import argparse, pickle import logging import os, random import numpy as np import torch import torchaudio from data.tokenizer import ( AudioTokenizer, TextTokenizer, tokenize_audio, tokenize_text ) import argparse, time, tqdm # this script only works for the musicgen architecture def get_args(): parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter) parser.add_argument("--manifest_fn", type=str, default="path/to/eval_metadata_file") parser.add_argument("--audio_root", type=str, default="path/to/audio_folder") parser.add_argument("--exp_dir", type=str, default="path/to/model_folder") parser.add_argument("--seed", type=int, default=1) parser.add_argument("--codec_audio_sr", type=int, default=16000, help='the sample rate of audio that the codec is trained for') parser.add_argument("--codec_sr", type=int, default=50, help='the sample rate of the codec codes') parser.add_argument("--top_k", type=int, default=0, help="sampling param") parser.add_argument("--top_p", type=float, default=0.8, help="sampling param") parser.add_argument("--temperature", type=float, default=1.0, help="sampling param") parser.add_argument("--output_dir", type=str, default=None) parser.add_argument("--device", type=str, default="cuda") parser.add_argument("--signature", type=str, default=None, help="path to the encodec model") parser.add_argument("--crop_concat", type=int, default=0) parser.add_argument("--stop_repetition", type=int, default=-1, help="used for inference, when the number of consecutive repetition of a token is bigger than this, stop it") parser.add_argument("--kvcache", type=int, default=1, help='if true, use kv cache, which is 4-8x faster than without') parser.add_argument("--sample_batch_size", type=int, default=1, help="batch size for sampling, NOTE that it's not running inference for several samples, but duplicate one input sample batch_size times, and during inference, we only return the shortest generation") parser.add_argument("--silence_tokens", type=str, default="[1388,1898,131]", help="note that if you are not using the pretrained encodec 6f79c6a8, make sure you specified it yourself, rather than using the default") return parser.parse_args() @torch.no_grad() def inference_one_sample(model, model_args, phn2num, text_tokenizer, audio_tokenizer, audio_fn, target_text, device, decode_config, prompt_end_frame, target_generation_length, delay_pattern_increment, prefix_transcript=None, quiet=False, repeat_prompt=0, multi_trial=[]): # seq_len_thres = 500 # 10s, 26% of the data in seed tts # encode audio encoded_frames = tokenize_audio(audio_tokenizer, audio_fn, offset=0, num_frames=prompt_end_frame) # if sequence length is shorter than seq_len_thres, repeat the audio # if encoded_frames.shape[2] < seq_len_thres: # encoded_frames = torch.cat([encoded_frames, encoded_frames, encoded_frames], dim=2) # doubled = True single_encoded_frames = encoded_frames if isinstance(repeat_prompt, int) and repeat_prompt > 0: cur_repeat_prompt = repeat_prompt while cur_repeat_prompt > 0: encoded_frames = torch.cat([encoded_frames, single_encoded_frames], dim=2) cur_repeat_prompt -= 1 elif isinstance(repeat_prompt, str) and repeat_prompt.lower() == "max": repeat_prompt = 0 while encoded_frames.shape[2] + decode_config['codec_sr'] * target_generation_length + delay_pattern_increment + single_encoded_frames.shape[2] < model_args.audio_max_length * decode_config['codec_sr']: encoded_frames = torch.cat([encoded_frames, single_encoded_frames], dim=2) repeat_prompt += 1 if getattr(model_args, "y_sep_token", None) != None: encoded_frames = torch.cat([encoded_frames, torch.LongTensor([model_args.y_sep_token]*model_args.n_codebooks).unsqueeze(0).unsqueeze(2).to(encoded_frames.device)], dim=2) # print(encoded_frames.shape) original_audio = encoded_frames.transpose(2,1) # [1,T,K] assert original_audio.ndim==3 and original_audio.shape[0] == 1 and