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import torch | |
import logging | |
import torch.nn as nn | |
from audioldm2.clap.open_clip import create_model | |
from audioldm2.clap.training.data import get_audio_features | |
import torchaudio | |
from transformers import RobertaTokenizer, AutoTokenizer, T5EncoderModel | |
import torch.nn.functional as F | |
from audioldm2.latent_diffusion.modules.audiomae.AudioMAE import Vanilla_AudioMAE | |
from audioldm2.latent_diffusion.modules.phoneme_encoder.encoder import TextEncoder | |
from transformers import AutoTokenizer, T5Config | |
from audioldm2.audiomae_gen.sequence_input import Sequence2AudioMAE | |
import numpy as np | |
""" | |
The model forward function can return three types of data: | |
1. tensor: used directly as conditioning signal | |
2. dict: where there is a main key as condition, there are also other key that you can use to pass loss function and itermediate result. etc. | |
3. list: the length is 2, in which the first element is tensor, the second element is attntion mask. | |
The output shape for the cross attention condition should be: | |
x,x_mask = [bs, seq_len, emb_dim], [bs, seq_len] | |
All the returned data, in which will be used as diffusion input, will need to be in float type | |
""" | |
class PhonemeEncoder(nn.Module): | |
def __init__(self, vocabs_size=41, pad_length=250, pad_token_id=None): | |
super().__init__() | |
""" | |
encoder = PhonemeEncoder(40) | |
data = torch.randint(0, 39, (2, 250)) | |
output = encoder(data) | |
import ipdb;ipdb.set_trace() | |
""" | |
assert pad_token_id is not None | |
self.device = None | |
self.PAD_LENGTH = int(pad_length) | |
self.pad_token_id = pad_token_id | |
self.pad_token_sequence = torch.tensor([self.pad_token_id] * self.PAD_LENGTH) | |
self.text_encoder = TextEncoder( | |
n_vocab=vocabs_size, | |
out_channels=192, | |
hidden_channels=192, | |
filter_channels=768, | |
n_heads=2, | |
n_layers=6, | |
kernel_size=3, | |
p_dropout=0.1, | |
) | |
self.learnable_positional_embedding = torch.nn.Parameter( | |
torch.zeros((1, 192, self.PAD_LENGTH)) | |
) # [batchsize, seqlen, padlen] | |
self.learnable_positional_embedding.requires_grad = True | |
# Required | |
def get_unconditional_condition(self, batchsize): | |
unconditional_tokens = self.pad_token_sequence.expand( | |
batchsize, self.PAD_LENGTH | |
) | |
return self(unconditional_tokens) # Need to return float type | |
# def get_unconditional_condition(self, batchsize): | |
# hidden_state = torch.zeros((batchsize, self.PAD_LENGTH, 192)).to(self.device) | |
# attention_mask = torch.ones((batchsize, self.PAD_LENGTH)).to(self.device) | |
# return [hidden_state, attention_mask] # Need to return float type | |
def _get_src_mask(self, phoneme): | |
src_mask = phoneme != self.pad_token_id | |
return src_mask | |
def _get_src_length(self, phoneme): | |
src_mask = self._get_src_mask(phoneme) | |
length = torch.sum(src_mask, dim=-1) | |
return length | |
# def make_empty_condition_unconditional(self, src_length, text_emb, attention_mask): | |
# # src_length: [bs] | |
# # text_emb: [bs, 192, pad_length] | |
# # attention_mask: [bs, pad_length] | |
# mask = src_length[..., None, None] > 1 | |
# text_emb = text_emb * mask | |
# attention_mask[src_length < 1] = attention_mask[src_length < 1] * 0.0 + 1.0 | |
# return text_emb, attention_mask | |
def forward(self, phoneme_idx): | |
if self.device is None: | |
self.device = self.learnable_positional_embedding.device | |
self.pad_token_sequence = self.pad_token_sequence.to(self.device) | |
phoneme_idx = phoneme_idx.to(self.device) | |
src_length = self._get_src_length(phoneme_idx) | |
