/onnx_pretrained.py new file mode 100755 index 0000000000..c7753715ac --- /dev/null +++ b/egs/audioset/AT/zipformer/onnx_pretrained.py @@ -0,0 +1,250 @@ +#!/usr/bin/env python3 +# Copyright 2022 Xiaomi Corp. (authors: Fangjun Kuang) +# 2022 Xiaomi Corp. (authors: Xiaoyu Yang) +# +# 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 loads ONNX models and uses them to decode waves. +You can use the following command to get the exported models: + +We use the pre-trained model from +https://huggingface.co/marcoyang/icefall-audio-tagging-audioset-zipformer-2024-03-12#/ +as an example to show how to use this file. + +1. Download the pre-trained model + +cd egs/librispeech/ASR + +repo_url=https://huggingface.co/marcoyang/icefall-audio-tagging-audioset-zipformer-2024-03-12#/ +GIT_LFS_SKIP_SMUDGE=1 git clone $repo_url +repo=$(basename $repo_url) + +pushd $repo +git lfs pull --include "exp/pretrained.pt" + +cd exp +ln -s pretrained.pt epoch-99.pt +popd + +2. Export the model to ONNX + +./zipformer/export-onnx.py \ + --use-averaged-model 0 \ + --epoch 99 \ + --avg 1 \ + --exp-dir $repo/exp \ + --causal False + +It will generate the following 3 files inside $repo/exp: + + - model-epoch-99-avg-1.onnx + +3. Run this file + +./zipformer/onnx_pretrained.py \ + --model-filename $repo/exp/model-epoch-99-avg-1.onnx \ + --tokens $repo/data/lang_bpe_500/tokens.txt \ + $repo/test_wavs/1089-134686-0001.wav \ + $repo/test_wavs/1221-135766-0001.wav \ + $repo/test_wavs/1221-135766-0002.wav +""" + +import argparse +import csv +import logging +import math +from typing import List, Tuple + +import k2 +import kaldifeat +import onnxruntime as ort +import torch +import torchaudio +from torch.nn.utils.rnn import pad_sequence + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--model-filename", + type=str, + required=True, + help="Path to the onnx model. ", + ) + + parser.add_argument( + "--label-dict", + type=str, + help="""class_labels_indices.csv.""", + ) + + parser.add_argument( + "sound_files", + type=str, + nargs="+", + help="The input sound file(s) to transcribe. " + "Supported formats are those supported by torchaudio.load(). " + "For example, wav and flac are supported. " + "The sample rate has to be 16kHz.", + ) + + parser.add_argument( + "--sample-rate", + type=int, + default=16000, + help="The sample rate of the input sound file", + ) + + return parser + + +class OnnxModel: + def __init__( + self, + nn_model: str, + ): + session_opts = ort.SessionOptions() + session_opts.inter_op_num_threads = 1 + session_opts.intra_op_num_threads = 4 + + self.session_opts = session_opts + + self.init_model(nn_model) + + def init_model(self, nn_model: str): + self.model = ort.InferenceSession( + nn_model, + sess_options=self.session_opts, + providers=["CPUExecutionProvider"], + ) + meta = self.model.get_modelmeta().custom_metadata_map + print(meta) + + def __call__( + self, + x: torch.Tensor, + x_lens: torch.Tensor, + ) -> torch.Tensor: + """ + Args: + x: + A 3-D tensor of shape (N, T, C) + x_lens: + A 2-D tensor of shape (N,). Its dtype is torch.int64 + Returns: + Return a Tensor: + - logits, its shape is (N, num_classes) + """ + out = self.model.run( + [ + self.model.get_outputs()[0].name, + ], + { + self.model.get_inputs()[0].name: x.numpy(), + self.model.get_inputs()[1].name: x_lens.numpy(), + }, + ) + return torch.from_numpy(out[0]) + + +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}. 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() + logging.info(vars(args)) + model = OnnxModel( + nn_model=args.model_filename, + ) + + # get the label dictionary + label_dict = {} + with open(args.label_dict, "r") as f: + reader = csv.reader(f, delimiter=",") + for i, row in enumerate(reader): + if i == 0: + continue + label_dict[int(row[0])] = row[2] + + logging.info("Constructing Fbank computer") + opts = kaldifeat.FbankOptions() + opts.device = "cpu" + opts.frame_opts.dither = 0 + opts.frame_opts.snip_edges = False + opts.frame_opts.samp_freq = args.sample_rate + opts.mel_opts.num_bins = 80 + opts.mel_opts.high_freq = -400 + + fbank = kaldifeat.Fbank(opts) + + logging.info(f"Reading sound files: {args.sound_files}") + waves = read_sound_files( + filenames=args.sound_files, + expected_sample_rate=args.sample_rate, + ) + + 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, dtype=torch.int64) + logits = model(features, feature_lengths) + + for filename, logit in zip(args.sound_files, logits): + topk_prob, topk_index = logit.sigmoid().topk(5) + topk_labels = [label_dict[index.item()] for index in topk_index] + logging.info( + f"{filename}: Top 5 predicted labels are {topk_labels} with probability of {topk_prob.tolist()}" + ) + + 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/audioset/AT/zipformer/optim.py b/egs/audioset/AT/zipformer/optim.py new file mode 120000 index 0000000000..5eaa3cffd4 --- /dev/null +++ b/egs/audioset/AT/zipformer/optim.py @@ -0,0 +1 @@ +../../../librispeech/ASR/zipformer/optim.py \ No newline at end of file diff --git a/egs/audioset/AT/zipformer/pretrained.py b/egs/audioset/AT/zipformer/pretrained.py new file mode 100755 index 0000000000..60e4d05182 --- /dev/null +++ b/egs/audioset/AT/zipformer/pretrained.py @@ -0,0 +1,202 @@ +#!