muse / train /encode_audio.py
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import time
import json
import torch
import multiprocessing
from tqdm import tqdm
import concurrent.futures
from model import PromptCondAudioDiffusion
from diffusers import DDIMScheduler, DDPMScheduler
import torchaudio
import librosa
import os
import math
import numpy as np
from tools.get_melvaehifigan48k import build_pretrained_models
import tools.torch_tools as torch_tools
from safetensors.torch import load_file
import subprocess
def get_free_gpu() -> int:
"""Return the GPU ID with the least memory usage"""
cmd = "nvidia-smi --query-gpu=index,memory.free --format=csv,noheader,nounits"
result = subprocess.check_output(cmd.split()).decode().strip().split("\n")
free_list = []
for line in result:
idx, free_mem = line.split(",")
free_list.append((int(idx), int(free_mem))) # (GPU id, free memory MiB)
# Sort by remaining memory
free_list.sort(key=lambda x: x[1], reverse=True)
return free_list[0][0]
SAVE_DIR = "xxx/suno_mucodec_en"
DATA_PATH = "xxx/paths_en.jsonl"
DEVICE = f"cuda:{get_free_gpu()}"
print(f"Use {DEVICE}")
class MuCodec:
def __init__(self, \
model_path, \
layer_num, \
load_main_model=True):
self.layer_num = layer_num - 1
self.sample_rate = 48000
self.device = DEVICE
self.MAX_DURATION = 360
if load_main_model:
audio_ldm_path = os.path.dirname(os.path.abspath(__file__)) + "/tools/audioldm_48k.pth"
self.vae, self.stft = build_pretrained_models(audio_ldm_path)
self.vae, self.stft = self.vae.eval().to(DEVICE), self.stft.eval().to(DEVICE)
main_config = {
"num_channels":32,
"unet_model_name":None,
"unet_model_config_path":os.path.dirname(os.path.abspath(__file__)) + "/configs/models/transformer2D.json",
"snr_gamma":None,
}
self.model = PromptCondAudioDiffusion(**main_config)
if model_path.endswith('.safetensors'):
main_weights = load_file(model_path)
else:
main_weights = torch.load(model_path, map_location='cpu')
self.model.load_state_dict(main_weights, strict=False)
self.model = self.model.to(DEVICE)
print ("Successfully loaded checkpoint from:", model_path)
else:
main_config = {
"num_channels":32,
"unet_model_name":None,
"unet_model_config_path":None,
"snr_gamma":None,
}
self.model = PromptCondAudioDiffusion(**main_config).to(DEVICE)
main_weights = torch.load(model_path, map_location='cpu')
self.model.load_state_dict(main_weights, strict=False)
self.model = self.model.to(DEVICE)
print ("Successfully loaded checkpoint from:", model_path)
self.model.eval()
self.model.init_device_dtype(torch.device(DEVICE), torch.float32)
print("scaling factor: ", self.model.normfeat.std)
def file2code(self, fname):
orig_samples, fs = torchaudio.load(fname)
if(fs!=self.sample_rate):
orig_samples = torchaudio.functional.resample(orig_samples, fs, self.sample_rate)
fs = self.sample_rate
if orig_samples.shape[0] == 1:
orig_samples = torch.cat([orig_samples, orig_samples], 0)
return self.sound2code(orig_samples)
@torch.no_grad()
@torch.autocast(device_type="cuda", dtype=torch.float32)
def sound2code(self, orig_samples, batch_size=3):
if(orig_samples.ndim == 2):
audios = orig_samples.unsqueeze(0).to(self.device)
elif(orig_samples.ndim == 3):
audios = orig_samples.to(self.device)
else:
assert orig_samples.ndim in (2,3), orig_samples.shape
audios = self.preprocess_audio(audios)
audios = audios.squeeze(0)
orig_length = audios.shape[-1]
min_samples = int(40.96 * self.sample_rate)
output_len = int(orig_length / float(self.sample_rate) * 25) + 1
print("output_len: ", output_len)
while(audios.shape[-1] < min_samples + 480):
audios = torch.cat([audios, audios], -1)
int_max_len=audios.shape[-1]//min_samples+1
# print("int_max_len: ", int_max_len)
audios = torch.cat([audios, audios], -1)
# print("audios:",audios.shape)
audios=audios[:,:int(int_max_len*(min_samples+480))]
codes_list=[]
audio_input = audios.reshape(2, -1, min_samples+480).permute(1, 0, 2).reshape(-1, 2, min_samples+480)
for audio_inx in range(0, audio_input.shape[0], batch_size):
