Datasets:
Languages:
English
ArXiv:
Tags:
query-by-example-spoken-term-detection
audio-slot-filling
speaker-diarization
automatic-speaker-verification
License:
# coding=utf-8 | |
# Copyright 2021 The TensorFlow Datasets Authors and the HuggingFace Datasets 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. | |
# Lint as: python3 | |
"""SUPERB: Speech processing Universal PERformance Benchmark.""" | |
import csv | |
import glob | |
import os | |
import textwrap | |
from dataclasses import dataclass | |
import datasets | |
from datasets.tasks import AutomaticSpeechRecognition | |
_CITATION = """\ | |
@article{DBLP:journals/corr/abs-2105-01051, | |
author = {Shu{-}Wen Yang and | |
Po{-}Han Chi and | |
Yung{-}Sung Chuang and | |
Cheng{-}I Jeff Lai and | |
Kushal Lakhotia and | |
Yist Y. Lin and | |
Andy T. Liu and | |
Jiatong Shi and | |
Xuankai Chang and | |
Guan{-}Ting Lin and | |
Tzu{-}Hsien Huang and | |
Wei{-}Cheng Tseng and | |
Ko{-}tik Lee and | |
Da{-}Rong Liu and | |
Zili Huang and | |
Shuyan Dong and | |
Shang{-}Wen Li and | |
Shinji Watanabe and | |
Abdelrahman Mohamed and | |
Hung{-}yi Lee}, | |
title = {{SUPERB:} Speech processing Universal PERformance Benchmark}, | |
journal = {CoRR}, | |
volume = {abs/2105.01051}, | |
year = {2021}, | |
url = {https://arxiv.org/abs/2105.01051}, | |
archivePrefix = {arXiv}, | |
eprint = {2105.01051}, | |
timestamp = {Thu, 01 Jul 2021 13:30:22 +0200}, | |
biburl = {https://dblp.org/rec/journals/corr/abs-2105-01051.bib}, | |
bibsource = {dblp computer science bibliography, https://dblp.org} | |
} | |
""" | |
_DESCRIPTION = """\ | |
Self-supervised learning (SSL) has proven vital for advancing research in | |
natural language processing (NLP) and computer vision (CV). The paradigm | |
pretrains a shared model on large volumes of unlabeled data and achieves | |
state-of-the-art (SOTA) for various tasks with minimal adaptation. However, the | |
speech processing community lacks a similar setup to systematically explore the | |
paradigm. To bridge this gap, we introduce Speech processing Universal | |
PERformance Benchmark (SUPERB). SUPERB is a leaderboard to benchmark the | |
performance of a shared model across a wide range of speech processing tasks | |
with minimal architecture changes and labeled data. Among multiple usages of the | |
shared model, we especially focus on extracting the representation learned from | |
SSL due to its preferable re-usability. We present a simple framework to solve | |
SUPERB tasks by learning task-specialized lightweight prediction heads on top of | |
the frozen shared model. Our results demonstrate that the framework is promising | |
as SSL representations show competitive generalizability and accessibility | |
across SUPERB tasks. We release SUPERB as a challenge with a leaderboard and a | |
benchmark toolkit to fuel the research in representation learning and general | |
speech processing. | |
Note that in order to limit the required storage for preparing this dataset, the | |
audio is stored in the .wav format and is not converted to a float32 array. To | |
convert the audio file to a float32 array, please make use of the `.map()` | |
function as follows: | |
```python | |
import soundfile as sf | |
def map_to_array(batch): | |
speech_array, _ = sf.read(batch["file"]) | |
batch["speech"] = speech_array | |
return batch | |
dataset = dataset.map(map_to_array, remove_columns=["file"]) | |
``` | |
""" | |
class SuperbConfig(datasets.BuilderConfig): | |
"""BuilderConfig for Superb.""" | |
def __init__( | |
self, | |
features, | |
url, | |
data_url=None, | |
supervised_keys=None, | |
task_templates=None, | |
**kwargs, | |
): | |
super().__init__(version=datasets.Version("1.9.0", ""), **kwargs) | |
self.features = features | |
self.data_url = data_url | |
self.url = url | |
self.supervised_keys = supervised_keys | |
self.task_templates = task_templates | |
class Superb(datasets.GeneratorBasedBuilder): | |
"""Superb dataset.""" | |
