# 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 random
import unittest

import numpy as np

from transformers import TvltFeatureExtractor, is_datasets_available
from transformers.testing_utils import 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

global_rng = random.Random()


# Copied from tests.models.whisper.test_feature_extraction_whisper.floats_list
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

    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_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(800, 1400, 200)]
        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)

        # Test 2-D numpy arrays are batched.
        speech_inputs = [floats_list((1, x))[0] for x in (800, 800, 800)]
        np_speech_inputs = np.asarray(speech_inputs)
        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)

    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)
        feature_extractor = TvltFeatureExtractor()
        audio_values = feature_extractor(input_speech, return_tensors="pt").audio_values

        self.assertEquals(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))