text_emb, m, logs, text_emb_mask = self.text_encoder(phoneme_idx, src_length) | |
text_emb = text_emb + self.learnable_positional_embedding | |
# text_emb, text_emb_mask = self.make_empty_condition_unconditional(src_length, text_emb, text_emb_mask) | |
return [ | |
text_emb.permute(0, 2, 1), | |
text_emb_mask.squeeze(1), | |
] # [2, 250, 192], [2, 250] | |
class FlanT5HiddenState(nn.Module): | |
""" | |
llama = FlanT5HiddenState() | |
data = ["","this is not an empty sentence"] | |
encoder_hidden_states = llama(data) | |
import ipdb;ipdb.set_trace() | |
""" | |
def __init__( | |
self, text_encoder_name="google/flan-t5-large", freeze_text_encoder=True | |
): | |
super().__init__() | |
self.freeze_text_encoder = freeze_text_encoder | |
self.tokenizer = AutoTokenizer.from_pretrained(text_encoder_name) | |
self.model = T5EncoderModel(T5Config.from_pretrained(text_encoder_name)) | |
if freeze_text_encoder: | |
self.model.eval() | |
for p in self.model.parameters(): | |
p.requires_grad = False | |
else: | |
print("=> The text encoder is learnable") | |
self.empty_hidden_state_cfg = None | |
self.device = None | |
# Required | |
def get_unconditional_condition(self, batchsize): | |
param = next(self.model.parameters()) | |
if self.freeze_text_encoder: | |
assert param.requires_grad == False | |
# device = param.device | |
if self.empty_hidden_state_cfg is None: | |
self.empty_hidden_state_cfg, _ = self([""]) | |
hidden_state = torch.cat([self.empty_hidden_state_cfg] * batchsize).float() | |
attention_mask = ( | |
torch.ones((batchsize, hidden_state.size(1))) | |
.to(hidden_state.device) | |
.float() | |
) | |
return [hidden_state, attention_mask] # Need to return float type | |
def forward(self, batch): | |
param = next(self.model.parameters()) | |
if self.freeze_text_encoder: | |
assert param.requires_grad == False | |
if self.device is None: | |
self.device = param.device | |
# print("Manually change text") | |
# for i in range(len(batch)): | |
# batch[i] = "dog barking" | |
try: | |
return self.encode_text(batch) | |
except Exception as e: | |
print(e, batch) | |
logging.exception("An error occurred: %s", str(e)) | |
def encode_text(self, prompt): | |
device = self.model.device | |
batch = self.tokenizer( | |
prompt, | |
max_length=128, # self.tokenizer.model_max_length | |
padding=True, | |
truncation=True, | |
return_tensors="pt", | |
) | |
input_ids, attention_mask = batch.input_ids.to(device), batch.attention_mask.to( | |
device | |
) | |
# Get text encoding | |
if self.freeze_text_encoder: | |
with torch.no_grad(): | |
encoder_hidden_states = self.model( | |
input_ids=input_ids, attention_mask=attention_mask | |
)[0] | |
else: | |
encoder_hidden_states = self.model( | |
input_ids=input_ids, attention_mask=attention_mask | |
)[0] | |
return [ | |
encoder_hidden_states.detach(), | |
attention_mask.float(), | |
] # Attention mask == 1 means usable token | |
class SequenceGenAudioMAECond(Sequence2AudioMAE): | |
def __init__( | |
self, | |
cond_stage_config, | |
base_learning_rate, | |
sequence_gen_length, | |
sequence_input_key, | |
sequence_input_embed_dim, | |
batchsize, | |
always_output_audiomae_gt=False, | |
pretrained_path=None, | |
force_reload_pretrain_avoid_overwrite=False, | |
learnable=True, | |
use_warmup=True, | |
device=None, | |
use_gt_mae_output=None, # False: does not use AudioMAE GT, True: Use AudioMAE GT | |
use_gt_mae_prob=None, | |
): # The prob of using AudioMAE GT | |
if use_warmup: | |
use_warmup = False | |
super().__init__( | |
base_learning_rate=base_learning_rate, | |
cond_stage_config=cond_stage_config, | |
sequence_gen_length=sequence_gen_length, | |
sequence_input_key=sequence_input_key, | |
use_warmup=use_warmup, | |
