mirror of
https://github.com/huggingface/transformers.git
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749 lines
30 KiB
Python
749 lines
30 KiB
Python
# coding=utf-8
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# Copyright 2024 The HuggingFace Inc. team. All rights reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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"""Testing suite for the PyTorch Dac model."""
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import inspect
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import os
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import tempfile
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import unittest
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from typing import Dict, List, Tuple
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import numpy as np
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from datasets import Audio, load_dataset
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from transformers import AutoProcessor, DacConfig, DacModel
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from transformers.testing_utils import is_torch_available, require_torch, slow, torch_device
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from ...test_configuration_common import ConfigTester
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from ...test_modeling_common import ModelTesterMixin, _config_zero_init, floats_tensor
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from ...test_pipeline_mixin import PipelineTesterMixin
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if is_torch_available():
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import torch
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@require_torch
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# Copied from transformers.tests.encodec.test_modeling_encodec.EncodecModelTester with Encodec->Dac
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class DacModelTester:
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# Ignore copy
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def __init__(
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self,
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parent,
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batch_size=3,
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num_channels=1,
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is_training=False,
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intermediate_size=1024,
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encoder_hidden_size=16,
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downsampling_ratios=[2, 4, 4],
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decoder_hidden_size=16,
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n_codebooks=6,
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codebook_size=512,
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codebook_dim=4,
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quantizer_dropout=0.0,
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commitment_loss_weight=0.25,
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codebook_loss_weight=1.0,
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sample_rate=16000,
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):
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self.parent = parent
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self.batch_size = batch_size
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self.num_channels = num_channels
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self.is_training = is_training
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self.intermediate_size = intermediate_size
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self.sample_rate = sample_rate
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self.encoder_hidden_size = encoder_hidden_size
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self.downsampling_ratios = downsampling_ratios
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self.decoder_hidden_size = decoder_hidden_size
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self.n_codebooks = n_codebooks
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self.codebook_size = codebook_size
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self.codebook_dim = codebook_dim
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self.quantizer_dropout = quantizer_dropout
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self.commitment_loss_weight = commitment_loss_weight
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self.codebook_loss_weight = codebook_loss_weight
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def prepare_config_and_inputs(self):
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input_values = floats_tensor([self.batch_size, self.num_channels, self.intermediate_size], scale=1.0)
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config = self.get_config()
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inputs_dict = {"input_values": input_values}
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return config, inputs_dict
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def prepare_config_and_inputs_for_common(self):
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config, inputs_dict = self.prepare_config_and_inputs()
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return config, inputs_dict
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def prepare_config_and_inputs_for_model_class(self, model_class):
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input_values = floats_tensor([self.batch_size, self.num_channels, self.intermediate_size], scale=1.0)
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config = self.get_config()
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inputs_dict = {"input_values": input_values}
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return config, inputs_dict
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# Ignore copy
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def get_config(self):
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return DacConfig(
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encoder_hidden_size=self.encoder_hidden_size,
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downsampling_ratios=self.downsampling_ratios,
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decoder_hidden_size=self.decoder_hidden_size,
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n_codebooks=self.n_codebooks,
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codebook_size=self.codebook_size,
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codebook_dim=self.codebook_dim,
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quantizer_dropout=self.quantizer_dropout,
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commitment_loss_weight=self.commitment_loss_weight,
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codebook_loss_weight=self.codebook_loss_weight,
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)
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# Ignore copy
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def create_and_check_model_forward(self, config, inputs_dict):
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model = DacModel(config=config).to(torch_device).eval()
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input_values = inputs_dict["input_values"]
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result = model(input_values)
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self.parent.assertEqual(result.audio_values.shape, (self.batch_size, self.intermediate_size))
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@require_torch
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# Copied from transformers.tests.encodec.test_modeling_encodec.EncodecModelTest with Encodec->Dac
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class DacModelTest(ModelTesterMixin, PipelineTesterMixin, unittest.TestCase):
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all_model_classes = (DacModel,) if is_torch_available() else ()
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is_encoder_decoder = True
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test_pruning = False
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test_headmasking = False
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test_resize_embeddings = False
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pipeline_model_mapping = {"feature-extraction": DacModel} if is_torch_available() else {}
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def _prepare_for_class(self, inputs_dict, model_class, return_labels=False):
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# model does not have attention and does not support returning hidden states
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inputs_dict = super()._prepare_for_class(inputs_dict, model_class, return_labels=return_labels)
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if "output_attentions" in inputs_dict:
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inputs_dict.pop("output_attentions")
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if "output_hidden_states" in inputs_dict:
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inputs_dict.pop("output_hidden_states")
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return inputs_dict