BUILDER_CONFIGS = [ | |
SuperbConfig( | |
name="asr", | |
description=textwrap.dedent( | |
"""\ | |
ASR transcribes utterances into words. While PR analyzes the | |
improvement in modeling phonetics, ASR reflects the significance of | |
the improvement in a real-world scenario. LibriSpeech | |
train-clean-100/dev-clean/test-clean subsets are used for | |
training/validation/testing. The evaluation metric is word error | |
rate (WER).""" | |
), | |
features=datasets.Features( | |
{ | |
"file": datasets.Value("string"), | |
"audio": datasets.Audio(sampling_rate=16_000), | |
"text": datasets.Value("string"), | |
"speaker_id": datasets.Value("int64"), | |
"chapter_id": datasets.Value("int64"), | |
"id": datasets.Value("string"), | |
} | |
), | |
supervised_keys=("file", "text"), | |
url="http://www.openslr.org/12", | |
data_url="http://www.openslr.org/resources/12/", | |
task_templates=[AutomaticSpeechRecognition(audio_column="audio", transcription_column="text")], | |
), | |
SuperbConfig( | |
name="ks", | |
description=textwrap.dedent( | |
"""\ | |
Keyword Spotting (KS) detects preregistered keywords by classifying utterances into a predefined set of | |
words. The task is usually performed on-device for the fast response time. Thus, accuracy, model size, and | |
inference time are all crucial. SUPERB uses the widely used Speech Commands dataset v1.0 for the task. | |
The dataset consists of ten classes of keywords, a class for silence, and an unknown class to include the | |
false positive. The evaluation metric is accuracy (ACC)""" | |
), | |
features=datasets.Features( | |
{ | |
"file": datasets.Value("string"), | |
"audio": datasets.Audio(sampling_rate=16_000), | |
"label": datasets.ClassLabel( | |
names=[ | |
"yes", | |
"no", | |
"up", | |
"down", | |
"left", | |
"right", | |
"on", | |
"off", | |
"stop", | |
"go", | |
"_silence_", | |
"_unknown_", | |
] | |
), | |
} | |
), | |
supervised_keys=("file", "label"), | |
url="https://www.tensorflow.org/datasets/catalog/speech_commands", | |
data_url="http://download.tensorflow.org/data/{filename}", | |
), | |
SuperbConfig( | |
name="ic", | |
description=textwrap.dedent( | |
"""\ | |
Intent Classification (IC) classifies utterances into predefined classes to determine the intent of | |
speakers. SUPERB uses the Fluent Speech Commands dataset, where each utterance is tagged with three intent | |
labels: action, object, and location. The evaluation metric is accuracy (ACC).""" | |
), | |
features=datasets.Features( | |
{ | |
"file": datasets.Value("string"), | |
"audio": datasets.Audio(sampling_rate=16_000), | |
"speaker_id": datasets.Value("string"), | |
"text": datasets.Value("string"), | |
"action": datasets.ClassLabel( | |
names=["activate", "bring", "change language", "deactivate", "decrease", "increase"] | |
), | |
"object": datasets.ClassLabel( | |
names=[ | |
"Chinese", | |
"English", | |
"German", | |
"Korean", | |
"heat", | |
"juice", | |
"lamp", | |
"lights", | |
"music", | |
"newspaper", | |
"none", | |
"shoes", | |
"socks", | |
"volume", | |
] | |
), | |
"location": datasets.ClassLabel(names=["bedroom", "kitchen", "none", "washroom"]), | |
} | |
), | |
supervised_keys=None, | |
url="https://fluent.ai/fluent-speech-commands-a-dataset-for-spoken-language-understanding-research/", | |
data_url="http://fluent.ai:2052/jf8398hf30f0381738rucj3828chfdnchs.tar.gz", | |
), | |
SuperbConfig( | |
name="si", | |
description=textwrap.dedent( | |
"""\ | |
Speaker Identification (SI) classifies each utterance for its speaker identity as a multi-class | |
classification, where speakers are in the same predefined set for both training and testing. The widely | |
used VoxCeleb1 dataset is adopted, and the evaluation metric is accuracy (ACC).""" | |
), | |
features=datasets.Features( | |
{ | |
"file": datasets.Value("string"), | |
"audio": datasets.Audio(sampling_rate=16_000), | |
# VoxCeleb1 contains 1251 speaker IDs in range ["id10001",..."id11251"] | |
"label": datasets.ClassLabel(names=[f"id{i + 10001}" for i in range(1251)]), | |
