# 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. import gc import unittest import numpy as np import torch from transformers import ( ClapAudioConfig, ClapConfig, ClapFeatureExtractor, ClapModel, ClapTextConfig, GPT2Config, GPT2Model, RobertaTokenizer, SpeechT5HifiGan, SpeechT5HifiGanConfig, T5Config, T5EncoderModel, T5Tokenizer, ) from diffusers import ( AudioLDM2Pipeline, AudioLDM2ProjectionModel, AudioLDM2UNet2DConditionModel, AutoencoderKL, DDIMScheduler, LMSDiscreteScheduler, PNDMScheduler, ) from diffusers.utils.testing_utils import enable_full_determinism, nightly, torch_device from ..pipeline_params import TEXT_TO_AUDIO_BATCH_PARAMS, TEXT_TO_AUDIO_PARAMS from ..test_pipelines_common import PipelineTesterMixin enable_full_determinism() class AudioLDM2PipelineFastTests(PipelineTesterMixin, unittest.TestCase): pipeline_class = AudioLDM2Pipeline params = TEXT_TO_AUDIO_PARAMS batch_params = TEXT_TO_AUDIO_BATCH_PARAMS required_optional_params = frozenset( [ "num_inference_steps", "num_waveforms_per_prompt", "generator", "latents", "output_type", "return_dict", "callback", "callback_steps", ] ) def get_dummy_components(self): torch.manual_seed(0) unet = AudioLDM2UNet2DConditionModel( block_out_channels=(32, 64), layers_per_block=2, sample_size=32, in_channels=4, out_channels=4, down_block_types=("DownBlock2D", "CrossAttnDownBlock2D"), up_block_types=("CrossAttnUpBlock2D", "UpBlock2D"), cross_attention_dim=([None, 16, 32], [None, 16, 32]), ) scheduler = DDIMScheduler( beta_start=0.00085, beta_end=0.012, beta_schedule="scaled_linear", clip_sample=False, set_alpha_to_one=False, ) torch.manual_seed(0) vae = AutoencoderKL( block_out_channels=[32, 64], in_channels=1, out_channels=1, down_block_types=["DownEncoderBlock2D", "DownEncoderBlock2D"], up_block_types=["UpDecoderBlock2D", "UpDecoderBlock2D"], latent_channels=4, ) torch.manual_seed(0) text_branch_config = ClapTextConfig( bos_token_id=0, eos_token_id=2, hidden_size=16, intermediate_size=37, layer_norm_eps=1e-05, num_attention_heads=2, num_hidden_layers=2, pad_token_id=1, vocab_size=1000, projection_dim=16, ) audio_branch_config = ClapAudioConfig( spec_size=64, window_size=4, num_mel_bins=64, intermediate_size=37, layer_norm_eps=1e-05, depths=[2, 2], num_attention_heads=[2, 2], num_hidden_layers=2, hidden_size=192, projection_dim=16, patch_size=2, patch_stride=2, patch_embed_input_channels=4, ) text_encoder_config = ClapConfig.from_text_audio_configs( text_config=text_branch_config, audio_config=audio_branch_config, projection_dim=16 ) text_encoder = ClapModel(text_encoder_config) tokenizer = RobertaTokenizer.from_pretrained("hf-internal-testing/tiny-random-roberta", model_max_length=77) feature_extractor = ClapFeatureExtractor.from_pretrained( "hf-internal-testing/tiny-random-ClapModel", hop_length=7900 ) torch.manual_seed(0) text_encoder_2_config = T5Config( vocab_size=32100, d_model=32, d_ff=37, d_kv=8, num_heads=2, num_layers=2, ) text_encoder_2 = T5EncoderModel(text_encoder_2_config) tokenizer_2 = T5Tokenizer.from_pretrained("hf-internal-testing/tiny-random-T5Model", model_max_length=77) torch.manual_seed(0) language_model_config = GPT2Config( n_embd=16, n_head=2, n_layer=2, vocab_size=1000, n_ctx=99, n_positions=99, ) language_model = GPT2Model(language_model_config) language_model.config.max_new_tokens = 8 torch.manual_seed(0) projection_model = AudioLDM2ProjectionModel(text_encoder_dim=16, text_encoder_1_dim=32, langauge_model_dim=16) vocoder_config = SpeechT5HifiGanConfig( model_in_dim=8, sampling_rate=16000, upsample_initial_channel=16, upsample_rates=[2, 2], upsample_kernel_sizes=[4, 4], resblock_kernel_sizes=[3, 7], resblock_dilation_sizes=[[1, 3, 5], [1, 3, 5]], normalize_before=False, ) vocoder = SpeechT5HifiGan(vocoder_config) components = { "unet": unet, "scheduler": scheduler, "vae": vae, "text_encoder": text_encoder, "text_encoder_2": text_encoder_2, "tokenizer": tokenizer, "tokenizer_2": tokenizer_2, "feature_extractor": feature_extractor, "language_model": language_model, "projection_model": projection_model, "vocoder": vocoder, } return components def