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def setUp(self):
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self.model_tester = DacModelTester(self)
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self.config_tester = ConfigTester(
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self, config_class=DacConfig, hidden_size=37, common_properties=[], has_text_modality=False
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)
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def test_config(self):
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self.config_tester.run_common_tests()
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def test_model_forward(self):
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config_and_inputs = self.model_tester.prepare_config_and_inputs()
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self.model_tester.create_and_check_model_forward(*config_and_inputs)
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def test_forward_signature(self):
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config, _ = self.model_tester.prepare_config_and_inputs_for_common()
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for model_class in self.all_model_classes:
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model = model_class(config)
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signature = inspect.signature(model.forward)
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# signature.parameters is an OrderedDict => so arg_names order is deterministic
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arg_names = [*signature.parameters.keys()]
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# Ignore copy
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expected_arg_names = ["input_values", "n_quantizers", "return_dict"]
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self.assertListEqual(arg_names[: len(expected_arg_names)], expected_arg_names)
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@unittest.skip("The DacModel is not transformers based, thus it does not have `inputs_embeds` logics")
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def test_inputs_embeds(self):
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pass
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@unittest.skip("The DacModel is not transformers based, thus it does not have `inputs_embeds` logics")
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def test_model_get_set_embeddings(self):
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pass
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@unittest.skip("The DacModel is not transformers based, thus it does not have the usual `attention` logic")
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def test_retain_grad_hidden_states_attentions(self):
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pass
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@unittest.skip("The DacModel is not transformers based, thus it does not have the usual `attention` logic")
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def test_torchscript_output_attentions(self):
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pass
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@unittest.skip("The DacModel is not transformers based, thus it does not have the usual `hidden_states` logic")
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def test_torchscript_output_hidden_state(self):
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pass
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def _create_and_check_torchscript(self, config, inputs_dict):
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if not self.test_torchscript:
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return
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configs_no_init = _config_zero_init(config) # To be sure we have no Nan
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configs_no_init.torchscript = True
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configs_no_init.return_dict = False
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for model_class in self.all_model_classes:
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model = model_class(config=configs_no_init)
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model.to(torch_device)
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model.eval()
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inputs = self._prepare_for_class(inputs_dict, model_class)
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main_input_name = model_class.main_input_name
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try:
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main_input = inputs[main_input_name]
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model(main_input)
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traced_model = torch.jit.trace(model, main_input)
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except RuntimeError:
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self.fail("Couldn't trace module.")
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with tempfile.TemporaryDirectory() as tmp_dir_name:
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pt_file_name = os.path.join(tmp_dir_name, "traced_model.pt")
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try:
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torch.jit.save(traced_model, pt_file_name)
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except Exception:
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self.fail("Couldn't save module.")
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try:
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loaded_model = torch.jit.load(pt_file_name)
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except Exception:
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self.fail("Couldn't load module.")
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model.to(torch_device)
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model.eval()
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loaded_model.to(torch_device)
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loaded_model.eval()
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model_state_dict = model.state_dict()
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loaded_model_state_dict = loaded_model.state_dict()
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non_persistent_buffers = {}
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for key in loaded_model_state_dict.keys():
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if key not in model_state_dict.keys():
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non_persistent_buffers[key] = loaded_model_state_dict[key]
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loaded_model_state_dict = {
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key: value for key, value in loaded_model_state_dict.items() if key not in non_persistent_buffers
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}
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self.assertEqual(set(model_state_dict.keys()), set(loaded_model_state_dict.keys()))
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model_buffers = list(model.buffers())
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for non_persistent_buffer in non_persistent_buffers.values():
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found_buffer = False
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for i, model_buffer in enumerate(model_buffers):
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if torch.equal(non_persistent_buffer, model_buffer):
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found_buffer = True
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break
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self.assertTrue(found_buffer)
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model_buffers.pop(i)
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model_buffers = list(model.buffers())
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for non_persistent_buffer in non_persistent_buffers.values():
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found_buffer = False
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for i, model_buffer in enumerate(model_buffers):
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if torch.equal(non_persistent_buffer, model_buffer):
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found_buffer = True
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break
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self.assertTrue(found_buffer)
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model_buffers.pop(i)
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models_equal = True
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for layer_name, p1 in model_state_dict.items():
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if layer_name in loaded_model_state_dict:
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p2 = loaded_model_state_dict[layer_name]
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if p1.data.ne(p2.data).sum() > 0:
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models_equal = False
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self.assertTrue(models_equal)
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# Avoid memory leak. Without this, each call increase RAM usage by ~20MB.
