import os import os.path as osp import random import torch import torchaudio import torchvision.utils from pytorch_fid.fid_score import calculate_frechet_distance from torch import distributed from tqdm import tqdm from transformers import Wav2Vec2ForCTC import torch.nn.functional as F import numpy as np import trainer.eval.evaluator as evaluator from data.audio.paired_voice_audio_dataset import load_tsv_aligned_codes from data.audio.unsupervised_audio_dataset import load_audio from data.audio.voice_tokenizer import VoiceBpeTokenizer from models.clip.mel_text_clip import MelTextCLIP from models.audio.tts.tacotron2 import text_to_sequence from scripts.audio.gen.speech_synthesis_utils import load_discrete_vocoder_diffuser, wav_to_mel, load_speech_dvae, \ convert_mel_to_codes, load_univnet_vocoder, wav_to_univnet_mel, load_clvp from trainer.injectors.audio_injectors import denormalize_mel, TorchMelSpectrogramInjector, normalize_mel from utils.util import ceil_multiple, opt_get, load_model_from_config, pad_or_truncate class AudioDiffusionFid(evaluator.Evaluator): """ Evaluator produces generate from a diffusion model, then uses a CLIP model to judge the similarity between text & speech. This evaluator is kind of a mess. It has been repeatedly modified to work with several different model types, which means it is bloated beyond belief. I would not recommend attempting to understand what is going on here. """ def __init__(self, model, opt_eval, env): super().__init__(model, opt_eval, env, uses_all_ddp=True) self.real_path = opt_eval['eval_tsv'] self.data = load_tsv_aligned_codes(self.real_path) # Deterministically shuffle the data. ostate = random.getstate() random.seed(5) random.shuffle(self.data) random.setstate(ostate) if 'clip_dataset' in opt_eval.keys(): self.data = self.data[:opt_eval['clip_dataset']] if distributed.is_initialized() and distributed.get_world_size() > 1: self.skip = distributed.get_world_size() # One batch element per GPU. else: self.skip = 1 diffusion_steps = opt_get(opt_eval, ['diffusion_steps'], 50) diffusion_schedule = opt_get(env['opt'], ['steps', 'generator', 'injectors', 'diffusion', 'beta_schedule', 'schedule_name'], None) if diffusion_schedule is None: print("Unable to infer diffusion schedule from master options. Getting it from eval (or guessing).") diffusion_schedule = opt_get(opt_eval, ['diffusion_schedule'], 'cosine') conditioning_free_diffusion_enabled = opt_get(opt_eval, ['conditioning_free'], False) conditioning_free_k = opt_get(opt_eval, ['conditioning_free_k'], 1) self.diffuser = load_discrete_vocoder_diffuser(desired_diffusion_steps=diffusion_steps, schedule=diffusion_schedule, enable_conditioning_free_guidance=conditioning_free_diffusion_enabled, conditioning_free_k=conditioning_free_k) self.bpe_tokenizer = VoiceBpeTokenizer('../experiments/bpe_lowercase_asr_256.json') self.dev = self.env['device'] mode = opt_get(opt_eval, ['diffusion_type'], 'tts') self.local_modules = {} if mode == 'tts': self.diffusion_fn = self.perform_diffusion_tts elif mode == 'original_vocoder': self.local_modules['dvae'] = load_speech_dvae().cpu() self.diffusion_fn = self.perform_original_diffusion_vocoder elif mode == 'vocoder': self.local_modules['dvae'] = load_speech_dvae().cpu() self.diffusion_fn = self.perform_diffusion_vocoder elif mode == 'ctc_to_mel': self.diffusion_fn = self.perform_diffusion_ctc self.local_modules['vocoder'] = load_univnet_vocoder().cpu() self.local_modules['clvp'] = load_clvp() elif 'tts9_mel' in mode: mel_means, self.mel_max, self.mel_min, mel_stds, mel_vars = torch.load('../experiments/univnet_mel_norms.pth') self.local_modules['dvae'] = load_speech_dvae().cpu() self.local_modules['vocoder'] = load_univnet_vocoder().cpu() self.diffusion_fn = self.perform_diffusion_tts9_mel_from_codes if mode == 'tts9_mel_autoin': self.local_modules['autoregressive'] = load_model_from_config("../experiments/train_gpt_tts_unified.yml", model_name='gpt', also_load_savepoint=False, load_path='../experiments/unified_large_diverse_basis.pth', device=torch.device('cpu')).cuda().eval() self.tts9_codegen = self.tts9_get_autoregressive_codes else: self.tts9_codegen = self.tts9_get_dvae_codes elif 'tfd' == mode: self.diffusion_fn = self.perform_diffusion_tfd self.local_modules['vocoder'] = load_univnet_vocoder().cpu() elif 'tfd_ar' == mode: self.local_modules['dvae'] = load_speech_dvae().cpu() self.local_modules['autoregressive'] = load_model_from_config("../experiments/train_gpt_tts_unified.yml", model_name='gpt', also_load_savepoint=False, load_path='../experiments/tortoise_ar.pth', device=torch.device('cpu')).cuda().eval() self.diffusion_fn = self.perform_diffusion_tfd_ar_prior self.local_modules['vocoder'] = load_univnet_vocoder().cpu() def perform_diffusion_tts(self, audio, codes, text, sample_rate=5500): real_resampled = torchaudio.functional.resample(audio, 22050, sample_rate).unsqueeze(0) aligned_codes_compression_factor = sample_rate * 221 // 11025 output_size = codes.shape[-1]*aligned_codes_compression_factor padded_size = ceil_multiple(output_size, 2048) padding_added = padded_size - output_size padding_needed_for_codes = padding_added // aligned_codes_compression_factor if padding_needed_for_codes > 0: codes = F.pad(codes, (0, padding_needed_for_codes)) output_shape = (1, 1, padded_size) gen = self.diffuser.p_sample_loop(self.model, output_shape, model_kwargs={'tokens': codes.unsqueeze(0), 'conditioning_input': real_resampled}) return gen, real_resampled, sample_rate def perform_original_diffusion_vocoder(self, audio, codes, text, sample_rate=11025): mel = wav_to_mel(audio) mel_codes = convert_mel_to_codes(self.local_modules['dvae'], mel) back_to_mel = self.local_modules['dvae'].decode(mel_codes)[0] orig_audio = audio real_resampled = torchaudio.functional.resample(audio, 22050, sample_rate).unsqueeze(0) output_size = real_resampled.shape[-1] aligned_mel_compression_factor = output_size // back_to_mel.shape[-1] padded_size = ceil_multiple(output_size, 2048) padding_added = padded_size - output_size padding_needed_for_codes = padding_added // aligned_mel_compression_factor if padding_needed_for_codes > 0: back_to_mel = F.pad(back_to_mel, (0, padding_needed_for_codes)) output_shape = (1, 1, padded_size) gen = self.diffuser.p_sample_loop(self.model, output_shape, model_kwargs={'spectrogram': back_to_mel, 'conditioning_input': orig_audio.unsqueeze(0)}) # Pop it back down to 5.5kHz for an accurate comparison with the other diffusers. real_resampled = torchaudio.functional.resample(real_resampled.squeeze(0), sample_rate, 5500).unsqueeze(0) gen = torchaudio.functional.resample(gen.squeeze(0), sample_rate, 5500).unsqueeze(0) return gen, real_resampled, 5500 def perform_diffusion_vocoder(self, audio, codes, text, sample_rate=5500): mel = wav_to_mel(audio) mel_codes = convert_mel_to_codes(self.local_modules['dvae'], mel) text_codes = text_to_sequence(text) real_resampled = torchaudio.functional.resample(audio, 22050, sample_rate).unsqueeze(0) output_size = real_resampled.shape[-1] aligned_codes_compression_factor = output_size // mel_codes.shape[-1] padded_size = ceil_multiple(output_size, 2048) padding_added = padded_size - output_size padding_needed_for_codes = padding_added // aligned_codes_compression_factor if padding_needed_for_codes > 0: mel_codes = F.pad(mel_codes, (0, padding_needed_for_codes)) output_shape = (1, 1, padded_size) gen = self.diffuser.p_sample_loop(self.model, output_shape, model_kwargs={'tokens': mel_codes, 'conditioning_input': audio.unsqueeze(0), 'unaligned_input': torch.tensor(text_codes, device=audio.device).unsqueeze(0)}) return gen, real_resampled, sample_rate def tts9_get_autoregressive_codes(self, mel, text): mel_codes = convert_mel_to_codes(self.local_modules['dvae'], mel) text_codes = torch.LongTensor(self.bpe_tokenizer.encode(text)).unsqueeze(0).to(mel.device) cond_inputs = pad_or_truncate(mel, 132300//256).unsqueeze(1) mlc = self.local_modules['autoregressive'].mel_length_compression auto_latents = self.local_modules['autoregressive'](cond_inputs, text_codes, torch.tensor([text_codes.shape[-1]], device=mel.device), mel_codes, torch.tensor([mel_codes.shape[-1]*mlc], device=mel.device), text_first=True, raw_mels=None, return_latent=True, clip_inputs=False) return auto_latents def tts9_get_dvae_codes(self, mel, text): return convert_mel_to_codes(self.local_modules['dvae'], mel) def perform_diffusion_tts9_mel_from_codes(self, audio, codes, text): SAMPLE_RATE = 24000 mel = wav_to_mel(audio) mel_codes = self.tts9_codegen(mel, text) real_resampled = torchaudio.functional.resample(audio, 22050, SAMPLE_RATE).unsqueeze(0) univnet_mel = wav_to_univnet_mel(real_resampled, do_normalization=False) # to be used for a conditioning input, but also guides output shape. output_shape = univnet_mel.shape gen_mel = self.diffuser.p_sample_loop(self.model, output_shape, model_kwargs={'aligned_conditioning': mel_codes, 'conditioning_input': univnet_mel}) # denormalize mel gen_mel = denormalize_mel(gen_mel) gen_wav = self.local_modules['vocoder'].inference(gen_mel) real_dec = self.local_modules['vocoder'].inference(univnet_mel) return gen_wav.float(), real_dec, SAMPLE_RATE def perform_diffusion_ctc(self, audio, codes, text): SAMPLE_RATE = 24000 text_codes = torch.LongTensor(self.bpe_tokenizer.encode(text)).unsqueeze(0).to(audio.device) clvp_latent = self.local_modules['clvp'].embed_text(text_codes) real_resampled = torchaudio.functional.resample(audio, 22050, SAMPLE_RATE).unsqueeze(0) univnet_mel = wav_to_univnet_mel(real_resampled, do_normalization=True) output_shape = univnet_mel.shape cond_mel = TorchMelSpectrogramInjector({'n_mel_channels': 100, 'mel_fmax': 11000, 'filter_length': 8000, 'normalize': True, 'true_normalization': True, 'in': 'in', 'out': 'out'}, {})({'in': audio})['out'] gen_mel = self.diffuser.p_sample_loop(self.model, output_shape, model_kwargs={'codes': codes.unsqueeze(0), 'conditioning_input': cond_mel, 'type': torch.tensor([0], device=codes.device), 'clvp_input': clvp_latent}) gen_mel_denorm = denormalize_mel(gen_mel) gen_wav = self.local_modules['vocoder'].inference(gen_mel_denorm) real_dec = self.local_modules['vocoder'].inference(denormalize_mel(univnet_mel)) return gen_wav.float(), real_dec, gen_mel, univnet_mel, SAMPLE_RATE def perform_diffusion_tfd(self, audio, codes, text): SAMPLE_RATE = 24000 audio_resampled = torchaudio.functional.resample(audio, 22050, SAMPLE_RATE).unsqueeze(0) vmel = wav_to_mel(audio) umel = wav_to_univnet_mel(audio_resampled, do_normalization=True) gen_mel = self.diffuser.p_sample_loop(self.model, umel.shape, model_kwargs={'truth_mel': vmel, 'conditioning_input': None, 'disable_diversity': True}) gen_wav = self.local_modules['vocoder'].inference(denormalize_mel(gen_mel)) real_dec = self.local_modules['vocoder'].inference(denormalize_mel(umel)) return gen_wav.float(), real_dec, gen_mel, umel, SAMPLE_RATE def perform_diffusion_tfd_ar_prior(self, audio, codes, text): SAMPLE_RATE = 24000 audio_resampled = torchaudio.functional.resample(audio, 22050, SAMPLE_RATE).unsqueeze(0) vmel = wav_to_mel(audio) umel = wav_to_univnet_mel(audio_resampled, do_normalization=True) mel_codes = convert_mel_to_codes(self.local_modules['dvae'], vmel) text_codes = torch.LongTensor(self.bpe_tokenizer.encode(text)).unsqueeze(0).to(vmel.device) cond_inputs = pad_or_truncate(vmel, 132300//256).unsqueeze(1) mlc = self.local_modules['autoregressive'].mel_length_compression auto_latents = self.local_modules['autoregressive'](cond_inputs, text_codes, torch.tensor([text_codes.shape[-1]], device=vmel.device), mel_codes, torch.tensor([mel_codes.shape[-1]*mlc], device=vmel.device), text_first=True, raw_mels=None, return_latent=True) gen_mel = self.diffuser.p_sample_loop(self.model, umel.shape, model_kwargs={'codes': auto_latents}) gen_wav = self.local_modules['vocoder'].inference(denormalize_mel(gen_mel)) real_dec = self.local_modules['vocoder'].inference(denormalize_mel(umel)) return gen_wav.float(), real_dec, gen_mel, umel, SAMPLE_RATE def load_projector(self): """ Builds the CLIP model used to project speech into a latent. This model has fixed parameters and a fixed loading path for the time being. """ model = MelTextCLIP(dim_text=512, dim_latent=512, dim_speech=512, num_text_tokens=148, text_enc_depth=8, text_seq_len=400, text_heads=8, speech_enc_depth=10, speech_heads=8, speech_seq_len=1000, text_mask_percentage=.15, voice_mask_percentage=.15) weights = torch.load('../experiments/clip_text_to_voice_for_speech_fid.pth') model.load_state_dict(weights) return model def project(self, projector, sample, sample_rate): sample = torchaudio.functional.resample(sample, sample_rate, 22050) mel = wav_to_mel(sample) return projector.get_speech_projection(mel).squeeze(0) # Getting rid of the batch dimension means it's just [hidden_dim] def load_w2v(self): return Wav2Vec2ForCTC.from_pretrained("jbetker/wav2vec2-large-robust-ft-libritts-voxpopuli") def intelligibility_loss(self, w2v, sample, real_sample, sample_rate, real_text): """ Measures the differences between CTC losses using wav2vec2 against the real sample and the generated sample. """ text_codes = torch.tensor(text_to_sequence(real_text), device=sample.device) results = [] for s in [sample, real_sample]: s = torchaudio.functional.resample(s, sample_rate, 16000) norm_s = (s - s.mean()) / torch.sqrt(s.var() + 1e-7) norm_s = norm_s.squeeze(1) loss = w2v(input_values=norm_s, labels=text_codes).loss results.append(loss) gen_loss, real_loss = results return gen_loss - real_loss def compute_frechet_distance(self, proj1, proj2): # I really REALLY FUCKING HATE that this is going to numpy. Why does "pytorch_fid" operate in numpy land. WHY? proj1 = proj1.cpu().numpy() proj2 = proj2.cpu().numpy() mu1 = np.mean(proj1, axis=0) mu2 = np.mean(proj2, axis=0) sigma1 = np.cov(proj1, rowvar=False) sigma2 = np.cov(proj2, rowvar=False) return torch.tensor(calculate_frechet_distance(mu1, sigma1, mu2, sigma2)) def perform_eval(self): save_path = osp.join(self.env['base_path'], "../", "audio_eval", str(self.env["step"])) os.makedirs(save_path, exist_ok=True) projector = self.load_projector().to(self.env['device']) projector.eval() w2v = self.load_w2v().to(self.env['device']) w2v.eval() for k, mod in self.local_modules.items(): self.local_modules[k] = mod.to(self.env['device']) # Attempt to