sequence_input_embed_dim=sequence_input_embed_dim, | |
batchsize=batchsize, | |
) | |
assert use_gt_mae_output is not None and use_gt_mae_prob is not None | |
self.always_output_audiomae_gt = always_output_audiomae_gt | |
self.force_reload_pretrain_avoid_overwrite = ( | |
force_reload_pretrain_avoid_overwrite | |
) | |
self.pretrained_path = pretrained_path | |
self.device = device | |
if self.force_reload_pretrain_avoid_overwrite: | |
self.is_reload = False | |
else: | |
self.is_reload = True | |
self.load_pretrain_model() | |
self.use_gt_mae_output = use_gt_mae_output | |
self.use_gt_mae_prob = use_gt_mae_prob | |
self.learnable = learnable | |
if not learnable: | |
# Only optimize the GPT2 model | |
for p in self.model.parameters(): | |
p.requires_grad = False | |
self.eval() | |
def load_pretrain_model(self): | |
if self.pretrained_path is not None: | |
print("Reload SequenceGenAudioMAECond from %s" % self.pretrained_path) | |
state_dict = torch.load(self.pretrained_path)["state_dict"] | |
self.load_state_dict(state_dict) | |
# Required | |
def get_unconditional_condition(self, batchsize): | |
return_dict = self.cfg_uncond(batchsize) | |
return_dict["crossattn_audiomae_generated"] = [ | |
return_dict["crossattn_audiomae_pooled"][0], | |
torch.ones_like(return_dict["crossattn_audiomae_pooled"][1]).float(), | |
] | |
return return_dict | |
def forward(self, batch): | |
# The conditional module can return both tensor or dictionaries | |
# The returned tensor will be corresponding to the cond_stage_key | |
# The returned dict will have keys that correspond to the cond_stage_key | |
ret_dict = {} | |
if self.force_reload_pretrain_avoid_overwrite and not self.is_reload: | |
self.load_pretrain_model() | |
self.is_reload = True | |
# if(self.always_output_audiomae_gt or (self.use_gt_mae_output and torch.rand(1).item() < self.use_gt_mae_prob)): | |
# cond_dict = self.get_input(batch) | |
# ret_dict["crossattn_audiomae_generated"] = [cond_dict["crossattn_audiomae_pooled"][0], torch.ones_like(cond_dict["crossattn_audiomae_pooled"][1]).float()] # Input sequence and mask | |
# else: | |
input_embeds, cond_dict = self.generate(batch) | |
input_embeds_mask = ( | |
torch.ones((input_embeds.size(0), input_embeds.size(1))) | |
.to(input_embeds.device) | |
.float() | |
) | |
ret_dict["crossattn_audiomae_generated"] = [ | |
input_embeds, | |
input_embeds_mask, | |
] # Input sequence and mask | |
# If the following two keys are not in cond_stage_key, then they will not be used as condition | |
for key in cond_dict.keys(): | |
ret_dict[key] = cond_dict[key] | |
return ret_dict | |
class AudioMAEConditionCTPoolRandTFSeparated(nn.Module): | |
""" | |
audiomae = AudioMAEConditionCTPool2x2() | |
data = torch.randn((4, 1024, 128)) | |
output = audiomae(data) | |
import ipdb;ipdb.set_trace() | |
exit(0) | |
""" | |
def __init__( | |
self, | |
time_pooling_factors=[1, 2, 4, 8], | |
freq_pooling_factors=[1, 2, 4, 8], | |
eval_time_pooling=None, | |
eval_freq_pooling=None, | |
mask_ratio=0.0, | |
regularization=False, | |
no_audiomae_mask=True, | |
no_audiomae_average=False, | |
): | |
super().__init__() | |
self.device = None | |
self.time_pooling_factors = time_pooling_factors | |
self.freq_pooling_factors = freq_pooling_factors | |
self.no_audiomae_mask = no_audiomae_mask | |
self.no_audiomae_average = no_audiomae_average | |
self.eval_freq_pooling = eval_freq_pooling | |
self.eval_time_pooling = eval_time_pooling | |
self.mask_ratio = mask_ratio | |
self.use_reg = regularization | |
self.audiomae = Vanilla_AudioMAE() | |
self.audiomae.eval() | |
for p in self.audiomae.parameters(): | |
p.requires_grad = False | |
# Required | |
def get_unconditional_condition(self, batchsize): | |