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# (Even with this call, there are still memory leak by ~0.04MB)
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self.clear_torch_jit_class_registry()
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@unittest.skip("The DacModel is not transformers based, thus it does not have the usual `attention` logic")
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def test_attention_outputs(self):
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pass
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@unittest.skip("The DacModel is not transformers based, thus it does not have the usual `hidden_states` logic")
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def test_hidden_states_output(self):
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pass
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@unittest.skip("No support for low_cpu_mem_usage=True.")
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def test_save_load_low_cpu_mem_usage(self):
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pass
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@unittest.skip("No support for low_cpu_mem_usage=True.")
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def test_save_load_low_cpu_mem_usage_checkpoints(self):
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pass
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@unittest.skip("No support for low_cpu_mem_usage=True.")
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def test_save_load_low_cpu_mem_usage_no_safetensors(self):
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pass
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def test_determinism(self):
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config, inputs_dict = self.model_tester.prepare_config_and_inputs_for_common()
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def check_determinism(first, second):
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# outputs are not tensors but list (since each sequence don't have the same frame_length)
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out_1 = first.cpu().numpy()
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out_2 = second.cpu().numpy()
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out_1 = out_1[~np.isnan(out_1)]
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out_2 = out_2[~np.isnan(out_2)]
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max_diff = np.amax(np.abs(out_1 - out_2))
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self.assertLessEqual(max_diff, 1e-5)
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for model_class in self.all_model_classes:
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model = model_class(config)
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model.to(torch_device)
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model.eval()
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with torch.no_grad():
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first = model(**self._prepare_for_class(inputs_dict, model_class))[0]
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second = model(**self._prepare_for_class(inputs_dict, model_class))[0]
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if isinstance(first, tuple) and isinstance(second, tuple):
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for tensor1, tensor2 in zip(first, second):
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check_determinism(tensor1, tensor2)
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else:
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check_determinism(first, second)
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def test_model_outputs_equivalence(self):
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config, inputs_dict = self.model_tester.prepare_config_and_inputs_for_common()
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def set_nan_tensor_to_zero(t):
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t[t != t] = 0
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return t
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def check_equivalence(model, tuple_inputs, dict_inputs, additional_kwargs={}):
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with torch.no_grad():
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tuple_output = model(**tuple_inputs, return_dict=False, **additional_kwargs)
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dict_output = model(**dict_inputs, return_dict=True, **additional_kwargs).to_tuple()
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def recursive_check(tuple_object, dict_object):
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if isinstance(tuple_object, (List, Tuple)):
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for tuple_iterable_value, dict_iterable_value in zip(tuple_object, dict_object):
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recursive_check(tuple_iterable_value, dict_iterable_value)
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elif isinstance(tuple_object, Dict):
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for tuple_iterable_value, dict_iterable_value in zip(
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tuple_object.values(), dict_object.values()
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):
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recursive_check(tuple_iterable_value, dict_iterable_value)
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elif tuple_object is None:
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return
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else:
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self.assertTrue(
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torch.allclose(
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set_nan_tensor_to_zero(tuple_object), set_nan_tensor_to_zero(dict_object), atol=1e-5
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),
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msg=(
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"Tuple and dict output are not equal. Difference:"
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f" {torch.max(torch.abs(tuple_object - dict_object))}. Tuple has `nan`:"
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f" {torch.isnan(tuple_object).any()} and `inf`: {torch.isinf(tuple_object)}. Dict has"
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f" `nan`: {torch.isnan(dict_object).any()} and `inf`: {torch.isinf(dict_object)}."
