From 287debd1d38475f82377407d889ed50e17e2bb42 Mon Sep 17 00:00:00 2001 From: James Betker Date: Fri, 1 Apr 2022 11:55:07 -0600 Subject: [PATCH] port do_tts to use the API --- api.py | 31 ++++++-- do_tts.py | 206 +++--------------------------------------------------- 2 files changed, 36 insertions(+), 201 deletions(-) diff --git a/api.py b/api.py index 799bd16..be07783 100644 --- a/api.py +++ b/api.py @@ -151,10 +151,10 @@ class TextToSpeech: def tts(self, text, voice_samples, k=1, # autoregressive generation parameters follow - num_autoregressive_samples=512, temperature=.9, length_penalty=1, repetition_penalty=1.0, top_k=50, top_p=.95, + num_autoregressive_samples=512, temperature=.5, length_penalty=2, repetition_penalty=2.0, top_p=.5, typical_sampling=False, typical_mass=.9, # diffusion generation parameters follow - diffusion_iterations=100, cond_free=True, cond_free_k=1, diffusion_temperature=1,): + diffusion_iterations=100, cond_free=True, cond_free_k=2, diffusion_temperature=.7,): text = torch.IntTensor(self.tokenizer.encode(text)).unsqueeze(0).cuda() text = F.pad(text, (0, 1)) # This may not be necessary. @@ -181,7 +181,6 @@ class TextToSpeech: for b in tqdm(range(num_batches)): codes = self.autoregressive.inference_speech(conds, text, do_sample=True, - top_k=top_k, top_p=top_p, temperature=temperature, num_return_sequences=self.autoregressive_batch_size, @@ -220,4 +219,28 @@ class TextToSpeech: if len(wav_candidates) > 1: return wav_candidates - return wav_candidates[0] \ No newline at end of file + return wav_candidates[0] + + def refine_for_intellibility(self, wav_candidates, corresponding_codes, output_path): + """ + Further refine the remaining candidates using a ASR model to pick out the ones that are the most understandable. + TODO: finish this function + :param wav_candidates: + :return: + """ + transcriber = ocotillo.Transcriber(on_cuda=True) + transcriptions = transcriber.transcribe_batch(torch.cat(wav_candidates, dim=0).squeeze(1), 24000) + best = 99999999 + for i, transcription in enumerate(transcriptions): + dist = lev_distance(transcription, args.text.lower()) + if dist < best: + best = dist + best_codes = corresponding_codes[i].unsqueeze(0) + best_wav = wav_candidates[i] + del transcriber + torchaudio.save(os.path.join(output_path, f'{voice}_poor.wav'), best_wav.squeeze(0).cpu(), 24000) + + # Perform diffusion again with the high-quality diffuser. + mel = do_spectrogram_diffusion(diffusion, final_diffuser, best_codes, cond_diffusion, mean=False) + wav = vocoder.inference(mel) + torchaudio.save(os.path.join(args.output_path, f'{voice}.wav'), wav.squeeze(0).cpu(), 24000) \ No newline at end of file diff --git a/do_tts.py b/do_tts.py index aa2cbdc..af5c780 100644 --- a/do_tts.py +++ b/do_tts.py @@ -1,123 +1,13 @@ import argparse import os -import random -from urllib import request import torch import torch.nn.functional as F import torchaudio -import progressbar -import ocotillo - -from models.diffusion_decoder import DiffusionTts -from models.autoregressive import UnifiedVoice -from tqdm import tqdm - -from models.arch_util import TorchMelSpectrogram -from models.text_voice_clip import VoiceCLIP -from models.vocoder import UnivNetGenerator -from utils.audio import load_audio, wav_to_univnet_mel, denormalize_tacotron_mel -from utils.diffusion import SpacedDiffusion, space_timesteps, get_named_beta_schedule -from utils.tokenizer import VoiceBpeTokenizer, lev_distance - -pbar = None -def download_models(): - MODELS = { - 'clip.pth': 'https://huggingface.co/jbetker/tortoise-tts-clip/resolve/main/pytorch-model.bin', - 'diffusion.pth': 'https://huggingface.co/jbetker/tortoise-tts-diffusion-v1/resolve/main/pytorch-model.bin', - 'autoregressive.pth': 'https://huggingface.co/jbetker/tortoise-tts-autoregressive/resolve/main/pytorch-model.bin' - } - os.makedirs('.models', exist_ok=True) - def show_progress(block_num, block_size, total_size): - global pbar - if pbar is None: - pbar = progressbar.ProgressBar(maxval=total_size) - pbar.start() - - downloaded = block_num * block_size - if downloaded < total_size: - pbar.update(downloaded) - else: - pbar.finish() - pbar = None - for model_name, url in MODELS.items(): - if os.path.exists(f'.models/{model_name}'): - continue - print(f'Downloading {model_name} from {url}...') - request.urlretrieve(url, f'.models/{model_name}', show_progress) - print('Done.') - - -def load_discrete_vocoder_diffuser(trained_diffusion_steps=4000, desired_diffusion_steps=200, cond_free=True): - """ - Helper function