# import pdb; pdb.set_trace()
codes, _, spk_embeds = self.model.fetch_codes_batch((audio_input[audio_inx:audio_inx+batch_size]), additional_feats=[],layer=self.layer_num)
codes_list.append(torch.cat(codes, 1))
# print("codes_list",codes_list[0].shape)
codes = torch.cat(codes_list, 0).permute(1,0,2).reshape(1, -1)[None] # B 3 T -> 3 B T
codes=codes[:,:,:output_len]
return codes
@torch.no_grad()
def code2sound(self, codes, prompt=None, duration=40.96, guidance_scale=1.5, num_steps=20, disable_progress=False):
codes = codes.to(self.device)
first_latent = torch.randn(codes.shape[0], 32, 512, 32).to(self.device)
first_latent_length = 0
first_latent_codes_length = 0
if(isinstance(prompt, torch.Tensor)):
prompt = prompt.to(self.device)
if(prompt.ndim == 3):
assert prompt.shape[0] == 1, prompt.shape
prompt = prompt[0]
elif(prompt.ndim == 1):
prompt = prompt.unsqueeze(0).repeat(2,1)
elif(prompt.ndim == 2):
if(prompt.shape[0] == 1):
prompt = prompt.repeat(2,1)
if(prompt.shape[-1] < int(30.76 * self.sample_rate)):
prompt = prompt[:,:int(10.24*self.sample_rate)] # limit max length to 10.24
else:
prompt = prompt[:,int(20.48*self.sample_rate):int(30.72*self.sample_rate)] # limit max length to 10.24
true_mel , _, _ = torch_tools.wav_to_fbank2(prompt, -1, fn_STFT=self.stft) # maximum 10.24s
true_mel = true_mel.unsqueeze(1).to(self.device)
true_latent = torch.cat([self.vae.get_first_stage_encoding(self.vae.encode_first_stage(true_mel[[m]])) for m in range(true_mel.shape[0])],0)
true_latent = true_latent.reshape(true_latent.shape[0]//2, -1, true_latent.shape[2], true_latent.shape[3]).detach()
first_latent[:,:,0:true_latent.shape[2],:] = true_latent
first_latent_length = true_latent.shape[2]
first_latent_codes = self.sound2code(prompt)[:,:,0:first_latent_length*2] # B 4 T
first_latent_codes_length = first_latent_codes.shape[-1]
codes = torch.cat([first_latent_codes, codes], -1)
min_samples = 1024
hop_samples = min_samples // 4 * 3
ovlp_samples = min_samples - hop_samples
hop_frames = hop_samples // 2
ovlp_frames = ovlp_samples // 2
codes_len= codes.shape[-1]
target_len = int((codes_len - first_latent_codes_length) / 100 * 4 * self.sample_rate)
if(codes_len < min_samples):
while(codes.shape[-1] < min_samples):
codes = torch.cat([codes, codes], -1)
codes = codes[:,:,0:min_samples]
codes_len = codes.shape[-1]
if((codes_len - ovlp_frames) % hop_samples > 0):
len_codes=math.ceil((codes_len - ovlp_samples) / float(hop_samples)) * hop_samples + ovlp_samples
while(codes.shape[-1] < len_codes):
codes = torch.cat([codes, codes], -1)
codes = codes[:,:,0:len_codes]
latent_length = 512
latent_list = []
spk_embeds = torch.zeros([1, 32, 1, 32], device=codes.device)
with torch.autocast(device_type="cuda", dtype=torch.float16):
for sinx in range(0, codes.shape[-1]-hop_samples, hop_samples):
codes_input=[]
codes_input.append(codes[:,:,sinx:sinx+min_samples])
if(sinx == 0):
incontext_length = first_latent_length
latents = self.model.inference_codes(codes_input, spk_embeds, first_latent, latent_length, incontext_length, additional_feats=[], guidance_scale=1.5, num_steps = num_steps, disable_progress=disable_progress, scenario='other_seg')
latent_list.append(latents)
else:
true_latent = latent_list[-1][:,:,-ovlp_frames:,:]
len_add_to_512 = 512 - true_latent.shape[-2]
incontext_length = true_latent.shape[-2]
true_latent = torch.cat([true_latent, torch.randn(true_latent.shape[0], true_latent.shape[1], len_add_to_512, true_latent.shape[-1]).to(self.device)], -2)
latents = self.model.inference_codes(codes_input, spk_embeds, true_latent, latent_length, incontext_length, additional_feats=[], guidance_scale=1.5, num_steps = num_steps, disable_progress=disable_progress, scenario='other_seg')
latent_list.append(latents)
latent_list = [l.float() for l in latent_list]
latent_list[0] = latent_list[0][:,:,first_latent_length:,:]
min_samples = int(duration * self.sample_rate)
hop_samples = min_samples // 4 * 3
ovlp_samples = min_samples - hop_samples