/usr/bin/env python3 +# Copyright 2024 Xiaomi Corp. (authors: Xiaoyu Yang) +# +# 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 loads a checkpoint and uses it to decode waves. +You can generate the checkpoint with the following command: + +Note: This is a example for librispeech dataset, if you are using different +dataset, you should change the argument values according to your dataset. + + +./zipformer/export.py \ + --exp-dir ./zipformer/exp \ + --tokens data/lang_bpe_500/tokens.txt \ + --epoch 30 \ + --avg 9 + +Usage of this script: + +./zipformer/pretrained.py \ + --checkpoint ./zipformer/exp/pretrained.pt \ + /path/to/foo.wav \ + /path/to/bar.wav + + +You can also use `./zipformer/exp/epoch-xx.pt`. + +Note: ./zipformer/exp/pretrained.pt is generated by ./zipformer/export.py +""" + + +import argparse +import csv +import logging +import math +from typing import List + +import kaldifeat +import torch +import torchaudio +from torch.nn.utils.rnn import pad_sequence +from train import add_model_arguments, get_model, get_params + + +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( + "--label-dict", + type=str, + help="""class_labels_indices.csv.""", + ) + + parser.add_argument( + "sound_files", + type=str, + nargs="+", + help="The input sound file(s) to transcribe. " + "Supported formats are those supported by torchaudio.load(). " + "For example, wav and flac are supported. " + "The sample rate has to be 16kHz.", + ) + + parser.add_argument( + "--sample-rate", + type=int, + default=16000, + help="The sample rate of the input sound file", + ) + + add_model_arguments(parser) + + return parser + + +def read_sound_files( + filenames: List[str], expected_sample_rate: float +) -> List[torch.Tensor]: + """Read a list of sound files into a list 1-D float32 torch tensors. + Args: + filenames: + A list of sound filenames. + expected_sample_rate: + The expected sample rate of the sound files. + Returns: + Return a list of 1-D float32 torch tensors. + """ + ans = [] + for f in filenames: + wave, sample_rate = torchaudio.load(f) + assert ( + sample_rate == expected_sample_rate + ), f"expected sample rate: {expected_sample_rate}. Given: {sample_rate}" + # We use only the first channel + ans.append(wave[0].contiguous()) + return ans + + +@torch.no_grad() +def main(): + parser = get_parser() + args = parser.parse_args() + + params = get_params() + + params.update(vars(args)) + + 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_model(params) + + num_param = sum([p.numel() for p in model.parameters()]) + logging.info(f"Number of model parameters: {num_param}") + + checkpoint = torch.load(args.checkpoint, map_location="cpu") + model.load_state_dict(checkpoint["model"], strict=False) + model.to(device) + model.eval() + + # get the label dictionary + label_dict = {} + with open(params.label_dict, "r") as f: + reader = csv.reader(f, delimiter=",") + for i, row in enumerate(reader): + if i == 0: + continue + label_dict[int(row[0])] = row[2] + + 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 + opts.mel_opts.high_freq = -400 + + 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) + + # model forward and predict the audio events + encoder_out, encoder_out_lens = model.forward_encoder(features, feature_lengths) + logits = model.forward_audio_tagging(encoder_out, encoder_out_lens) + + for filename, logit in zip(args.sound_files, logits): + topk_prob, topk_index = logit.sigmoid().topk(5) + topk_labels = [label_dict[index.item()] for index in topk_index] + logging.info( + f"{filename}: Top 5 predicted labels are {topk_labels} with probability of {topk_prob.tolist()}" + ) + + logging.info("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/audioset/AT/zipformer/scaling.py b/egs/audioset/AT/zipformer/scaling.py new file mode 120000 index 0000000000..6f398f431d --- /dev/null +++ b/egs/audioset/AT/zipformer/scaling.py @@ -0,0 +1 @@ +../../../librispeech/ASR/zipformer/scaling.py \ No newline at end of file diff --git a/egs/audioset/AT/zipformer/scaling_converter.py b/egs/audioset/AT/zipformer/scaling_converter.py new file mode 120000 index 0000000000..b0ecee05e1 --- /dev/null +++ b/egs/audioset/AT/zipformer/scaling_converter.py @@ -0,0 +1 @@ +../../../librispeech/ASR/zipformer/scaling_converter.py \ No newline at end of file diff --git a/egs/audioset/AT/zipformer/subsampling.py b/egs/audioset/AT/zipformer/subsampling.py new file mode 120000 index 0000000000..01ae9002c6 --- /dev/null +++ b/egs/audioset/AT/zipformer/subsampling.py @@ -0,0 +1 @@ +../../../librispeech/ASR/zipformer/subsampling.py \ No newline at end of file diff --git a/egs/audioset/AT/zipformer/train.py b/egs/audioset/AT/zipformer/train.py new file mode 100644 index 0000000000..0e234c59f5 --- /dev/null +++ b/egs/audioset/AT/zipformer/train.py @@ -0,0 +1,1186 @@ +#!