to load a GaussianDiffusion instance configured for use as a vocoder. - """ - return SpacedDiffusion(use_timesteps=space_timesteps(trained_diffusion_steps, [desired_diffusion_steps]), model_mean_type='epsilon', - model_var_type='learned_range', loss_type='mse', betas=get_named_beta_schedule('linear', trained_diffusion_steps), - conditioning_free=cond_free, conditioning_free_k=1) - - -def load_conditioning(path, sample_rate=22050, cond_length=132300): - rel_clip = load_audio(path, sample_rate) - gap = rel_clip.shape[-1] - cond_length - if gap < 0: - rel_clip = F.pad(rel_clip, pad=(0, abs(gap))) - elif gap > 0: - rand_start = random.randint(0, gap) - rel_clip = rel_clip[:, rand_start:rand_start + cond_length] - mel_clip = TorchMelSpectrogram()(rel_clip.unsqueeze(0)).squeeze(0) - return mel_clip.unsqueeze(0).cuda(), rel_clip.unsqueeze(0).cuda() - - -def fix_autoregressive_output(codes, stop_token): - """ - This function performs some padding on coded audio that fixes a mismatch issue between what the diffusion model was - trained on and what the autoregressive code generator creates (which has no padding or end). - This is highly specific to the DVAE being used, so this particular coding will not necessarily work if used with - a different DVAE. This can be inferred by feeding a audio clip padded with lots of zeros on the end through the DVAE - and copying out the last few codes. - - Failing to do this padding will produce speech with a harsh end that sounds like "BLAH" or similar. - """ - # Strip off the autoregressive stop token and add padding. - stop_token_indices = (codes == stop_token).nonzero() - if len(stop_token_indices) == 0: - print("No stop tokens found, enjoy that output of yours!") - return - else: - codes[stop_token_indices] = 83 - stm = stop_token_indices.min().item() - codes[stm:] = 83 - if stm - 3 < codes.shape[0]: - codes[-3] = 45 - codes[-2] = 45 - codes[-1] = 248 - - return codes - - -def do_spectrogram_diffusion(diffusion_model, diffuser, mel_codes, conditioning_input, mean=False): - """ - Uses the specified diffusion model and DVAE model to convert the provided MEL & conditioning inputs into an audio clip. - """ - with torch.no_grad(): - cond_mel = wav_to_univnet_mel(conditioning_input.squeeze(1), do_normalization=False) - # Pad MEL to multiples of 32 - msl = mel_codes.shape[-1] - dsl = 32 - gap = dsl - (msl % dsl) - if gap > 0: - mel = torch.nn.functional.pad(mel_codes, (0, gap)) - - output_shape = (mel.shape[0], 100, mel.shape[-1]*4) - precomputed_embeddings = diffusion_model.timestep_independent(mel_codes, cond_mel) - if mean: - mel = diffuser.p_sample_loop(diffusion_model, output_shape, noise=torch.zeros(output_shape, device=mel_codes.device), - model_kwargs={'precomputed_aligned_embeddings': precomputed_embeddings}) - else: - mel = diffuser.p_sample_loop(diffusion_model, output_shape, model_kwargs={'precomputed_aligned_embeddings': precomputed_embeddings}) - return denormalize_tacotron_mel(mel)[:,:,:msl*4] +from api import TextToSpeech, load_conditioning +from utils.audio import load_audio +from utils.tokenizer import VoiceBpeTokenizer if __name__ == '__main__': # These are voices drawn randomly from the training set. You are free to substitute your own voices in, but testing @@ -139,101 +29,23 @@ if __name__ == '__main__': parser.add_argument('-text', type=str, help='Text to speak.', default="I am a language model that has learned to speak.") parser.add_argument('-voice', type=str, help='Use a preset conditioning voice (defined above). Overrides cond_path.', default='dotrice,harris,lescault,otto,atkins,grace,kennard,mol') parser.add_argument('-num_samples', type=int, help='How many total outputs the autoregressive transformer should produce.', default=512) - parser.add_argument('-num_batches', type=int, help='How many batches those samples should be produced over.', default=16) + parser.add_argument('-batch_size', type=int, help='How many samples to process at once in the autoregressive model.', default=16) parser.add_argument('-num_diffusion_samples', type=int, help='Number of outputs that progress to the diffusion stage.', default=16) parser.add_argument('-output_path', type=str, help='Where to store outputs.', default='results/') args = parser.parse_args() - os.makedirs(args.output_path, exist_ok=True) - download_models() + + tts = TextToSpeech(autoregressive_batch_size=args.batch_size) for voice in args.voice.split(','): - print("Loading data..") tokenizer = VoiceBpeTokenizer() text = torch.IntTensor(tokenizer.encode(args.text)).unsqueeze(0).cuda() text = F.pad(text, (0,1)) # This may not be necessary. cond_paths = preselected_cond_voices[voice] conds = [] for cond_path in cond_paths: - c, cond_wav = load_conditioning(cond_path) + c = load_audio(cond_path, 22050) conds.append(c) - conds = torch.stack(conds, dim=1) - cond_diffusion = cond_wav[:, :88200] # The diffusion model expects <= 88200 conditioning samples. - - print("Loading GPT TTS..") - autoregressive = UnifiedVoice(max_mel_tokens=300, max_text_tokens=200, max_conditioning_inputs=2, layers=30, model_dim=1024, - heads=16, number_text_tokens=256, start_text_token=255, checkpointing=False, train_solo_embeddings=False, - average_conditioning_embeddings=True).cuda().eval() - autoregressive.load_state_dict(torch.load('.models/autoregressive.pth')) - stop_mel_token = autoregressive.stop_mel_token - - with torch.no_grad(): - print("Performing autoregressive inference..") - samples = [] - for b in tqdm(range(args.num_batches)): - codes = autoregressive.inference_speech(conds, text, num_beams=1, repetition_penalty=1.0, do_sample=True, top_k=50, top_p=.95, - temperature=.9, num_return_sequences=args.num_samples//args.num_batches, length_penalty=1) - padding_needed = 250 - codes.shape[1] - codes = F.pad(codes, (0, padding_needed), value=stop_mel_token) - samples.append(codes) - del autoregressive - - print("Loading CLIP..") - clip = VoiceCLIP(dim_text=512, dim_speech=512, dim_latent=512, num_text_tokens=256, text_enc_depth=12, text_seq_len=350, text_heads=8, - num_speech_tokens=8192, speech_enc_depth=12, speech_heads=8, speech_seq_len=430, use_xformers=True).cuda().eval() - clip.load_state_dict(torch.load('.models/clip.pth')) - print("Performing CLIP filtering..") - clip_results = [] - for batch in samples: - for i in range(batch.shape[0]): - batch[i] = fix_autoregressive_output(batch[i], stop_mel_token) - clip_results.append(clip(text.repeat(batch.shape[0], 1), batch, return_loss=False)) - clip_results = torch.cat(clip_results, dim=0) - samples = torch.cat(samples, dim=0) - best_results = samples[torch.topk(clip_results, k=args.num_diffusion_samples).indices] - - # Delete the autoregressive and clip models to free up GPU memory - del samples, clip - - print("Loading Diffusion Model..") - diffusion = DiffusionTts(model_channels=512, in_channels=100, out_channels=200, in_latent_channels=1024, - channel_mult=[1, 2, 3, 4], num_res_blocks=[3, 3, 3, 3], token_conditioning_resolutions=[1,4,8], - dropout=0, attention_resolutions=[4,8], num_heads=8, kernel_size=3, scale_factor=2, - time_embed_dim_multiplier=4, unconditioned_percentage=0, conditioning_dim_factor=2, - conditioning_expansion=1) - diffusion.load_state_dict(torch.load('.models/diffusion.pth')) - diffusion = diffusion.cuda().eval() - print("Loading vocoder..") - vocoder = UnivNetGenerator() - vocoder.load_state_dict(torch.load('.models/vocoder.pth')['model_g']) - vocoder = vocoder.cuda() - vocoder.eval(inference=True) - initial_diffuser = load_discrete_vocoder_diffuser(desired_diffusion_steps=40, cond_free=False) - final_diffuser = load_discrete_vocoder_diffuser(desired_diffusion_steps=500) - - print("Performing vocoding..") - wav_candidates = [] - for b in range(best_results.shape[0]): - code = best_results[b].unsqueeze(0) - mel = do_spectrogram_diffusion(diffusion, initial_diffuser, code, cond_diffusion, mean=False) - wav = vocoder.inference(mel) - wav_candidates.append(wav.cpu()) - - # Further refine the remaining candidates using a ASR model to pick out the ones that are the most understandable. - transcriber = ocotillo.Transcriber(on_cuda=True) - transcriptions = transcriber.transcribe_batch(torch.cat(wav_candidates, dim=0).squeeze(1), 24000) - best = 99999999 - for i, transcription in enumerate(transcriptions): - dist = lev_distance(transcription, args.text.lower()) - if dist < best: - best = dist - best_codes = best_results[i].unsqueeze(0) - best_wav = wav_candidates[i] - del transcriber - torchaudio.save(os.path.join(args.output_path, f'{voice}_poor.wav'), best_wav.squeeze(0).cpu(), 24000) - - # Perform diffusion again with the high-quality diffuser. - mel = do_spectrogram_diffusion(diffusion, final_diffuser, best_codes, cond_diffusion, mean=False) - wav = vocoder.inference(mel) - torchaudio.save(os.path.join(args.output_path, f'{voice}.wav'), wav.squeeze(0).cpu(), 24000) + gen = tts.tts(args.text, conds, num_autoregressive_samples=args.num_samples) + torchaudio.save(os.path.join(args.output_path, f'{voice}.wav'), gen.squeeze(0).cpu(), 24000)