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),
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)
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recursive_check(tuple_output, dict_output)
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for model_class in self.all_model_classes:
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model = model_class(config)
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model.to(torch_device)
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model.eval()
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tuple_inputs = self._prepare_for_class(inputs_dict, model_class)
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dict_inputs = self._prepare_for_class(inputs_dict, model_class)
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check_equivalence(model, tuple_inputs, dict_inputs)
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# Ignore copy
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def test_initialization(self):
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config, inputs_dict = self.model_tester.prepare_config_and_inputs_for_common()
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configs_no_init = _config_zero_init(config)
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for model_class in self.all_model_classes:
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model = model_class(config=configs_no_init)
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for name, param in model.named_parameters():
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uniform_init_parms = ["conv", "in_proj", "out_proj", "codebook"]
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if param.requires_grad:
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if any(x in name for x in uniform_init_parms):
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self.assertTrue(
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-1.0 <= ((param.data.mean() * 1e9).round() / 1e9).item() <= 1.0,
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msg=f"Parameter {name} of model {model_class} seems not properly initialized",
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)
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def test_identity_shortcut(self):
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config, inputs_dict = self.model_tester.prepare_config_and_inputs()
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config.use_conv_shortcut = False
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self.model_tester.create_and_check_model_forward(config, inputs_dict)
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def normalize(arr):
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norm = np.linalg.norm(arr)
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normalized_arr = arr / norm
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return normalized_arr
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def compute_rmse(arr1, arr2):
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arr1_normalized = normalize(arr1)
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arr2_normalized = normalize(arr2)
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return np.sqrt(((arr1_normalized - arr2_normalized) ** 2).mean())
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@slow
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@require_torch
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class DacIntegrationTest(unittest.TestCase):
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def test_integration_16khz(self):
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expected_rmse = 0.004
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expected_encoder_sums_dict = {
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"loss": 24.8596,
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"quantized_representation": -0.0745,
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"audio_codes": 504.0948,
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"projected_latents": 0.0682,
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}
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librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
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model_name = "dac_16khz"
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model_id = "descript/{}".format(model_name)
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model = DacModel.from_pretrained(model_id, force_download=True).to(torch_device).eval()
