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			| ee6e328 | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 | # Copyright 2021 The HuggingFace Team. All rights reserved.
#
# 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 unittest
import numpy as np
from transformers import MODEL_FOR_AUDIO_CLASSIFICATION_MAPPING, TF_MODEL_FOR_AUDIO_CLASSIFICATION_MAPPING
from transformers.pipelines import AudioClassificationPipeline, pipeline
from transformers.testing_utils import (
    is_pipeline_test,
    nested_simplify,
    require_tf,
    require_torch,
    require_torchaudio,
    slow,
)
from .test_pipelines_common import ANY
@is_pipeline_test
class AudioClassificationPipelineTests(unittest.TestCase):
    model_mapping = MODEL_FOR_AUDIO_CLASSIFICATION_MAPPING
    tf_model_mapping = TF_MODEL_FOR_AUDIO_CLASSIFICATION_MAPPING
    def get_test_pipeline(self, model, tokenizer, processor):
        audio_classifier = AudioClassificationPipeline(model=model, feature_extractor=processor)
        # test with a raw waveform
        audio = np.zeros((34000,))
        audio2 = np.zeros((14000,))
        return audio_classifier, [audio2, audio]
    def run_pipeline_test(self, audio_classifier, examples):
        audio2, audio = examples
        output = audio_classifier(audio)
        # by default a model is initialized with num_labels=2
        self.assertEqual(
            output,
            [
                {"score": ANY(float), "label": ANY(str)},
                {"score": ANY(float), "label": ANY(str)},
            ],
        )
        output = audio_classifier(audio, top_k=1)
        self.assertEqual(
            output,
            [
                {"score": ANY(float), "label": ANY(str)},
            ],
        )
        self.run_torchaudio(audio_classifier)
    @require_torchaudio
    def run_torchaudio(self, audio_classifier):
        import datasets
        # test with a local file
        dataset = datasets.load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
        audio = dataset[0]["audio"]["array"]
        output = audio_classifier(audio)
        self.assertEqual(
            output,
            [
                {"score": ANY(float), "label": ANY(str)},
                {"score": ANY(float), "label": ANY(str)},
            ],
        )
    @require_torch
    def test_small_model_pt(self):
        model = "anton-l/wav2vec2-random-tiny-classifier"
        audio_classifier = pipeline("audio-classification", model=model)
        audio = np.ones((8000,))
        output = audio_classifier(audio, top_k=4)
        EXPECTED_OUTPUT = [
            {"score": 0.0842, "label": "no"},
            {"score": 0.0838, "label": "up"},
            {"score": 0.0837, "label": "go"},
            {"score": 0.0834, "label": "right"},
        ]
        EXPECTED_OUTPUT_PT_2 = [
            {"score": 0.0845, "label": "stop"},
            {"score": 0.0844, "label": "on"},
            {"score": 0.0841, "label": "right"},
            {"score": 0.0834, "label": "left"},
        ]
        self.assertIn(nested_simplify(output, decimals=4), [EXPECTED_OUTPUT, EXPECTED_OUTPUT_PT_2])
        audio_dict = {"array": np.ones((8000,)), "sampling_rate": audio_classifier.feature_extractor.sampling_rate}
        output = audio_classifier(audio_dict, top_k=4)
        self.assertIn(nested_simplify(output, decimals=4), [EXPECTED_OUTPUT, EXPECTED_OUTPUT_PT_2])
    @require_torch
    @slow
    def test_large_model_pt(self):
        import datasets
        model = "superb/wav2vec2-base-superb-ks"
        audio_classifier = pipeline("audio-classification", model=model)
        dataset = datasets.load_dataset("anton-l/superb_dummy", "ks", split="test")
        audio = np.array(dataset[3]["speech"], dtype=np.float32)
        output = audio_classifier(audio, top_k=4)
        self.assertEqual(
            nested_simplify(output, decimals=3),
            [
                {"score": 0.981, "label": "go"},
                {"score": 0.007, "label": "up"},
                {"score": 0.006, "label": "_unknown_"},
                {"score": 0.001, "label": "down"},
            ],
        )
    @require_tf
    @unittest.skip("Audio classification is not implemented for TF")
    def test_small_model_tf(self):
        pass
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