/usr/bin/env python3 +# Copyright 2023 Xiaomi Corp. (authors: Xiaoyu Yang) +# +# 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" + + +./zipformer/train.py \ + --world-size 4 \ + --num-epochs 30 \ + --start-epoch 1 \ + --use-fp16 1 \ + --exp-dir zipformer/exp \ + --audioset-subset full \ + --max-duration 1000 + + +""" + +import argparse +import copy +import logging +import warnings +from pathlib import Path +from shutil import copyfile +from typing import Any, Dict, List, Optional, Tuple, Union + +import optim +import sentencepiece as spm +import torch +import torch.multiprocessing as mp +import torch.nn as nn +from at_datamodule import AudioSetATDatamodule +from lhotse.cut import Cut +from lhotse.dataset.sampling.base import CutSampler +from lhotse.utils import fix_random_seed +from model import AudioTaggingModel +from optim import Eden, ScaledAdam +from scaling import ScheduledFloat +from subsampling import Conv2dSubsampling +from torch import Tensor +from torch.cuda.amp import GradScaler +from torch.nn.parallel import DistributedDataParallel as DDP +from torch.utils.tensorboard import SummaryWriter +from zipformer import Zipformer2 + +from icefall import diagnostics +from icefall.checkpoint import load_checkpoint, remove_checkpoints +from icefall.checkpoint import save_checkpoint as save_checkpoint_impl +from icefall.checkpoint import ( + save_checkpoint_with_global_batch_idx, + update_averaged_model, +) +from icefall.dist import cleanup_dist, setup_dist +from icefall.env import get_env_info +from icefall.hooks import register_inf_check_hooks +from icefall.utils import ( + AttributeDict, + MetricsTracker, + get_parameter_groups_with_lrs, + setup_logger, + str2bool, +) + +LRSchedulerType = Union[torch.optim.lr_scheduler._LRScheduler, optim.LRScheduler] + + +def get_adjusted_batch_count(params: AttributeDict) -> float: + # returns the number of batches we would have used so far if we had used the reference + # duration. This is for purposes of set_batch_count(). + return ( + params.batch_idx_train + * (params.max_duration * params.world_size) + / params.ref_duration + ) + + +def set_batch_count(model: Union[nn.Module, DDP], batch_count: float) -> None: + if isinstance(model, DDP): + # get underlying nn.Module + model = model.module + for name, module in model.named_modules(): + if hasattr(module, "batch_count"): + module.batch_count = batch_count + if hasattr(module, "name"): + module.name = name + + +def add_model_arguments(parser: argparse.ArgumentParser): + parser.add_argument( + "--num-encoder-layers", + type=str, + default="2,2,3,4,3,2", + help="Number of zipformer encoder layers per stack, comma separated.", + ) + + parser.add_argument( + "--downsampling-factor", + type=str, + default="1,2,4,8,4,2", + help="Downsampling factor for each stack of encoder layers.", + ) + + parser.add_argument( + "--feedforward-dim", + type=str, + default="512,768,1024,1536,1024,768", + help="Feedforward dimension of the zipformer encoder layers, per stack, comma separated.", + ) + + parser.add_argument( + "--num-heads", + type=str, + default="4,4,4,8,4,4", + help="Number of attention heads in the zipformer encoder layers: a single int or comma-separated list.", + ) + + parser.add_argument( + "--encoder-dim", + type=str, + default="192,256,384,512,384,256", + help="Embedding dimension in encoder stacks: a single int or comma-separated list.", + ) + + parser.add_argument( + "--query-head-dim", + type=str, + default="32", + help="Query/key dimension per head in encoder stacks: a single int or comma-separated list.", + ) + + parser.add_argument( + "--value-head-dim", + type=str, + default="12", + help="Value dimension per head in encoder stacks: a single int or comma-separated list.", + ) + + parser.add_argument( + "--pos-head-dim", + type=str, + default="4", + help="Positional-encoding dimension per head in encoder stacks: a single int or comma-separated list.", + ) + + parser.add_argument( + "--pos-dim", + type=int, + default="48", + help="Positional-encoding embedding dimension", + ) + + parser.add_argument( + "--encoder-unmasked-dim", + type=str, + default="192,192,256,256,256,192", + help="Unmasked dimensions in the encoders, relates to augmentation during training. " + "A single int or comma-separated list. Must be <= each corresponding encoder_dim.", + ) + + parser.add_argument( + "--cnn-module-kernel", + type=str, + default="31,31,15,15,15,31", + help="Sizes of convolutional kernels in convolution modules in each encoder stack: " + "a single int or comma-separated list.", + ) + + parser.add_argument( + "--causal", + type=str2bool, + default=False, + help="If True, use causal version of model. Do not recommend to use this for AT", + ) + + parser.add_argument( + "--chunk-size", + type=str, + default="16,32,64,-1", + help="Chunk sizes (at 50Hz frame rate) will be chosen randomly from this list during training. " + " Must be just -1 if --causal=False", + ) + + parser.add_argument( + "--left-context-frames", + type=str, + default="64,128,256,-1", + help="Maximum left-contexts for causal training, measured in frames which will " + "be converted to a number of chunks. If splitting into chunks, " + "chunk left-context frames will be chosen randomly from this list; else not relevant.", + ) + + parser.add_argument( + "--num-events", type=int, default=527, help="Number of sound events" + ) + + +def