get_dummy_inputs(self, device, seed=0): if str(device).startswith("mps"): generator = torch.manual_seed(seed) else: generator = torch.Generator(device=device).manual_seed(seed) inputs = { "prompt": "A hammer hitting a wooden surface", "generator": generator, "num_inference_steps": 2, "guidance_scale": 6.0, } return inputs def test_audioldm2_ddim(self): device = "cpu" # ensure determinism for the device-dependent torch.Generator components = self.get_dummy_components() audioldm_pipe = AudioLDM2Pipeline(**components) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) inputs = self.get_dummy_inputs(device) output = audioldm_pipe(**inputs) audio = output.audios[0] assert audio.ndim == 1 assert len(audio) == 256 audio_slice = audio[:10] expected_slice = np.array( [0.0025, 0.0018, 0.0018, -0.0023, -0.0026, -0.0020, -0.0026, -0.0021, -0.0027, -0.0020] ) assert np.abs(audio_slice - expected_slice).max() < 1e-4 def test_audioldm2_prompt_embeds(self): components = self.get_dummy_components() audioldm_pipe = AudioLDM2Pipeline(**components) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) inputs = self.get_dummy_inputs(torch_device) inputs["prompt"] = 3 * [inputs["prompt"]] # forward output = audioldm_pipe(**inputs) audio_1 = output.audios[0] inputs = self.get_dummy_inputs(torch_device) prompt = 3 * [inputs.pop("prompt")] text_inputs = audioldm_pipe.tokenizer( prompt, padding="max_length", max_length=audioldm_pipe.tokenizer.model_max_length, truncation=True, return_tensors="pt", ) text_inputs = text_inputs["input_ids"].to(torch_device) clap_prompt_embeds = audioldm_pipe.text_encoder.get_text_features(text_inputs) clap_prompt_embeds = clap_prompt_embeds[:, None, :] text_inputs = audioldm_pipe.tokenizer_2( prompt, padding="max_length", max_length=True, truncation=True, return_tensors="pt", ) text_inputs = text_inputs["input_ids"].to(torch_device) t5_prompt_embeds = audioldm_pipe.text_encoder_2( text_inputs, ) t5_prompt_embeds = t5_prompt_embeds[0] projection_embeds = audioldm_pipe.projection_model(clap_prompt_embeds, t5_prompt_embeds)[0] generated_prompt_embeds = audioldm_pipe.generate_language_model(projection_embeds, max_new_tokens=8) inputs["prompt_embeds"] = t5_prompt_embeds inputs["generated_prompt_embeds"] = generated_prompt_embeds # forward output = audioldm_pipe(**inputs) audio_2 = output.audios[0] assert np.abs(audio_1 - audio_2).max() < 1e-2 def test_audioldm2_negative_prompt_embeds(self): components = self.get_dummy_components() audioldm_pipe = AudioLDM2Pipeline(**components) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) inputs = self.get_dummy_inputs(torch_device) negative_prompt = 3 * ["this is a negative prompt"] inputs["negative_prompt"] = negative_prompt inputs["prompt"] = 3 * [inputs["prompt"]] # forward output = audioldm_pipe(**inputs) audio_1 = output.audios[0] inputs = self.get_dummy_inputs(torch_device) prompt = 3 * [inputs.pop("prompt")] embeds = [] generated_embeds = [] for p in [prompt, negative_prompt]: text_inputs = audioldm_pipe.tokenizer( p, padding="max_length", max_length=audioldm_pipe.tokenizer.model_max_length, truncation=True, return_tensors="pt", ) text_inputs = text_inputs["input_ids"].to(torch_device) clap_prompt_embeds = audioldm_pipe.text_encoder.get_text_features(text_inputs) clap_prompt_embeds = clap_prompt_embeds[:, None, :] text_inputs = audioldm_pipe.tokenizer_2( prompt, padding="max_length", max_length=True if len(embeds) == 0 else embeds[0].shape[1], truncation=True, return_tensors="pt", ) text_inputs = text_inputs["input_ids"].to(torch_device) t5_prompt_embeds = audioldm_pipe.text_encoder_2( text_inputs, ) t5_prompt_embeds = t5_prompt_embeds[0] projection_embeds = audioldm_pipe.projection_model(clap_prompt_embeds, t5_prompt_embeds)[0] generated_prompt_embeds = audioldm_pipe.generate_language_model(projection_embeds, max_new_tokens=8) embeds.append(t5_prompt_embeds) generated_embeds.append(generated_prompt_embeds) inputs["prompt_embeds"], inputs["negative_prompt_embeds"] = embeds inputs["generated_prompt_embeds"], inputs["negative_generated_prompt_embeds"] = generated_embeds # forward output = audioldm_pipe(**inputs) audio_2 = output.audios[0] assert np.abs(audio_1 - audio_2).max() < 1e-2 def test_audioldm2_negative_prompt(self): device = "cpu" # ensure determinism for the device-dependent torch.Generator components = self.get_dummy_components() components["scheduler"] = PNDMScheduler(skip_prk_steps=True) audioldm_pipe = AudioLDM2Pipeline(**components) audioldm_pipe = audioldm_pipe.to(device) audioldm_pipe.set_progress_bar_config(disable=None) inputs = self.get_dummy_inputs(device) negative_prompt = "egg cracking" output = audioldm_pipe(**inputs, negative_prompt=negative_prompt) audio = output.audios[0] assert audio.ndim == 1 assert len(audio) == 256 audio_slice = audio[:10] expected_slice = np.array( [0.0025, 0.0018, 0.0018, -0.0023, -0.0026, -0.0020, -0.0026, -0.0021, -0.0027, -0.0020] ) assert np.abs(audio_slice - expected_slice).max() < 1e-4 def test_audioldm2_num_waveforms_per_prompt(self): device = "cpu" # ensure determinism for the device-dependent torch.Generator components = self.get_dummy_components() components["scheduler"] = PNDMScheduler(skip_prk_steps=True) audioldm_pipe = AudioLDM2Pipeline(**components) audioldm_pipe = audioldm_pipe.to(device) audioldm_pipe.set_progress_bar_config(disable=None) prompt = "A hammer hitting a wooden surface" # test num_waveforms_per_prompt=1 (default) audios = audioldm_pipe(prompt, num_inference_steps=2).audios assert audios.shape == (1, 256) # test num_waveforms_per_prompt=1 (default) for batch of prompts batch_size = 2 audios = audioldm_pipe([prompt] * batch_size, num_inference_steps=2).audios assert audios.shape == (batch_size, 256) # test num_waveforms_per_prompt for single prompt num_waveforms_per_prompt = 2 audios = audioldm_pipe(prompt, num_inference_steps=2, num_waveforms_per_prompt=num_waveforms_per_prompt).audios assert audios.shape == (num_waveforms_per_prompt, 256) # test num_waveforms_per_prompt for batch of prompts batch_size = 2 audios = audioldm_pipe( [prompt] * batch_size, num_inference_steps=2, num_waveforms_per_prompt=num_waveforms_per_prompt ).audios assert audios.shape == (batch_size * num_waveforms_per_prompt, 256) def test_audioldm2_audio_length_in_s(self): device = "cpu" # ensure determinism for the device-dependent torch.Generator components = self.get_dummy_components() audioldm_pipe = AudioLDM2Pipeline(**components) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) vocoder_sampling_rate = audioldm_pipe.vocoder.config.sampling_rate inputs = self.get_dummy_inputs(device) output = audioldm_pipe(audio_length_in_s=0.016, **inputs) audio = output.audios[0] assert audio.ndim == 1 assert len(audio) / vocoder_sampling_rate == 0.016 output = audioldm_pipe(audio_length_in_s=0.032, **inputs) audio = output.audios[0] assert audio.ndim == 1 assert len(audio) / vocoder_sampling_rate == 0.032 def test_audioldm2_vocoder_model_in_dim(self): components = self.get_dummy_components() audioldm_pipe = AudioLDM2Pipeline(**components) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) prompt = ["hey"] output = audioldm_pipe(prompt, num_inference_steps=1) audio_shape = output.audios.shape assert audio_shape == (1, 256) config = audioldm_pipe.vocoder.config config.model_in_dim *= 2 audioldm_pipe.vocoder = SpeechT5HifiGan(config).to(torch_device) output = audioldm_pipe(prompt, num_inference_steps=1) audio_shape = output.audios.shape # waveform shape is unchanged, we just have 2x the number of mel channels in the spectrogram assert audio_shape == (1, 256) def test_attention_slicing_forward_pass(self): self._test_attention_slicing_forward_pass(test_mean_pixel_difference=False) @unittest.skip("Raises a not implemented error in AudioLDM2") def test_xformers_attention_forwardGenerator_pass(self): pass def test_dict_tuple_outputs_equivalent(self): # increase tolerance from 1e-4 -> 2e-4 to account for large composite model super().test_dict_tuple_outputs_equivalent(expected_max_difference=2e-4) def test_inference_batch_single_identical(self): # increase tolerance from 1e-4 -> 2e-4 to account for large composite model self._test_inference_batch_single_identical(expected_max_diff=2e-4) def test_save_load_local(self): # increase