} | |
), | |
supervised_keys=("file", "label"), | |
url="https://www.robots.ox.ac.uk/~vgg/data/voxceleb/vox1.html", | |
), | |
SuperbConfig( | |
name="sd", | |
description=textwrap.dedent( | |
"""\ | |
Speaker Diarization (SD) predicts `who is speaking when` for each timestamp, and multiple speakers can | |
speak simultaneously. The model has to encode rich speaker characteristics for each frame and should be | |
able to represent mixtures of signals. [LibriMix] is adopted where LibriSpeech | |
train-clean-100/dev-clean/test-clean are used to generate mixtures for training/validation/testing. | |
We focus on the two-speaker scenario as the first step. The time-coded speaker labels were generated using | |
alignments from Kaldi LibriSpeech ASR model. The evaluation metric is diarization error rate (DER).""" | |
), | |
features=datasets.Features( | |
{ | |
"record_id": datasets.Value("string"), | |
"file": datasets.Value("string"), | |
"audio": datasets.Audio(sampling_rate=16_000), | |
"start": datasets.Value("int64"), | |
"end": datasets.Value("int64"), | |
"speakers": [ | |
{ | |
"speaker_id": datasets.Value("string"), | |
"start": datasets.Value("int64"), | |
"end": datasets.Value("int64"), | |
} | |
], | |
} | |
), # TODO | |
supervised_keys=None, # TODO | |
url="https://github.com/ftshijt/LibriMix", | |
data_url="https://huggingface.co/datasets/superb/superb-data/resolve/main/sd/{split}/{filename}", | |
), | |
SuperbConfig( | |
name="er", | |
description=textwrap.dedent( | |
"""\ | |
Emotion Recognition (ER) predicts an emotion class for each utterance. The most widely used ER dataset | |
IEMOCAP is adopted, and we follow the conventional evaluation protocol: we drop the unbalanced emotion | |
classes to leave the final four classes with a similar amount of data points and cross-validate on five | |
folds of the standard splits. The evaluation metric is accuracy (ACC).""" | |
), | |
features=datasets.Features( | |
{ | |
"file": datasets.Value("string"), | |
"audio": datasets.Audio(sampling_rate=16_000), | |
"label": datasets.ClassLabel(names=["neu", "hap", "ang", "sad"]), | |
} | |
), | |
supervised_keys=("file", "label"), | |
url="https://sail.usc.edu/iemocap/", | |
), | |
] | |
def manual_download_instructions(self): | |
if self.config.name == "si": | |
return textwrap.dedent( | |
""" | |
Please download the VoxCeleb dataset using the following script, | |
which should create `VoxCeleb1/wav/id*` directories for both train and test speakers`: | |
``` | |
mkdir VoxCeleb1 | |
cd VoxCeleb1 | |
wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partaa | |
wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partab | |
wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partac | |
wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_dev_wav_partad | |
cat vox1_dev* > vox1_dev_wav.zip | |
unzip vox1_dev_wav.zip | |
wget https://thor.robots.ox.ac.uk/~vgg/data/voxceleb/vox1a/vox1_test_wav.zip | |
unzip vox1_test_wav.zip | |
# download the official SUPERB train-dev-test split | |
wget https://raw.githubusercontent.com/s3prl/s3prl/master/s3prl/downstream/voxceleb1/veri_test_class.txt | |
```""" | |
) | |
elif self.config.name == "er": | |
return textwrap.dedent( | |
""" | |
Please download the IEMOCAP dataset after submitting the request form here: | |
https://sail.usc.edu/iemocap/iemocap_release.htm | |
Having downloaded the dataset you can extract it with `tar -xvzf IEMOCAP_full_release.tar.gz` | |
which should create a folder called `IEMOCAP_full_release` | |
""" | |
) | |
return None | |
def _info(self): | |
return datasets.DatasetInfo( | |
description=_DESCRIPTION, | |
features=self.config.features, | |
supervised_keys=self.config.supervised_keys, | |
homepage=self.config.url, | |
citation=_CITATION, | |
task_templates=self.config.task_templates, | |
) | |
def _split_generators(self, dl_manager): | |
if self.config.name == "asr": | |