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processor = AutoProcessor.from_pretrained(model_id)
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librispeech_dummy = librispeech_dummy.cast_column("audio", Audio(sampling_rate=processor.sampling_rate))
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audio_sample = librispeech_dummy[0]["audio"]["array"]
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inputs = processor(
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raw_audio=audio_sample,
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sampling_rate=processor.sampling_rate,
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return_tensors="pt",
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).to(torch_device)
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with torch.no_grad():
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encoder_outputs = model.encode(inputs["input_values"])
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expected_encoder_sums = torch.tensor(list(expected_encoder_sums_dict.values()), dtype=torch.float32)
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encoder_outputs_mean = torch.tensor([v.float().mean().cpu().item() for v in encoder_outputs.to_tuple()])
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# make sure audio encoded codes are correct
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self.assertTrue(torch.allclose(encoder_outputs_mean, expected_encoder_sums, atol=1e-3))
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_, quantized_representation, _, _ = encoder_outputs.to_tuple()
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input_values_dec = model.decode(quantized_representation)[0]
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input_values_enc_dec = model(inputs["input_values"])[1]
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# make sure forward and decode gives same result
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self.assertTrue(torch.allclose(input_values_dec, input_values_enc_dec, atol=1e-3))
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arr = inputs["input_values"][0].cpu().numpy()
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arr_enc_dec = input_values_enc_dec[0].cpu().numpy()
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max_length = min(arr_enc_dec.shape[-1], arr.shape[-1])
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arr_cut = arr[0, :max_length].copy()
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arr_enc_dec_cut = arr_enc_dec[:max_length].copy()
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# make sure audios are more or less equal
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rmse = compute_rmse(arr_cut, arr_enc_dec_cut)
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self.assertTrue(rmse < expected_rmse)
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def test_integration_24khz(self):
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expected_rmse = 0.0039
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|
|
expected_encoder_output_dict = {
|
|
"quantized_representation": torch.tensor([0.9807, 2.8212, 5.2514, 2.7241, 1.0426]),
|
|
"audio_codes": torch.tensor([919, 919, 234, 777, 234]),
|
|
"projected_latents": torch.tensor([-4.7822, -5.0046, -4.5574, -5.0363, -5.4271]),
|
|
}
|
|
librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
|
|
|
|
model_name = "dac_24khz"
|
|
|
|
model_id = "descript/{}".format(model_name)
|
|
model = DacModel.from_pretrained(model_id, force_download=True).to(torch_device).eval()
|
|
processor = AutoProcessor.from_pretrained(model_id)
|
|
|
|
librispeech_dummy = librispeech_dummy.cast_column("audio", Audio(sampling_rate=processor.sampling_rate))
|
|
audio_sample = librispeech_dummy[0]["audio"]["array"]
|
|
|
|
inputs = processor(
|
|
raw_audio=audio_sample,
|
|
sampling_rate=processor.sampling_rate,
|
|
return_tensors="pt",
|
|
).to(torch_device)
|
|
|
|
with torch.no_grad():
|
|
encoder_outputs = model.encode(inputs["input_values"])
|
|
|
|
expected_quantized_representation = encoder_outputs["quantized_representation"][0, 0, :5].cpu()
|
|
expected_audio_codes = encoder_outputs["audio_codes"][0, 0, :5].cpu()
|
|
expected_projected_latents = encoder_outputs["projected_latents"][0, 0, :5].cpu()
|
|
|
|
# make sure values are correct for audios slices
|
|
self.assertTrue(
|
|
torch.allclose(
|
|
expected_quantized_representation,
|
|
expected_encoder_output_dict["quantized_representation"],
|
|
atol=1e-3,
|
|
)
|
|
)
|
|
self.assertTrue(
|
|
torch.allclose(expected_audio_codes, expected_encoder_output_dict["audio_codes"], atol=1e-3)
|
|
)
|
|
self.assertTrue(