original_audio.shape[2] == model_args.n_codebooks, original_audio.shape # phonemize if isinstance(target_text, list): text_tokens = [phn2num[phn] for phn in target_text if phn in phn2num] else: text_tokens = [phn2num[phn] for phn in tokenize_text( text_tokenizer, text=target_text.strip() ) if phn in phn2num ] if getattr(model_args, "x_sep_token", None) != None: assert prefix_transcript != None, "prefix_transcript must be provided if x_sep_token is not None" if prefix_transcript is not None: if isinstance(prefix_transcript, list): prefix_tokens = [phn2num[phn] for phn in prefix_transcript if phn in phn2num] else: prefix_tokens = [phn2num[phn] for phn in tokenize_text( text_tokenizer, text=prefix_transcript.strip() ) if phn in phn2num ] # if doubled: # prefix_tokens = prefix_tokens + prefix_tokens + prefix_tokens single_prefix_tokens = prefix_tokens while repeat_prompt > 0: prefix_tokens = prefix_tokens + single_prefix_tokens repeat_prompt -= 1 if getattr(model_args, "x_sep_token", None) != None: text_tokens = prefix_tokens + [getattr(model_args, "x_sep_token", None)] + text_tokens else: text_tokens = prefix_tokens + text_tokens if getattr(model_args, "add_eos_to_text", 0) != 0: text_tokens.append(model_args.add_eos_to_text) if getattr(model_args, "add_bos_to_text", 0) != 0: text_tokens = [model_args.add_bos_to_text] + text_tokens text_tokens = torch.LongTensor(text_tokens).unsqueeze(0) text_tokens_lens = torch.LongTensor([text_tokens.shape[-1]]) if not quiet: logging.info(f"original audio length: {original_audio.shape[1]} codec frames, which is {original_audio.shape[1]/decode_config['codec_sr']:.2f} sec.") if getattr(model_args, "parallel_pattern", 0) != 0: tgt_y_lens = torch.LongTensor([int(original_audio.shape[1] + decode_config['codec_sr'] * target_generation_length + 2)]) # parallel pattern, therefore only add the empty_token (i.e. the sos token) and eos (i.e. 2 more tokens). Note that the delayed pattern between, both sos and eos is counted (sos is counted in the n_codebooks, eos is counted in the 1) else: tgt_y_lens = torch.LongTensor([int(original_audio.shape[1] + decode_config['codec_sr'] * target_generation_length + delay_pattern_increment)]) # delay pattern increment has accounted for the added eos # forward assert decode_config['sample_batch_size'] <= 1 stime = time.time() assert multi_trial == [] if not quiet: logging.info(f"running inference with batch size 1") concat_frames, gen_frames = model.inference_tts( text_tokens.to(device), text_tokens_lens.to(device), original_audio[...,:model_args.n_codebooks].to(device), # [1,T,8] tgt_y_lens = tgt_y_lens.to(device), top_k=decode_config['top_k'], top_p=decode_config['top_p'], min_p=decode_config['min_p'], temperature=decode_config['temperature'], stop_repetition=decode_config['stop_repetition'], kvcache=decode_config['kvcache'], silence_tokens=eval(decode_config['silence_tokens']) if type(decode_config['silence_tokens'])==str else decode_config['silence_tokens'] ) # output is [1,K,T] if not quiet: logging.info(f"inference on one sample take: {time.time() - stime:.4f} sec.") logging.info(f"generated encoded_frames.shape: {gen_frames.shape}, which is {gen_frames.shape[-1]/decode_config['codec_sr']} sec.") # for timestamp, codes in enumerate(gen_frames[0].transpose(1,0)): # logging.info(f"{timestamp}: {codes.tolist()}") # decode (both original and generated) # concat_sample = audio_tokenizer.decode( # [(concat_frames, None)] # [1,T,8] -> [1,8,T] # ) if getattr(model_args, "y_sep_token", None) != None: concat_frames = torch.cat([concat_frames[:, :, :original_audio.shape[1]-1], concat_frames[:, :, original_audio.shape[1]:]], dim=2) concat_sample = audio_tokenizer.decode( concat_frames # [1,8,T] ) gen_sample = audio_tokenizer.decode( gen_frames ) #Empty cuda cache between runs if torch.cuda.is_available(): torch.cuda.empty_cache() # return return concat_sample, gen_sample