_DL_URLS = { | |
"dev": self.config.data_url + "dev-clean.tar.gz", | |
"test": self.config.data_url + "test-clean.tar.gz", | |
"train": self.config.data_url + "train-clean-100.tar.gz", | |
} | |
archive_path = dl_manager.download_and_extract(_DL_URLS) | |
return [ | |
datasets.SplitGenerator(name=datasets.Split.TRAIN, gen_kwargs={"archive_path": archive_path["train"]}), | |
datasets.SplitGenerator( | |
name=datasets.Split.VALIDATION, gen_kwargs={"archive_path": archive_path["dev"]} | |
), | |
datasets.SplitGenerator(name=datasets.Split.TEST, gen_kwargs={"archive_path": archive_path["test"]}), | |
] | |
elif self.config.name == "ks": | |
_DL_URLS = { | |
"train_val_test": self.config.data_url.format(filename="speech_commands_v0.01.tar.gz"), | |
"test": self.config.data_url.format(filename="speech_commands_test_set_v0.01.tar.gz"), | |
} | |
archive_path = dl_manager.download_and_extract(_DL_URLS) | |
return [ | |
datasets.SplitGenerator( | |
name=datasets.Split.TRAIN, | |
gen_kwargs={"archive_path": archive_path["train_val_test"], "split": "train"}, | |
), | |
datasets.SplitGenerator( | |
name=datasets.Split.VALIDATION, | |
gen_kwargs={"archive_path": archive_path["train_val_test"], "split": "val"}, | |
), | |
datasets.SplitGenerator( | |
name=datasets.Split.TEST, gen_kwargs={"archive_path": archive_path["test"], "split": "test"} | |
), | |
] | |
elif self.config.name == "ic": | |
archive_path = dl_manager.download_and_extract(self.config.data_url) | |
return [ | |
datasets.SplitGenerator( | |
name=datasets.Split.TRAIN, | |
gen_kwargs={"archive_path": archive_path, "split": "train"}, | |
), | |
datasets.SplitGenerator( | |
name=datasets.Split.VALIDATION, | |
gen_kwargs={"archive_path": archive_path, "split": "valid"}, | |
), | |
datasets.SplitGenerator( | |
name=datasets.Split.TEST, gen_kwargs={"archive_path": archive_path, "split": "test"} | |
), | |
] | |
elif self.config.name == "si": | |
manual_dir = os.path.abspath(os.path.expanduser(dl_manager.manual_dir)) | |
return [ | |
datasets.SplitGenerator( | |
name=datasets.Split.TRAIN, | |
gen_kwargs={"archive_path": manual_dir, "split": 1}, | |
), | |
datasets.SplitGenerator( | |
name=datasets.Split.VALIDATION, | |
gen_kwargs={"archive_path": manual_dir, "split": 2}, | |
), | |
datasets.SplitGenerator(name=datasets.Split.TEST, gen_kwargs={"archive_path": manual_dir, "split": 3}), | |
] | |
elif self.config.name == "sd": | |
splits = ["train", "dev", "test"] | |
_DL_URLS = { | |
split: { | |
filename: self.config.data_url.format(split=split, filename=filename) | |
for filename in ["reco2dur", "segments", "utt2spk", "wav.zip"] | |
} | |
for split in splits | |
} | |
archive_path = dl_manager.download_and_extract(_DL_URLS) | |
return [ | |
datasets.SplitGenerator( | |
name=datasets.NamedSplit(split), gen_kwargs={"archive_path": archive_path[split], "split": split} | |
) | |
for split in splits | |
] | |
elif self.config.name == "er": | |
manual_dir = os.path.abspath(os.path.expanduser(dl_manager.manual_dir)) | |
return [ | |
datasets.SplitGenerator( | |
name=f"session{i}", | |
gen_kwargs={"archive_path": manual_dir, "split": i}, | |
) | |
for i in range(1, 6) | |
] | |
def _generate_examples(self, archive_path, split=None): | |
"""Generate examples.""" | |
if self.config.name == "asr": | |
transcripts_glob = os.path.join(archive_path, "LibriSpeech", "*", "*", "*", "*.txt") | |
key = 0 | |
for transcript_path in sorted(glob.glob(transcripts_glob)): | |
transcript_dir_path = os.path.dirname(transcript_path) | |
with open(transcript_path, "r", encoding="utf-8") as f: | |
for line in f: | |
line = line.strip() | |
id_, transcript = line.split(" ", 1) | |
audio_file = f"{id_}.flac" | |
speaker_id, chapter_id = [int(el) for el in id_.split("-")[:2]] | |
audio_path = os.path.join(transcript_dir_path, audio_file) | |
yield key, { | |
"id": id_, | |
"speaker_id": speaker_id, | |
"chapter_id": chapter_id, | |
"file": audio_path, | |
"audio": audio_path, | |