get_parser(): + parser = argparse.ArgumentParser( + formatter_class=argparse.ArgumentDefaultsHelpFormatter + ) + + parser.add_argument( + "--world-size", + type=int, + default=1, + help="Number of GPUs for DDP training.", + ) + + parser.add_argument( + "--master-port", + type=int, + default=12354, + help="Master port to use for DDP training.", + ) + + parser.add_argument( + "--tensorboard", + type=str2bool, + default=True, + help="Should various information be logged in tensorboard.", + ) + + parser.add_argument( + "--num-epochs", + type=int, + default=30, + help="Number of epochs to train.", + ) + + parser.add_argument( + "--start-epoch", + type=int, + default=1, + help="""Resume training from this epoch. It should be positive. + If larger than 1, it will load checkpoint from + exp-dir/epoch-{start_epoch-1}.pt + """, + ) + + parser.add_argument( + "--start-batch", + type=int, + default=0, + help="""If positive, --start-epoch is ignored and + it loads the checkpoint from exp-dir/checkpoint-{start_batch}.pt + """, + ) + + parser.add_argument( + "--exp-dir", + type=str, + default="zipformer/exp", + help="""The experiment dir. + It specifies the directory where all training related + files, e.g., checkpoints, log, etc, are saved + """, + ) + + parser.add_argument( + "--base-lr", type=float, default=0.045, help="The base learning rate." + ) + + parser.add_argument( + "--lr-batches", + type=float, + default=7500, + help="""Number of steps that affects how rapidly the learning rate + decreases. We suggest not to change this.""", + ) + + parser.add_argument( + "--lr-epochs", + type=float, + default=3.5, + help="""Number of epochs that affects how rapidly the learning rate decreases. + """, + ) + + parser.add_argument( + "--ref-duration", + type=float, + default=600, + help="Reference batch duration for purposes of adjusting batch counts for setting various " + "schedules inside the model", + ) + + parser.add_argument( + "--seed", + type=int, + default=42, + help="The seed for random generators intended for reproducibility", + ) + + parser.add_argument( + "--print-diagnostics", + type=str2bool, + default=False, + help="Accumulate stats on activations, print them and exit.", + ) + + parser.add_argument( + "--inf-check", + type=str2bool, + default=False, + help="Add hooks to check for infinite module outputs and gradients.", + ) + + parser.add_argument( + "--save-every-n", + type=int, + default=4000, + help="""Save checkpoint after processing this number of batches" + periodically. We save checkpoint to exp-dir/ whenever + params.batch_idx_train % save_every_n == 0. The checkpoint filename + has the form: f'exp-dir/checkpoint-{params.batch_idx_train}.pt' + Note: It also saves checkpoint to `exp-dir/epoch-xxx.pt` at the + end of each epoch where `xxx` is the epoch number counting from 1. + """, + ) + + parser.add_argument( + "--keep-last-k", + type=int, + default=30, + help="""Only keep this number of checkpoints on disk. + For instance, if it is 3, there are only 3 checkpoints + in the exp-dir with filenames `checkpoint-xxx.pt`. + It does not affect checkpoints with name `epoch-xxx.pt`. + """, + ) + + parser.add_argument( + "--average-period", + type=int, + default=200, + help="""Update the averaged model, namely `model_avg`, after processing + this number of batches. `model_avg` is a separate version of model, + in which each floating-point parameter is the average of all the + parameters from the start of training. Each time we take the average, + we do: `model_avg = model * (average_period / batch_idx_train) + + model_avg * ((batch_idx_train - average_period) / batch_idx_train)`. + """, + ) + + parser.add_argument( + "--use-fp16", + type=str2bool, + default=False, + help="Whether to use half precision training.", + ) + + add_model_arguments(parser) + + return parser + + +def _str2modulelist(s: str, add_dot: bool = True): + if add_dot: + return [ss.strip() + "." for ss in s.split(",")] if s is not None else None + else: + return [ss.strip() for ss in s.split(",")] if s is not None else None + + +def get_params() -> AttributeDict: + """Return a dict containing training parameters. + + All training related parameters that are not passed from the commandline + are saved in the variable `params`. + + Commandline options are merged into `params` after they are parsed, so + you can also access them via `params`. + + Explanation of options saved in `params`: + + - best_train_loss: Best training loss so far. It is used to select + the model that has the lowest training loss. It is + updated during the training. + + - best_valid_loss: Best validation loss so far. It is used to select + the model that has the lowest validation loss. It is + updated during the training. + + - best_train_epoch: It is the epoch that has the best training loss. + + - best_valid_epoch: It is the epoch that has the best validation loss. + + - batch_idx_train: Used to writing statistics to tensorboard. It + contains number of batches trained so far across + epochs. + + - log_interval: Print training loss if batch_idx % log_interval` is 0 + + - reset_interval: Reset statistics