|
|
torch.allclose(
|
|
expected_projected_latents, expected_encoder_output_dict["projected_latents"], atol=1e-3
|
|
)
|
|
)
|
|
|
|
_, quantized_representation, _, _ = encoder_outputs.to_tuple()
|
|
input_values_dec = model.decode(quantized_representation)[0]
|
|
input_values_enc_dec = model(inputs["input_values"])[1]
|
|
|
|
# make sure forward and decode gives same result
|
|
self.assertTrue(torch.allclose(input_values_dec, input_values_enc_dec, atol=1e-3))
|
|
|
|
arr = inputs["input_values"][0].cpu().numpy()
|
|
arr_enc_dec = input_values_enc_dec[0].cpu().numpy()
|
|
|
|
max_length = min(arr_enc_dec.shape[-1], arr.shape[-1])
|
|
|
|
arr_cut = arr[0, :max_length].copy()
|
|
arr_enc_dec_cut = arr_enc_dec[:max_length].copy()
|
|
|
|
# make sure audios are more or less equal
|
|
rmse = compute_rmse(arr_cut, arr_enc_dec_cut)
|
|
self.assertTrue(rmse < expected_rmse)
|
|
|
|
def test_integration_44khz(self):
|
|
expected_rmse = 0.002
|
|
|
|
expected_encoder_sums_dict = {
|
|
"loss": 34.3612,
|
|
"quantized_representation": 0.0078,
|
|
"audio_codes": 509.6812,
|
|
"projected_latents": -0.1054,
|
|
}
|
|
librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
|
|
|
|
model_name = "dac_44khz"
|
|
|
|
model_id = "descript/{}".format(model_name)
|
|
model = DacModel.from_pretrained(model_id).to(torch_device).eval()
|
|
processor = AutoProcessor.from_pretrained(model_id)
|
|
|
|
librispeech_dummy = librispeech_dummy.cast_column("audio", Audio(sampling_rate=processor.sampling_rate))
|
|
audio_sample = librispeech_dummy[0]["audio"]["array"]
|
|
|
|
inputs = processor(
|
|
raw_audio=audio_sample,
|
|
sampling_rate=processor.sampling_rate,
|
|
return_tensors="pt",
|
|
).to(torch_device)
|
|
|
|
with torch.no_grad():
|
|
encoder_outputs = model.encode(inputs["input_values"])
|
|
|
|
expected_encoder_sums = torch.tensor(list(expected_encoder_sums_dict.values()), dtype=torch.float32)
|
|
encoder_outputs_mean = torch.tensor([v.float().mean().cpu().item() for v in encoder_outputs.to_tuple()])
|
|
|
|
# make sure audio encoded codes are correct
|
|
self.assertTrue(torch.allclose(encoder_outputs_mean, expected_encoder_sums, atol=1e-3))
|
|
|
|
_, quantized_representation, _, _ = encoder_outputs.to_tuple()
|
|
input_values_dec = model.decode(quantized_representation)[0]
|
|
input_values_enc_dec = model(inputs["input_values"])[1]
|
|
|
|
# make sure forward and decode gives same result
|
|
self.assertTrue(torch.allclose(input_values_dec, input_values_enc_dec, atol=1e-3))
|
|
|
|
arr = inputs["input_values"][0].cpu().numpy()
|
|
arr_enc_dec = input_values_enc_dec[0].cpu().numpy()
|
|
|
|
max_length = min(arr_enc_dec.shape[-1], arr.shape[-1])
|
|
|
|
arr_cut = arr[0, :max_length].copy()
|
|
arr_enc_dec_cut = arr_enc_dec[:max_length].copy()
|
|
|
|
# make sure audios are more or less equal
|
|
rmse = compute_rmse(arr_cut, arr_enc_dec_cut)
|
|
self.assertTrue(rmse < expected_rmse)
|
|
|
|
def test_integration_batch_16khz(self):
|
|
expected_rmse = 0.002
|
|
|
|
expected_encoder_sums_dict = {
|
|
"loss": 20.3913,
|
|
"quantized_representation": -0.0538,
|
|
"audio_codes": 487.8470,
|
|
"projected_latents": 0.0237,
|
|
}
|
|
|
|
librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
|
|
|
|
model_name = "dac_16khz"
|
|
|
|
model_id = "descript/{}".format(model_name)
|
|
model = DacModel.from_pretrained(model_id).to(torch_device)
|
|
processor = AutoProcessor.from_pretrained(model_id)
|
|
|
|
librispeech_dummy = librispeech_dummy.cast_column("audio", Audio(sampling_rate=processor.sampling_rate))
|
|
|
|
audio_samples = [np.array([audio_sample["array"]])[0] for audio_sample in librispeech_dummy[-2:]["audio"]]
|
|
|
|
inputs = processor(
|
|
raw_audio=audio_samples,
|
|
sampling_rate=processor.sampling_rate,
|
|
truncation=False,
|
|
return_tensors="pt",
|
|
).to(torch_device)
|
|
|
|
with torch.no_grad():
|
|
encoder_outputs = model.encode(inputs["input_values"])
|
|
|
|
expected_encoder_sums = torch.tensor(list(expected_encoder_sums_dict.values()), dtype=torch.float32)
|
|
encoder_outputs_mean = torch.tensor([v.float().mean().item() for v in encoder_outputs.to_tuple()])
|
|
|
|
# make sure audio encoded codes are correct
|
|
self.assertTrue(torch.allclose(encoder_outputs_mean, expected_encoder_sums, atol=1e-3))
|
|
|
|
_, quantized_representation, _, _ = encoder_outputs.to_tuple()
|
|
input_values_dec = model.decode(quantized_representation)[0]
|
|
input_values_enc_dec = model(inputs["input_values"])[1]
|
|
|
|
# make sure forward and decode gives same result
|
|
self.assertTrue(torch.allclose(input_values_dec, input_values_enc_dec, atol=1e-3))
|
|
|
|