tolerance from 1e-4 -> 2e-4 to account for large composite model super().test_save_load_local(expected_max_difference=2e-4) def test_save_load_optional_components(self): # increase tolerance from 1e-4 -> 2e-4 to account for large composite model super().test_save_load_optional_components(expected_max_difference=2e-4) def test_to_dtype(self): components = self.get_dummy_components() pipe = self.pipeline_class(**components) pipe.set_progress_bar_config(disable=None) # The method component.dtype returns the dtype of the first parameter registered in the model, not the # dtype of the entire model. In the case of CLAP, the first parameter is a float64 constant (logit scale) model_dtypes = {key: component.dtype for key, component in components.items() if hasattr(component, "dtype")} # Without the logit scale parameters, everything is float32 model_dtypes.pop("text_encoder") self.assertTrue(all(dtype == torch.float32 for dtype in model_dtypes.values())) # the CLAP sub-models are float32 model_dtypes["clap_text_branch"] = components["text_encoder"].text_model.dtype self.assertTrue(all(dtype == torch.float32 for dtype in model_dtypes.values())) # Once we send to fp16, all params are in half-precision, including the logit scale pipe.to(torch_dtype=torch.float16) model_dtypes = {key: component.dtype for key, component in components.items() if hasattr(component, "dtype")} self.assertTrue(all(dtype == torch.float16 for dtype in model_dtypes.values())) def test_sequential_cpu_offload_forward_pass(self): pass @nightly class AudioLDM2PipelineSlowTests(unittest.TestCase): def tearDown(self): super().tearDown() gc.collect() torch.cuda.empty_cache() def get_inputs(self, device, generator_device="cpu", dtype=torch.float32, seed=0): generator = torch.Generator(device=generator_device).manual_seed(seed) latents = np.random.RandomState(seed).standard_normal((1, 8, 128, 16)) latents = torch.from_numpy(latents).to(device=device, dtype=dtype) inputs = { "prompt": "A hammer hitting a wooden surface", "latents": latents, "generator": generator, "num_inference_steps": 3, "guidance_scale": 2.5, } return inputs def test_audioldm2(self): audioldm_pipe = AudioLDM2Pipeline.from_pretrained("cvssp/audioldm2") audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) inputs = self.get_inputs(torch_device) inputs["num_inference_steps"] = 25 audio = audioldm_pipe(**inputs).audios[0] assert audio.ndim == 1 assert len(audio) == 81952 # check the portion of the generated audio with the largest dynamic range (reduces flakiness) audio_slice = audio[17275:17285] expected_slice = np.array([0.0791, 0.0666, 0.1158, 0.1227, 0.1171, -0.2880, -0.1940, -0.0283, -0.0126, 0.1127]) max_diff = np.abs(expected_slice - audio_slice).max() assert max_diff < 1e-3 def test_audioldm2_lms(self): audioldm_pipe = AudioLDM2Pipeline.from_pretrained("cvssp/audioldm2") audioldm_pipe.scheduler = LMSDiscreteScheduler.from_config(audioldm_pipe.scheduler.config) audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) inputs = self.get_inputs(torch_device) audio = audioldm_pipe(**inputs).audios[0] assert audio.ndim == 1 assert len(audio) == 81952 # check the portion of the generated audio with the largest dynamic range (reduces flakiness) audio_slice = audio[31390:31400] expected_slice = np.array( [-0.1318, -0.0577, 0.0446, -0.0573, 0.0659, 0.1074, -0.2600, 0.0080, -0.2190, -0.4301] ) max_diff = np.abs(expected_slice - audio_slice).max() assert max_diff < 1e-3 def test_audioldm2_large(self): audioldm_pipe = AudioLDM2Pipeline.from_pretrained("cvssp/audioldm2-large") audioldm_pipe = audioldm_pipe.to(torch_device) audioldm_pipe.set_progress_bar_config(disable=None) inputs = self.get_inputs(torch_device) audio = audioldm_pipe(**inputs).audios[0] assert audio.ndim == 1 assert len(audio) == 81952 # check the portion of the generated audio with the largest dynamic range (reduces flakiness) audio_slice = audio[8825:8835] expected_slice = np.array( [-0.1829, -0.1461, 0.0759, -0.1493, -0.1396, 0.5783, 0.3001, -0.3038, -0.0639, -0.2244] ) max_diff = np.abs(expected_slice - audio_slice).max() assert max_diff < 1e-3