if batch_idx % reset_interval is 0 + + - valid_interval: Run validation if batch_idx % valid_interval is 0 + + - feature_dim: The model input dim. It has to match the one used + in computing features. + + - subsampling_factor: The subsampling factor for the model. + + - encoder_dim: Hidden dim for multi-head attention model. + + - num_decoder_layers: Number of decoder layer of transformer decoder. + + - warm_step: The warmup period that dictates the decay of the + scale on "simple" (un-pruned) loss. + """ + 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 zipformer + "feature_dim": 80, + "subsampling_factor": 4, # not passed in, this is fixed. + "warm_step": 2000, + "env_info": get_env_info(), + } + ) + + return params + + +def _to_int_tuple(s: str): + return tuple(map(int, s.split(","))) + + +def get_encoder_embed(params: AttributeDict) -> nn.Module: + # encoder_embed converts the input of shape (N, T, num_features) + # to the shape (N, (T - 7) // 2, encoder_dims). + # That is, it does two things simultaneously: + # (1) subsampling: T -> (T - 7) // 2 + # (2) embedding: num_features -> encoder_dims + # In the normal configuration, we will downsample once more at the end + # by a factor of 2, and most of the encoder stacks will run at a lower + # sampling rate. + encoder_embed = Conv2dSubsampling( + in_channels=params.feature_dim, + out_channels=_to_int_tuple(params.encoder_dim)[0], + dropout=ScheduledFloat((0.0, 0.3), (20000.0, 0.1)), + ) + return encoder_embed + + +def get_encoder_model(params: AttributeDict) -> nn.Module: + encoder = Zipformer2( + output_downsampling_factor=2, + downsampling_factor=_to_int_tuple(params.downsampling_factor), + num_encoder_layers=_to_int_tuple(params.num_encoder_layers), + encoder_dim=_to_int_tuple(params.encoder_dim), + encoder_unmasked_dim=_to_int_tuple(params.encoder_unmasked_dim), + query_head_dim=_to_int_tuple(params.query_head_dim), + pos_head_dim=_to_int_tuple(params.pos_head_dim), + value_head_dim=_to_int_tuple(params.value_head_dim), + pos_dim=params.pos_dim, + num_heads=_to_int_tuple(params.num_heads), + feedforward_dim=_to_int_tuple(params.feedforward_dim), + cnn_module_kernel=_to_int_tuple(params.cnn_module_kernel), + dropout=ScheduledFloat((0.0, 0.3), (20000.0, 0.1)), + warmup_batches=4000.0, + causal=params.causal, + chunk_size=_to_int_tuple(params.chunk_size), + left_context_frames=_to_int_tuple(params.left_context_frames), + ) + return encoder + + +def get_model(params: AttributeDict) -> nn.Module: + + encoder_embed = get_encoder_embed(params) + encoder = get_encoder_model(params) + + model = AudioTaggingModel( + encoder_embed=encoder_embed, + encoder=encoder, + encoder_dim=max(_to_int_tuple(params.encoder_dim)), + num_events=params.num_events, + ) + return model + + +def load_checkpoint_if_available( + params: AttributeDict, + model: nn.Module, + model_avg: nn.Module = None, + optimizer: Optional[torch.optim.Optimizer] = None, + scheduler: Optional[LRSchedulerType] = None, +) -> Optional[Dict[str, Any]]: + """Load checkpoint from file. + + If params.start_batch is positive, it will load the checkpoint from + `params.exp_dir/checkpoint-{params.start_batch}.pt`. Otherwise, if + params.start_epoch is larger than 1, it will load the checkpoint from + `params.start_epoch - 1`. + + Apart from loading state dict for `model` and `optimizer` it also updates + `best_train_epoch`, `best_train_loss`, `best_valid_epoch`, + and `best_valid_loss` in `params`. + + Args: + params: + The return value of :func:`get_params`. + model: + The training model. + model_avg: + The stored model averaged from the start of training. + optimizer: + The optimizer that we are using. + scheduler: + The scheduler that we are using. + Returns: + Return a dict containing previously saved training info. + """ + if params.start_batch > 0: + filename = params.exp_dir / f"checkpoint-{params.start_batch}.pt" + elif params.start_epoch > 1: + filename = params.exp_dir / f"epoch-{params.start_epoch-1}.pt" + else: + return None + + assert filename.is_file(), f"{filename} does not exist!" + + saved_params = load_checkpoint( + filename, + model=model, + model_avg=model_avg, + optimizer=optimizer, + scheduler=scheduler, + ) + + keys = [ + "best_train_epoch", + "best_valid_epoch", + "batch_idx_train", + "best_train_loss", + "best_valid_loss", + ] + for k in keys: + params[k] = saved_params[k] + + if params.start_batch > 0: + if "cur_epoch" in saved_params: + params["start_epoch"] = saved_params["cur_epoch"] + + return saved_params + + +def save_checkpoint( + params: AttributeDict, + model: Union[nn.Module, DDP], + model_avg: Optional[nn.Module] = None, + optimizer: Optional[torch.optim.Optimizer] = None, + scheduler: Optional[LRSchedulerType] = None, + sampler: Optional[CutSampler] = None, + scaler: Optional[GradScaler] = None, + rank: int = 0, +) -> None: + """Save model, optimizer, scheduler and training stats to file. + + Args: + params: + It is returned by :func:`get_params`. + model: + The training model. + model_avg: + The stored model averaged from the start of training. + optimizer: + The optimizer used in the training. + sampler: + The sampler for the training dataset. + scaler: + The scaler used for mix precision training. + """ + if rank != 0: + return + filename = params.exp_dir / f"epoch-{params.cur_epoch}.pt" + save_checkpoint_impl( + filename=filename, + model=model, + model_avg=model_avg, + params=params, + optimizer=optimizer, + scheduler=scheduler, + sampler=sampler, + scaler=scaler, + rank=rank, + ) + + if params.best_train_epoch == params.cur_epoch: + best_train_filename = params.exp_dir / "best-train-loss.pt" + copyfile(src=filename, dst=best_train_filename) + + if params.best_valid_epoch == params.cur_epoch: + best_valid_filename = params.exp_dir / "best-valid-loss.pt" + copyfile(src=filename, dst=best_valid_filename) + + +def compute_loss( + params: AttributeDict, + model: Union[nn.Module, DDP], + batch: dict, + is_training: bool, +) -> Tuple[Tensor, MetricsTracker]: + """ + Compute 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 Zipformer in our case. + batch: + A batch of data. See `lhotse.dataset.K2SpeechRecognitionDataset()` + for the content in it. + is_training: + True for training. False for validation. When it is True, this + function enables autograd during computation; when it is False, it + disables autograd. + warmup: a floating point value which increases throughout training; + values >= 1.0 are fully warmed up and have all modules present. + """ + device = model.device if isinstance(model, DDP) else next(model.parameters()).device + feature = batch["inputs"] + # at entry, feature is (N, T, C) + assert feature.ndim == 3 + feature = feature.to(device) + + supervisions = batch["supervisions"] + events = supervisions[ + "audio_event" + ] # the label indices are in CED format (https://github.com/RicherMans/CED) + labels, _ = str2multihot(events, n_classes=params.num_events) + labels = labels.to(device) + + feature_lens = supervisions["num_frames"].to(device) + + batch_idx_train = params.batch_idx_train + warm_step = params.warm_step + + with torch.set_grad_enabled(is_training): + loss = model( + x=feature, + x_lens=feature_lens, + target=labels, + ) + + assert loss.requires_grad == is_training + + info = MetricsTracker() + with warnings.catch_warnings(): + warnings.simplefilter("ignore") + info["frames"] = (feature_lens // params.subsampling_factor).sum().item() + + # Note: We use reduction=sum while computing the loss. + info["loss"] = loss.detach().cpu().item() + + return loss, info + + +def str2multihot(events: List[str], n_classes=527, id_mapping=None): + # Convert strings separated by semi-colon to multi-hot class labels + # input: ["0;1", "1;2"] + # output: torch.tensor([[1,1,0], [0,1,1]]) + labels = [list(map(int, event.split(";"))) for event in events] + batch_size = len(labels) + out = torch.zeros(batch_size, n_classes) + + for i, label in enumerate(labels): + if id_mapping is not None: + label = [id_mapping[lb] for lb in label] + out[i, label] = 1 + + return out, labels + + +def compute_validation_loss( + params: AttributeDict, + model: Union[nn.Module, DDP], + 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, + 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: Union[nn.Module, DDP], + optimizer: torch.optim.Optimizer, + scheduler: LRSchedulerType, + train_dl: torch.utils.data.DataLoader, + valid_dl: torch.utils.data.DataLoader, + scaler: GradScaler, + model_avg: Optional[nn.Module] = None, + tb_writer: Optional[SummaryWriter] = None, + world_size: int = 1, + rank: int = 0, +) -> None: + """Train the model for one epoch. + + The training loss from the mean of all frames is saved in + `params.train_loss`. It runs the validation process every + `params.valid_interval` batches. + + Args: + params: + It is returned by :func:`get_params`. + model: + The model for training. + optimizer: + The optimizer we are using. + scheduler: + The learning rate scheduler, we call step() every step. + train_dl: + Dataloader for the training dataset. + valid_dl: + Dataloader for the validation dataset. + scaler: + The scaler used for mix precision training. + model_avg: + The stored model averaged from the start of training. + tb_writer: + Writer to write log messages to tensorboard. + world_size: + Number of nodes in DDP training. If it is 1, DDP is disabled. + rank: + The rank of the node in DDP training. If no DDP is used, it should + be set to 0. + """ + model.train() + + tot_loss = MetricsTracker() + + saved_bad_model = False + + def save_bad_model(suffix: str = ""): + save_checkpoint_impl( + filename=params.exp_dir / f"bad-model{suffix}-{rank}.pt", + model=model, + model_avg=model_avg, + params=params, + optimizer=optimizer, + scheduler=scheduler, + sampler=train_dl.sampler, + scaler=scaler, + rank=0, + ) + + for batch_idx, batch in enumerate(train_dl): + if batch_idx % 10 == 0: + set_batch_count(model, get_adjusted_batch_count(params)) + + params.batch_idx_train += 1 + batch_size = batch["inputs"].size(0) + + try: + with torch.cuda.amp.autocast(enabled=params.use_fp16): + loss, loss_info = compute_loss( + params=params, + model=model, + batch=batch, + is_training=True, + ) + # summary stats + tot_loss = (tot_loss * (1 - 1 / params.reset_interval)) + loss_info + + # NOTE: We use reduction==sum and loss is computed over utterances + # in the batch and there is no normalization to it so far. + scaler.scale(loss).backward() + scheduler.step_batch(params.batch_idx_train) + + scaler.step(optimizer) + scaler.update() + optimizer.zero_grad() + except: # noqa + save_bad_model() + display_and_save_batch(batch, params=params) + raise + + if params.print_diagnostics and batch_idx == 5: + return + + if ( + rank == 0 + and params.batch_idx_train > 0 + and params.batch_idx_train % params.average_period == 0 + ): + update_averaged_model( + params=params, + model_cur=model, + model_avg=model_avg, + ) + + if ( + params.batch_idx_train > 0 + and params.batch_idx_train % params.save_every_n == 0 + ): + save_checkpoint_with_global_batch_idx( + out_dir=params.exp_dir, + global_batch_idx=params.batch_idx_train, + model=model, + model_avg=model_avg, + params=params, + optimizer=optimizer, + scheduler=scheduler, + sampler=train_dl.sampler, + scaler=scaler, + rank=rank, + ) + remove_checkpoints( + out_dir=params.exp_dir, + topk=params.keep_last_k, + rank=rank, + ) + + if batch_idx % 100 == 0 and params.use_fp16: + # If the grad scale was less than 1, try increasing it. The _growth_interval + # of the grad scaler is configurable, but we can't configure it to have different + # behavior depending on the current grad scale. + cur_grad_scale = scaler._scale.item() + + if cur_grad_scale < 8.0 or (cur_grad_scale < 32.0 and batch_idx % 400 == 0): + scaler.update(cur_grad_scale * 2.0) + if cur_grad_scale < 0.01: + if not saved_bad_model: + save_bad_model(suffix="-first-warning") + saved_bad_model = True + logging.warning(f"Grad scale is small: {cur_grad_scale}") + if cur_grad_scale < 1.0e-05: + save_bad_model() + raise RuntimeError( + f"grad_scale is too small, exiting: {cur_grad_scale}" + ) + + if batch_idx % params.log_interval == 0: + cur_lr = max(scheduler.get_last_lr()) + cur_grad_scale = scaler._scale.item() if params.use_fp16 else 1.0 + + logging.info( + f"Epoch {params.cur_epoch}, " + f"batch {batch_idx}, loss[{loss_info}], " + f"tot_loss[{tot_loss}], batch size: {batch_size}, " + f"lr: {cur_lr:.2e}, " + + (f"grad_scale: {scaler._scale.item()}" if params.use_fp16 else "") + ) + + if tb_writer is not None: + tb_writer.add_scalar( + "train/learning_rate", cur_lr, params.batch_idx_train + ) + + loss_info.write_summary( + tb_writer, "train/current_", params.batch_idx_train + ) + tot_loss.write_summary(tb_writer, "train/tot_", params.batch_idx_train) + if params.use_fp16: + tb_writer.add_scalar( + "train/grad_scale", cur_grad_scale, params.batch_idx_train + ) + + if batch_idx % params.valid_interval == 0 and not params.print_diagnostics: + logging.info("Computing validation loss") + valid_info = compute_validation_loss( + params=params, + model=model, + valid_dl=valid_dl, + world_size=world_size, + ) + model.train() + logging.info(f"Epoch {params.cur_epoch}, validation: {valid_info}") + logging.info( + f"Maximum memory allocated so far is {torch.cuda.max_memory_allocated()//1000000}MB" + ) + if tb_writer is not None: + valid_info.write_summary( + tb_writer, "train/valid_", params.batch_idx_train + ) + + loss_value = tot_loss["loss"] / tot_loss["frames"] + params.train_loss = loss_value + if params.train_loss < params.best_train_loss: + params.best_train_epoch = params.cur_epoch + params.best_train_loss = params.train_loss + + +def run(rank, world_size, args): + """ + Args: + rank: + It is a value between 0 and `world_size-1`, which is + passed automatically by `mp.spawn()` in :func:`main`. + The node with rank 0 is responsible for saving checkpoint. + world_size: + Number of GPUs for DDP training. + args: + The return value of get_parser().parse_args() + """ + params = get_params() + params.update(vars(args)) + + fix_random_seed(params.seed) + if world_size > 1: + setup_dist(rank, world_size, params.master_port) + + setup_logger(f"{params.exp_dir}/log/log-train") + logging.info("Training started") + + if args.tensorboard and rank == 0: + tb_writer = SummaryWriter(log_dir=f"{params.exp_dir}/tensorboard") + else: + tb_writer = None + + device = torch.device("cpu") + if torch.cuda.is_available(): + device = torch.device("cuda", rank) + logging.info(f"Device: {device}") + + logging.info(params) + + logging.info("About to create model") + model = get_model(params) + + num_param = sum([p.numel() for p in model.parameters()]) + logging.info(f"Number of model parameters: {num_param}") + + assert params.save_every_n >= params.average_period + model_avg: Optional[nn.Module] = None + if rank == 0: + # model_avg is only used with rank 0 + model_avg = copy.deepcopy(model).to(torch.float64) + + assert params.start_epoch > 0, params.start_epoch + checkpoints = load_checkpoint_if_available( + params=params, model=model, model_avg=model_avg + ) + + model.to(device) + if world_size > 1: + logging.info("Using DDP") + model = DDP(model, device_ids=[rank], find_unused_parameters=True) + + optimizer = ScaledAdam( + get_parameter_groups_with_lrs( + model, + lr=params.base_lr, + include_names=True, + ), + lr=params.base_lr, # should