fix the random state as much as possible. RNG state will be restored before returning. rng_state = torch.get_rng_state() torch.manual_seed(5) self.model.eval() with torch.no_grad(): gen_projections = [] real_projections = [] intelligibility_losses = [] for i in tqdm(list(range(0, len(self.data), self.skip))): path, text, codes = self.data[(i + self.env['rank']) % len(self.data)] audio = load_audio(path, 22050).to(self.dev) codes = codes.to(self.dev) sample, ref, gen_mel, ref_mel, sample_rate = self.diffusion_fn(audio, codes, text) gen_projections.append(self.project(projector, sample, sample_rate).cpu()) # Store on CPU to avoid wasting GPU memory. real_projections.append(self.project(projector, ref, sample_rate).cpu()) intelligibility_losses.append(self.intelligibility_loss(w2v, sample, ref, sample_rate, text)) torchvision.utils.save_image((gen_mel.unsqueeze(1) + 1) / 2, os.path.join(save_path, f'{self.env["rank"]}_{i}_mel.png')) torchvision.utils.save_image((ref_mel.unsqueeze(1) + 1) / 2, os.path.join(save_path, f'{self.env["rank"]}_{i}_mel_target.png')) torchaudio.save(os.path.join(save_path, f"{self.env['rank']}_{i}_gen.wav"), sample.squeeze(0).cpu(), sample_rate) torchaudio.save(os.path.join(save_path, f"{self.env['rank']}_{i}_real.wav"), ref.squeeze(0).cpu(), sample_rate) gen_projections = torch.stack(gen_projections, dim=0) real_projections = torch.stack(real_projections, dim=0) intelligibility_loss = torch.stack(intelligibility_losses, dim=0).mean() frechet_distance = torch.tensor(self.compute_frechet_distance(gen_projections, real_projections), device=self.env['device']) if distributed.is_initialized() and distributed.get_world_size() > 1: distributed.all_reduce(frechet_distance) frechet_distance = frechet_distance / distributed.get_world_size() distributed.all_reduce(intelligibility_loss) intelligibility_loss = intelligibility_loss / distributed.get_world_size() self.model.train() torch.set_rng_state(rng_state) # Put modules used for evaluation back into CPU memory. for k, mod in self.local_modules.items(): self.local_modules[k] = mod.cpu() return {"frechet_distance": frechet_distance, "intelligibility_loss": intelligibility_loss} """ if __name__ == '__main__': from utils.util import load_model_from_config diffusion = load_model_from_config('X:\\dlas\\experiments\\train_diffusion_tts7_dvae_thin_with_text.yml', 'generator', also_load_savepoint=False, load_path='X:\\dlas\\experiments\\train_diffusion_tts7_dvae_thin_with_text\\models\\56500_generator_ema.pth').cuda() opt_eval = {'eval_tsv': 'Y:\\libritts\\test-clean\\transcribed-brief-w2v.tsv', 'diffusion_steps': 100, 'conditioning_free': False, 'conditioning_free_k': 1, 'diffusion_schedule': 'linear', 'diffusion_type': 'vocoder'} env = {'rank': 0, 'base_path': 'D:\\tmp\\test_eval', 'step': 2, 'device': 'cuda', 'opt': {}} eval = AudioDiffusionFid(diffusion, opt_eval, env) print(eval.perform_eval()) """ if __name__ == '__main__': diffusion = load_model_from_config('X:\\dlas\\experiments\\train_tts_diffusion_tfd12_ar_inputs.yml', 'generator', also_load_savepoint=False, load_path='X:\\dlas\\experiments\\train_tts_diffusion_tfd12_ar_inputs_pretrain\\models\\4500_generator.pth').cuda() opt_eval = {'eval_tsv': 'Y:\\libritts\\test-clean\\transcribed-brief-w2v.tsv', 'diffusion_steps': 50, 'conditioning_free': False, 'conditioning_free_k': 1, 'diffusion_schedule': 'linear', 'diffusion_type': 'tfd_ar'} env = {'rank': 0, 'base_path': 'D:\\tmp\\test_eval', 'step': 101, 'device': 'cuda', 'opt': {}} eval = AudioDiffusionFid(diffusion, opt_eval, env) print(eval.perform_eval())