arr = inputs["input_values"].cpu().numpy()
|
|
arr_enc_dec = input_values_enc_dec.cpu().numpy()
|
|
|
|
max_length = min(arr_enc_dec.shape[-1], arr.shape[-1])
|
|
|
|
arr_cut = arr[:, 0, :max_length].copy()
|
|
arr_enc_dec_cut = arr_enc_dec[:, :max_length].copy()
|
|
|
|
# make sure audios are more or less equal
|
|
rmse = compute_rmse(arr_cut, arr_enc_dec_cut)
|
|
self.assertTrue(rmse < expected_rmse)
|
|
|
|
def test_integration_batch_24khz(self):
|
|
expected_rmse = 0.002
|
|
|
|
expected_encoder_sums_dict = {
|
|
"loss": 24.2309,
|
|
"quantized_representation": 0.0520,
|
|
"audio_codes": 510.2700,
|
|
"projected_latents": -0.0076,
|
|
}
|
|
|
|
librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
|
|
|
|
model_name = "dac_24khz"
|
|
|
|
model_id = "descript/{}".format(model_name)
|
|
model = DacModel.from_pretrained(model_id).to(torch_device)
|
|
processor = AutoProcessor.from_pretrained(model_id)
|
|
|
|
librispeech_dummy = librispeech_dummy.cast_column("audio", Audio(sampling_rate=processor.sampling_rate))
|
|
|
|
audio_samples = [np.array([audio_sample["array"]])[0] for audio_sample in librispeech_dummy[-2:]["audio"]]
|
|
|
|
inputs = processor(
|
|
raw_audio=audio_samples,
|
|
sampling_rate=processor.sampling_rate,
|
|
truncation=False,
|
|
return_tensors="pt",
|
|
).to(torch_device)
|
|
|
|
with torch.no_grad():
|
|
encoder_outputs = model.encode(inputs["input_values"])
|
|
|
|
expected_encoder_sums = torch.tensor(list(expected_encoder_sums_dict.values()), dtype=torch.float32)
|
|
encoder_outputs_mean = torch.tensor([v.float().mean().cpu().item() for v in encoder_outputs.to_tuple()])
|
|
|
|
# make sure audio encoded codes are correct
|
|
self.assertTrue(torch.allclose(encoder_outputs_mean, expected_encoder_sums, atol=1e-3))
|
|
|
|
_, quantized_representation, _, _ = encoder_outputs.to_tuple()
|
|
input_values_dec = model.decode(quantized_representation)[0]
|
|
input_values_enc_dec = model(inputs["input_values"])[1]
|
|
|
|
# make sure forward and decode gives same result
|
|
self.assertTrue(torch.allclose(input_values_dec, input_values_enc_dec, atol=1e-3))
|
|
|
|
arr = inputs["input_values"].cpu().numpy()
|
|
arr_enc_dec = input_values_enc_dec.cpu().numpy()
|
|
|
|
max_length = min(arr_enc_dec.shape[-1], arr.shape[-1])
|
|
|
|
arr_cut = arr[:, 0, :max_length].copy()
|
|
arr_enc_dec_cut = arr_enc_dec[:, :max_length].copy()
|
|
|
|
# make sure audios are more or less equal
|
|
rmse = compute_rmse(arr_cut, arr_enc_dec_cut)
|
|
self.assertTrue(rmse < expected_rmse)
|
|
|
|
def test_integration_batch_44khz(self):
|
|
expected_rmse = 0.001
|
|
|
|
expected_encoder_sums_dict = {
|
|
"loss": 25.9233,
|
|
"quantized_representation": 0.0013,
|
|
"audio_codes": 528.5620,
|
|
"projected_latents": -0.1194,
|
|
}
|
|
|
|
librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
|
|
|
|
model_name = "dac_44khz"
|
|
|
|
model_id = "descript/{}".format(model_name)
|
|
model = DacModel.from_pretrained(model_id).to(torch_device)
|
|
processor = AutoProcessor.from_pretrained(model_id)
|
|
|
|
librispeech_dummy = librispeech_dummy.cast_column("audio", Audio(sampling_rate=processor.sampling_rate))
|
|
|
|
audio_samples = [np.array([audio_sample["array"]])[0] for audio_sample in librispeech_dummy[-2:]["audio"]]
|
|
|
|
inputs = processor(
|
|
raw_audio=audio_samples,
|
|
sampling_rate=processor.sampling_rate,
|
|
truncation=False,
|
|
return_tensors="pt",
|
|
).to(torch_device)
|
|
|
|
with torch.no_grad():
|
|
encoder_outputs = model.encode(inputs["input_values"])
|
|
|
|
expected_encoder_sums = torch.tensor(list(expected_encoder_sums_dict.values()), dtype=torch.float32)
|
|
encoder_outputs_mean = torch.tensor([v.float().mean().cpu().item() for v in encoder_outputs.to_tuple()])
|
|
|
|
# make sure audio encoded codes are correct
|
|
self.assertTrue(torch.allclose(encoder_outputs_mean, expected_encoder_sums, atol=1e-3))
|
|
|
|
_, quantized_representation, _, _ = encoder_outputs.to_tuple()
|
|
input_values_dec = model.decode(quantized_representation)[0]
|
|
input_values_enc_dec = model(inputs["input_values"])[1]
|
|
|
|
# make sure forward and decode gives same result
|
|
self.assertTrue(torch.allclose(input_values_dec, input_values_enc_dec, atol=1e-3))
|
|
|
|
arr = inputs["input_values"].cpu().numpy()
|
|
arr_enc_dec = input_values_enc_dec.cpu().numpy()
|
|
|
|
max_length = min(arr_enc_dec.shape[-1], arr.shape[-1])
|
|
|
|
arr_cut = arr[:, 0, :max_length].copy()
|
|
arr_enc_dec_cut = arr_enc_dec[:, :max_length].copy()
|
|
|
|
# make sure audios are more or less equal
|
|
rmse = compute_rmse(arr_cut, arr_enc_dec_cut)
|
|
self.assertTrue(rmse < expected_rmse)
|