with torch.no_grad():
output = None
for i in range(len(latent_list)):
latent = latent_list[i]
bsz , ch, t, f = latent.shape
latent = latent.reshape(bsz*2, ch//2, t, f)
mel = self.vae.decode_first_stage(latent)
cur_output = self.vae.decode_to_waveform(mel)
cur_output = torch.from_numpy(cur_output)[:, 0:min_samples]
if output is None:
output = cur_output
else:
ov_win = torch.from_numpy(np.linspace(0, 1, ovlp_samples)[None, :])
ov_win = torch.cat([ov_win, 1 - ov_win], -1)
output[:, -ovlp_samples:] = output[:, -ovlp_samples:] * ov_win[:, -ovlp_samples:] + cur_output[:, 0:ovlp_samples] * ov_win[:, 0:ovlp_samples]
output = torch.cat([output, cur_output[:, ovlp_samples:]], -1)
output = output[:, 0:target_len]
return output
@torch.no_grad()
def preprocess_audio(self, input_audios, threshold=0.8):
assert len(input_audios.shape) == 3, input_audios.shape
nchan = input_audios.shape[1]
input_audios = input_audios.reshape(input_audios.shape[0], -1)
norm_value = torch.ones_like(input_audios[:,0])
max_volume = input_audios.abs().max(dim=-1)[0]
norm_value[max_volume>threshold] = max_volume[max_volume>threshold] / threshold
return input_audios.reshape(input_audios.shape[0], nchan, -1)/norm_value.unsqueeze(-1).unsqueeze(-1)
@torch.no_grad()
def sound2sound(self, sound, prompt=None, min_duration=40.96, steps=50, disable_progress=False):
start_time = time.time()
codes = self.sound2code(sound)
mid_time = time.time()
elapsed_1 = mid_time - start_time
print(f"sound2code: {elapsed_1:.3f}s")
wave = self.code2sound(codes, prompt, duration=min_duration, guidance_scale=1.5, num_steps=steps, disable_progress=disable_progress)
end_time = time.time()
elapsed_2 = end_time - mid_time
print(f"code2sound: step-{steps}, {elapsed_2:.3f}s")
return wave
def load_jsonl(path:str) -> list[dict]:
dataset = []
with open(path, 'r') as file:
for line in tqdm(file, desc=f"Loading {path}"):
dataset.append(json.loads(line))
return dataset
ckpt_path = os.path.join(os.path.dirname(os.path.abspath(__file__)), "ckpt/mucodec.pt")
mucodec = MuCodec(model_path=ckpt_path,layer_num=7,load_main_model=True)
def pure_name(path:str):
"""Get the original name of a file path (without extension)"""
basename = os.path.basename(path)
dot_pos = basename.rfind('.')
if dot_pos == -1:
return basename
return basename[:dot_pos]
def music_encode(path:str):
# Process to get path after vocal separation
name = pure_name(path)
# path = path.replace("enhanced", "separated").replace(".mp3", "/vocals.wav")
sound, fs = torchaudio.load(path)
if(fs!=48000):
sound = torchaudio.functional.resample(sound, fs, 48000)
if(sound.shape[0]==1):
sound = torch.cat([sound, sound],0)
code = mucodec.sound2code(sound)
# save_path = f"xxx/seperate_codes/{name}.pt"
save_path = os.path.join(SAVE_DIR, f"{name}.pt")
torch.save(code, save_path)
return save_path
# ls xxx/seperate_codes/ -l | grep -c '^-'
def remove_finish(dataset:list[dict]) -> list[dict]:
"""Remove already encoded items from dataset"""
names = []
for path in os.listdir(SAVE_DIR):
names.append(pure_name(path))
new_dataset = []
dup_num = 0
for line in tqdm(dataset):
name = pure_name(line['path'])
if name not in names:
new_dataset.append(line)
else:
dup_num += 1
print(f"Remove dup num: {dup_num}")
return new_dataset
if __name__=="__main__":
multiprocessing.set_start_method('spawn', force=True)
dataset = load_jsonl(DATA_PATH)
dataset = remove_finish(dataset)
with concurrent.futures.ProcessPoolExecutor(max_workers=4) as executor:
# Submit to task pool
futures = [executor.submit(music_encode, line['path']) for line in dataset]
# Progress bar display
for future in tqdm(concurrent.futures.as_completed(futures),
total=len(dataset),
desc="Encoding audio count",
unit="files",
ncols=100,
bar_format="{l_bar}{bar}| {n_fmt}/{total_fmt} [{elapsed}<{remaining}]"):
try:
result = future.result() # Explicitly get result (must call even if not needed)
except Exception as e:
print(f"Task failed: {e}") # Catch exception to avoid interrupting progress bar