"text": transcript, | |
} | |
key += 1 | |
elif self.config.name == "ks": | |
words = ["yes", "no", "up", "down", "left", "right", "on", "off", "stop", "go"] | |
splits = _split_ks_files(archive_path, split) | |
for key, audio_file in enumerate(sorted(splits[split])): | |
base_dir, file_name = os.path.split(audio_file) | |
_, word = os.path.split(base_dir) | |
if word in words: | |
label = word | |
elif word == "_silence_" or word == "_background_noise_": | |
label = "_silence_" | |
else: | |
label = "_unknown_" | |
yield key, {"file": audio_file, "audio": audio_file, "label": label} | |
elif self.config.name == "ic": | |
root_path = os.path.join(archive_path, "fluent_speech_commands_dataset") | |
csv_path = os.path.join(root_path, "data", f"{split}_data.csv") | |
with open(csv_path, encoding="utf-8") as csv_file: | |
csv_reader = csv.reader(csv_file, delimiter=",", skipinitialspace=True) | |
next(csv_reader) | |
for row in csv_reader: | |
key, file_path, speaker_id, text, action, object_, location = row | |
audio_path = os.path.join(root_path, file_path) | |
yield key, { | |
"file": audio_path, | |
"audio": audio_path, | |
"speaker_id": speaker_id, | |
"text": text, | |
"action": action, | |
"object": object_, | |
"location": location, | |
} | |
elif self.config.name == "si": | |
wav_path = os.path.join(archive_path, "wav") | |
splits_path = os.path.join(archive_path, "veri_test_class.txt") | |
with open(splits_path, "r", encoding="utf-8") as f: | |
for key, line in enumerate(f): | |
split_id, file_path = line.strip().split(" ") | |
if int(split_id) != split: | |
continue | |
speaker_id = file_path.split("/")[0] | |
audio_path = os.path.join(wav_path, file_path) | |
yield key, { | |
"file": audio_path, | |
"audio": audio_path, | |
"label": speaker_id, | |
} | |
elif self.config.name == "sd": | |
data = SdData(archive_path) | |
args = SdArgs() | |
chunk_indices = _generate_chunk_indices(data, args, split=split) | |
if split != "test": | |
for key, (rec, st, ed) in enumerate(chunk_indices): | |
speakers = _get_speakers(rec, data, args) | |
yield key, { | |
"record_id": rec, | |
"file": data.wavs[rec], | |
"audio": data.wavs[rec], | |
"start": st, | |
"end": ed, | |
"speakers": speakers, | |
} | |
else: | |
key = 0 | |
for rec in chunk_indices: | |
for rec, st, ed in chunk_indices[rec]: | |
speakers = _get_speakers(rec, data, args) | |
yield key, { | |
"record_id": rec, | |
"file": data.wavs[rec], | |
"audio": data.wavs[rec], | |
"start": st, | |
"end": ed, | |
"speakers": speakers, | |
} | |
key += 1 | |
elif self.config.name == "er": | |
root_path = os.path.join(archive_path, f"Session{split}") | |
wav_path = os.path.join(root_path, "sentences", "wav") | |
labels_path = os.path.join(root_path, "dialog", "EmoEvaluation", "*.txt") | |
emotions = ["neu", "hap", "ang", "sad", "exc"] | |
key = 0 | |
for labels_file in sorted(glob.glob(labels_path)): | |
with open(labels_file, "r", encoding="utf-8") as f: | |
for line in f: | |
if line[0] != "[": | |
continue | |
_, filename, emo, _ = line.split("\t") | |
if emo not in emotions: | |
continue | |
wav_subdir = filename.rsplit("_", 1)[0] | |
filename = f"{filename}.wav" | |
audio_path = os.path.join(wav_path, wav_subdir, filename) | |
yield key, { | |
"file": audio_path, | |
"audio": audio_path, | |
"label": emo.replace("exc", "hap"), | |
} | |
key += 1 | |
class SdData: | |
def __init__(self, data_dir): | |
"""Load sd data.""" | |
self.segments = self._load_segments_rechash(data_dir["segments"]) | |
self.utt2spk = self._load_utt2spk(data_dir["utt2spk"]) | |
self.wavs = self._load_wav_zip(data_dir["wav.zip"]) | |
self.reco2dur = self._load_reco2dur(data_dir["reco2dur"]) | |
def _load_segments_rechash(self, segments_file): | |
"""Load segments file as dict with recid index.""" | |
ret = {} | |
if not os.path.exists(segments_file): | |
return None | |
with open(segments_file, encoding="utf-8") as f: | |
for line in f: | |
utt, rec, st, et = line.strip().split() | |