have no effect + clipping_scale=2.0, + ) + + scheduler = Eden(optimizer, params.lr_batches, params.lr_epochs) + + if checkpoints and "optimizer" in checkpoints: + logging.info("Loading optimizer state dict") + optimizer.load_state_dict(checkpoints["optimizer"]) + + if ( + checkpoints + and "scheduler" in checkpoints + and checkpoints["scheduler"] is not None + ): + logging.info("Loading scheduler state dict") + scheduler.load_state_dict(checkpoints["scheduler"]) + + if params.print_diagnostics: + opts = diagnostics.TensorDiagnosticOptions( + 512 + ) # allow 4 megabytes per sub-module + diagnostic = diagnostics.attach_diagnostics(model, opts) + + if params.inf_check: + register_inf_check_hooks(model) + + audioset = AudioSetATDatamodule(args) + train_cuts = audioset.audioset_train_cuts() + + def remove_short_and_long_utt(c: Cut): + # Keep only utterances with duration between 1 second and 20 seconds + # + # Caution: There is a reason to select 20.0 here. Please see + # ../local/display_manifest_statistics.py + # + # You should use ../local/display_manifest_statistics.py to get + # an utterance duration distribution for your dataset to select + # the threshold + if c.duration < 1.0 or c.duration > 30.0: + return False + + return True + + train_cuts = train_cuts.filter(remove_short_and_long_utt) + + if params.start_batch > 0 and checkpoints and "sampler" in checkpoints: + # We only load the sampler's state dict when it loads a checkpoint + # saved in the middle of an epoch + sampler_state_dict = checkpoints["sampler"] + else: + sampler_state_dict = None + + train_dl = audioset.train_dataloaders( + train_cuts, sampler_state_dict=sampler_state_dict + ) + + valid_cuts = audioset.audioset_eval_cuts() + valid_dl = audioset.valid_dataloaders(valid_cuts) + + scaler = GradScaler(enabled=params.use_fp16, init_scale=1.0) + if checkpoints and "grad_scaler" in checkpoints: + logging.info("Loading grad scaler state dict") + scaler.load_state_dict(checkpoints["grad_scaler"]) + + for epoch in range(params.start_epoch, params.num_epochs + 1): + scheduler.step_epoch(epoch - 1) + fix_random_seed(params.seed + epoch - 1) + train_dl.sampler.set_epoch(epoch - 1) + + if tb_writer is not None: + tb_writer.add_scalar("train/epoch", epoch, params.batch_idx_train) + + params.cur_epoch = epoch + + train_one_epoch( + params=params, + model=model, + model_avg=model_avg, + optimizer=optimizer, + scheduler=scheduler, + train_dl=train_dl, + valid_dl=valid_dl, + scaler=scaler, + tb_writer=tb_writer, + world_size=world_size, + rank=rank, + ) + + if params.print_diagnostics: + diagnostic.print_diagnostics() + break + + save_checkpoint( + params=params, + model=model, + model_avg=model_avg, + optimizer=optimizer, + scheduler=scheduler, + sampler=train_dl.sampler, + scaler=scaler, + rank=rank, + ) + + logging.info("Done!") + + if world_size > 1: + torch.distributed.barrier() + cleanup_dist() + + +def display_and_save_batch( + batch: dict, + params: AttributeDict, +) -> None: + """Display the batch statistics and save the batch into disk. + + Args: + batch: + A batch of data. See `lhotse.dataset.K2SpeechRecognitionDataset()` + for the content in it. + params: + Parameters for training. See :func:`get_params`. + """ + from lhotse.utils import uuid4 + + filename = f"{params.exp_dir}/batch-{uuid4()}.pt" + logging.info(f"Saving batch to {filename}") + torch.save(batch, filename) + + supervisions = batch["supervisions"] + features = batch["inputs"] + + logging.info(f"features shape: {features.shape}") + + +def scan_pessimistic_batches_for_oom( + model: Union[nn.Module, DDP], + train_dl: torch.utils.data.DataLoader, + optimizer: torch.optim.Optimizer, + params: AttributeDict, +): + from lhotse.dataset import find_pessimistic_batches + + logging.info( + "Sanity check -- see if any of the batches in epoch 1 would cause OOM." + ) + batches, crit_values = find_pessimistic_batches(train_dl.sampler) + for criterion, cuts in batches.items(): + batch = train_dl.dataset[cuts] + try: + with torch.cuda.amp.autocast(enabled=params.use_fp16): + loss, _ = compute_loss( + params=params, + model=model, + batch=batch, + is_training=True, + ) + loss.backward() + optimizer.zero_grad() + except Exception 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]}) ..." + ) + display_and_save_batch( + batch, + params=params, + ) + raise + logging.info( + f"Maximum memory allocated so far is {torch.cuda.max_memory_allocated()//1000000}MB" + ) + + +def main(): + parser = get_parser() + AudioSetATDatamodule.add_arguments(parser) + args = parser.parse_args() + args.exp_dir = Path(args.exp_dir) + + world_size = args.world_size + assert world_size >= 1 + if world_size > 1: + mp.spawn(run, args=(world_size, args), nprocs=world_size, join=True) + else: + run(rank=0, world_size=1, args=args) + + +if __name__ == "__main__": + main() diff --git a/egs/audioset/AT/zipformer/zipformer.py b/egs/audioset/AT/zipformer/zipformer.py new file mode 120000 index 0000000000..23011dda71 --- /dev/null +++ b/egs/audioset/AT/zipformer/zipformer.py @@ -0,0 +1 @@ +../../../librispeech/ASR/zipformer/zipformer.py \ No newline at end of file