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# coding=utf-8
# Copyright 2023 HuggingFace Inc.
#
# 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.
""" Testing suite for the TVLT feature extraction. """
import itertools
import os
import random
import tempfile
import unittest
import numpy as np
from transformers import is_datasets_available, is_speech_available
from transformers.testing_utils import check_json_file_has_correct_format, require_torch, require_torchaudio
from transformers.utils.import_utils import is_torch_available
from ...test_sequence_feature_extraction_common import SequenceFeatureExtractionTestMixin
if is_torch_available():
import torch
if is_datasets_available():
from datasets import load_dataset
if is_speech_available():
from transformers import TvltFeatureExtractor
global_rng = random.Random()
def floats_list(shape, scale=1.0, rng=None, name=None):
"""Creates a random float32 tensor"""
if rng is None:
rng = global_rng
values = []
for batch_idx in range(shape[0]):
values.append([])
for _ in range(shape[1]):
values[-1].append(rng.random() * scale)
return values
class TvltFeatureExtractionTester(unittest.TestCase):
def __init__(
self,
parent,
batch_size=7,
min_seq_length=400,
max_seq_length=2000,
spectrogram_length=2048,
feature_size=128,
num_audio_channels=1,
hop_length=512,
chunk_length=30,
sampling_rate=44100,
):
self.parent = parent
self.batch_size = batch_size
self.min_seq_length = min_seq_length
self.max_seq_length = max_seq_length
self.seq_length_diff = (self.max_seq_length - self.min_seq_length) // (self.batch_size - 1)
self.spectrogram_length = spectrogram_length
self.feature_size = feature_size
self.num_audio_channels = num_audio_channels
self.hop_length = hop_length
self.chunk_length = chunk_length
self.sampling_rate = sampling_rate
def prepare_feat_extract_dict(self):
return {
"spectrogram_length": self.spectrogram_length,
"feature_size": self.feature_size,
"num_audio_channels": self.num_audio_channels,
"hop_length": self.hop_length,
"chunk_length": self.chunk_length,
"sampling_rate": self.sampling_rate,
}
def prepare_inputs_for_common(self, equal_length=False, numpify=False):
def _flatten(list_of_lists):
return list(itertools.chain(*list_of_lists))
if equal_length:
speech_inputs = [floats_list((self.max_seq_length, self.feature_size)) for _ in range(self.batch_size)]
else:
# make sure that inputs increase in size
speech_inputs = [
floats_list((x, self.feature_size))
for x in range(self.min_seq_length, self.max_seq_length, self.seq_length_diff)
]
if numpify:
speech_inputs = [np.asarray(x) for x in speech_inputs]
return speech_inputs
@require_torch
@require_torchaudio
class TvltFeatureExtractionTest(SequenceFeatureExtractionTestMixin, unittest.TestCase):
feature_extraction_class = TvltFeatureExtractor if is_speech_available() else None
def setUp(self):
self.feat_extract_tester = TvltFeatureExtractionTester(self)
def test_feat_extract_properties(self):
feature_extractor = self.feature_extraction_class(**self.feat_extract_dict)
self.assertTrue(hasattr(feature_extractor, "spectrogram_length"))
self.assertTrue(hasattr(feature_extractor, "feature_size"))
self.assertTrue(hasattr(feature_extractor, "num_audio_channels"))
self.assertTrue(hasattr(feature_extractor, "hop_length"))
self.assertTrue(hasattr(feature_extractor, "chunk_length"))
self.assertTrue(hasattr(feature_extractor, "sampling_rate"))
def test_feat_extract_from_and_save_pretrained(self):
feat_extract_first = self.feature_extraction_class(**self.feat_extract_dict)
with tempfile.TemporaryDirectory() as tmpdirname:
saved_file = feat_extract_first.save_pretrained(tmpdirname)[0]
check_json_file_has_correct_format(saved_file)
feat_extract_second = self.feature_extraction_class.from_pretrained(tmpdirname)
dict_first = feat_extract_first.to_dict()
dict_second = feat_extract_second.to_dict()
mel_1 = dict_first.pop("mel_filters")
mel_2 = dict_second.pop("mel_filters")
self.assertTrue(np.allclose(mel_1, mel_2))
self.assertEqual(dict_first, dict_second)
def test_feat_extract_to_json_file(self):
feat_extract_first = self.feature_extraction_class(**self.feat_extract_dict)
with tempfile.TemporaryDirectory() as tmpdirname:
json_file_path = os.path.join(tmpdirname, "feat_extract.json")
feat_extract_first.to_json_file(json_file_path)
feat_extract_second = self.feature_extraction_class.from_json_file(json_file_path)
dict_first = feat_extract_first.to_dict()
dict_second = feat_extract_second.to_dict()
mel_1 = dict_first.pop("mel_filters")
mel_2 = dict_second.pop("mel_filters")
self.assertTrue(np.allclose(mel_1, mel_2))
self.assertEqual(dict_first, dict_second)
def test_call(self):
# Initialize feature_extractor
feature_extractor = self.feature_extraction_class(**self.feat_extract_dict)
# create three inputs of length 800, 1000, and 1200
speech_inputs = [floats_list((1, x))[0] for x in range(8000, 14000, 20000)]
np_speech_inputs = [np.asarray(speech_input) for speech_input in speech_inputs]
# Test not batched input
encoded_audios = feature_extractor(np_speech_inputs[0], return_tensors="np", sampling_rate=44100).audio_values
self.assertTrue(encoded_audios.ndim == 4)
self.assertTrue(encoded_audios.shape[-1] == feature_extractor.feature_size)
self.assertTrue(encoded_audios.shape[-2] <= feature_extractor.spectrogram_length)
self.assertTrue(encoded_audios.shape[-3] == feature_extractor.num_channels)
# Test batched
encoded_audios = feature_extractor(np_speech_inputs, return_tensors="np", sampling_rate=44100).audio_values
self.assertTrue(encoded_audios.ndim == 4)
self.assertTrue(encoded_audios.shape[-1] == feature_extractor.feature_size)
self.assertTrue(encoded_audios.shape[-2] <= feature_extractor.spectrogram_length)
self.assertTrue(encoded_audios.shape[-3] == feature_extractor.num_channels)
# Test audio masking
encoded_audios = feature_extractor(
np_speech_inputs, return_tensors="np", sampling_rate=44100, mask_audio=True
).audio_values
self.assertTrue(encoded_audios.ndim == 4)
self.assertTrue(encoded_audios.shape[-1] == feature_extractor.feature_size)
self.assertTrue(encoded_audios.shape[-2] <= feature_extractor.spectrogram_length)
self.assertTrue(encoded_audios.shape[-3] == feature_extractor.num_channels)
def _load_datasamples(self, num_samples):
ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
# automatic decoding with librispeech
speech_samples = ds.sort("id").select(range(num_samples))[:num_samples]["audio"]
return [x["array"] for x in speech_samples]
def test_integration(self):
input_speech = self._load_datasamples(1)
feaure_extractor = TvltFeatureExtractor()
audio_values = feaure_extractor(input_speech, return_tensors="pt").audio_values
self.assertTrue(audio_values.shape, [1, 1, 192, 128])
expected_slice = torch.tensor([[-0.3032, -0.2708], [-0.4434, -0.4007]])
self.assertTrue(torch.allclose(audio_values[0, 0, :2, :2], expected_slice, atol=1e-4))
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