if rec not in ret: | |
ret[rec] = [] | |
ret[rec].append({"utt": utt, "st": float(st), "et": float(et)}) | |
return ret | |
def _load_wav_zip(self, wav_zip): | |
"""Return dictionary { rec: wav_rxfilename }.""" | |
wav_dir = os.path.join(wav_zip, "wav") | |
return { | |
os.path.splitext(filename)[0]: os.path.join(wav_dir, filename) for filename in sorted(os.listdir(wav_dir)) | |
} | |
def _load_utt2spk(self, utt2spk_file): | |
"""Returns dictionary { uttid: spkid }.""" | |
with open(utt2spk_file, encoding="utf-8") as f: | |
lines = [line.strip().split(None, 1) for line in f] | |
return {x[0]: x[1] for x in lines} | |
def _load_reco2dur(self, reco2dur_file): | |
"""Returns dictionary { recid: duration }.""" | |
if not os.path.exists(reco2dur_file): | |
return None | |
with open(reco2dur_file, encoding="utf-8") as f: | |
lines = [line.strip().split(None, 1) for line in f] | |
return {x[0]: float(x[1]) for x in lines} | |
class SdArgs: | |
chunk_size: int = 2000 | |
frame_shift: int = 160 | |
subsampling: int = 1 | |
label_delay: int = 0 | |
num_speakers: int = 2 | |
rate: int = 16000 | |
use_last_samples: bool = True | |
def _generate_chunk_indices(data, args, split=None): | |
chunk_indices = [] if split != "test" else {} | |
# make chunk indices: filepath, start_frame, end_frame | |
for rec in data.wavs: | |
data_len = int(data.reco2dur[rec] * args.rate / args.frame_shift) | |
data_len = int(data_len / args.subsampling) | |
if split == "test": | |
chunk_indices[rec] = [] | |
if split != "test": | |
for st, ed in _gen_frame_indices( | |
data_len, | |
args.chunk_size, | |
args.chunk_size, | |
args.use_last_samples, | |
label_delay=args.label_delay, | |
subsampling=args.subsampling, | |
): | |
chunk_indices.append((rec, st * args.subsampling, ed * args.subsampling)) | |
else: | |
for st, ed in _gen_chunk_indices(data_len, args.chunk_size): | |
chunk_indices[rec].append((rec, st * args.subsampling, ed * args.subsampling)) | |
return chunk_indices | |
def _count_frames(data_len, size, step): | |
# no padding at edges, last remaining samples are ignored | |
return int((data_len - size + step) / step) | |
def _gen_frame_indices(data_length, size=2000, step=2000, use_last_samples=False, label_delay=0, subsampling=1): | |
i = -1 | |
for i in range(_count_frames(data_length, size, step)): | |
yield i * step, i * step + size | |
if use_last_samples and i * step + size < data_length: | |
if data_length - (i + 1) * step - subsampling * label_delay > 0: | |
yield (i + 1) * step, data_length | |
def _gen_chunk_indices(data_len, chunk_size): | |
step = chunk_size | |
start = 0 | |
while start < data_len: | |
end = min(data_len, start + chunk_size) | |
yield start, end | |
start += step | |
def _get_speakers(rec, data, args): | |
return [ | |
{ | |
"speaker_id": data.utt2spk[segment["utt"]], | |
"start": round(segment["st"] * args.rate / args.frame_shift), | |
"end": round(segment["et"] * args.rate / args.frame_shift), | |
} | |
for segment in data.segments[rec] | |
] | |
def _split_ks_files(archive_path, split): | |
audio_path = os.path.join(archive_path, "**", "*.wav") | |
audio_paths = glob.glob(audio_path) | |
if split == "test": | |
# use all available files for the test archive | |
return {"test": audio_paths} | |
val_list_file = os.path.join(archive_path, "validation_list.txt") | |
test_list_file = os.path.join(archive_path, "testing_list.txt") | |
with open(val_list_file, encoding="utf-8") as f: | |
val_paths = f.read().strip().splitlines() | |
val_paths = [os.path.join(archive_path, p) for p in val_paths] | |
with open(test_list_file, encoding="utf-8") as f: | |
test_paths = f.read().strip().splitlines() | |
test_paths = [os.path.join(archive_path, p) for p in test_paths] | |
# the paths for the train set is just whichever paths that do not exist in | |
# either the test or validation splits | |
train_paths = list(set(audio_paths) - set(val_paths) - set(test_paths)) | |
return {"train": train_paths, "val": val_paths} | |