param = next(self.audiomae.parameters()) | |
assert param.requires_grad == False | |
device = param.device | |
# time_pool, freq_pool = max(self.time_pooling_factors), max(self.freq_pooling_factors) | |
time_pool, freq_pool = min(self.eval_time_pooling, 64), min( | |
self.eval_freq_pooling, 8 | |
) | |
# time_pool = self.time_pooling_factors[np.random.choice(list(range(len(self.time_pooling_factors))))] | |
# freq_pool = self.freq_pooling_factors[np.random.choice(list(range(len(self.freq_pooling_factors))))] | |
token_num = int(512 / (time_pool * freq_pool)) | |
return [ | |
torch.zeros((batchsize, token_num, 768)).to(device).float(), | |
torch.ones((batchsize, token_num)).to(device).float(), | |
] | |
def pool(self, representation, time_pool=None, freq_pool=None): | |
assert representation.size(-1) == 768 | |
representation = representation[:, 1:, :].transpose(1, 2) | |
bs, embedding_dim, token_num = representation.size() | |
representation = representation.reshape(bs, embedding_dim, 64, 8) | |
if self.training: | |
if time_pool is None and freq_pool is None: | |
time_pool = min( | |
64, | |
self.time_pooling_factors[ | |
np.random.choice(list(range(len(self.time_pooling_factors)))) | |
], | |
) | |
freq_pool = min( | |
8, | |
self.freq_pooling_factors[ | |
np.random.choice(list(range(len(self.freq_pooling_factors)))) | |
], | |
) | |
# freq_pool = min(8, time_pool) # TODO here I make some modification. | |
else: | |
time_pool, freq_pool = min(self.eval_time_pooling, 64), min( | |
self.eval_freq_pooling, 8 | |
) | |
self.avgpooling = nn.AvgPool2d( | |
kernel_size=(time_pool, freq_pool), stride=(time_pool, freq_pool) | |
) | |
self.maxpooling = nn.MaxPool2d( | |
kernel_size=(time_pool, freq_pool), stride=(time_pool, freq_pool) | |
) | |
pooled = ( | |
self.avgpooling(representation) + self.maxpooling(representation) | |
) / 2 # [bs, embedding_dim, time_token_num, freq_token_num] | |
pooled = pooled.flatten(2).transpose(1, 2) | |
return pooled # [bs, token_num, embedding_dim] | |
def regularization(self, x): | |
assert x.size(-1) == 768 | |
x = F.normalize(x, p=2, dim=-1) | |
return x | |
# Required | |
def forward(self, batch, time_pool=None, freq_pool=None): | |
assert batch.size(-2) == 1024 and batch.size(-1) == 128 | |
if self.device is None: | |
self.device = batch.device | |
batch = batch.unsqueeze(1) | |
with torch.no_grad(): | |
representation = self.audiomae( | |
batch, | |
mask_ratio=self.mask_ratio, | |
no_mask=self.no_audiomae_mask, | |
no_average=self.no_audiomae_average, | |
) | |
representation = self.pool(representation, time_pool, freq_pool) | |
if self.use_reg: | |
representation = self.regularization(representation) | |
return [ | |
representation, | |
torch.ones((representation.size(0), representation.size(1))) | |
.to(representation.device) | |
.float(), | |
] | |
class AudioMAEConditionCTPoolRand(nn.Module): | |
""" | |
audiomae = AudioMAEConditionCTPool2x2() | |
data = torch.randn((4, 1024, 128)) | |
output = audiomae(data) | |
import ipdb;ipdb.set_trace() | |
exit(0) | |
""" | |
def __init__( | |
self, | |
time_pooling_factors=[1, 2, 4, 8], | |
freq_pooling_factors=[1, 2, 4, 8], | |
eval_time_pooling=None, | |
eval_freq_pooling=None, | |
mask_ratio=0.0, | |
regularization=False, | |
no_audiomae_mask=True, | |
no_audiomae_average=False, | |
): | |
super().__init__() | |
self.device = None | |
self.time_pooling_factors = time_pooling_factors | |
self.freq_pooling_factors = freq_pooling_factors | |
self.no_audiomae_mask = no_audiomae_mask | |
self.no_audiomae_average = no_audiomae_average | |
self.eval_freq_pooling = eval_freq_pooling | |
self.eval_time_pooling = eval_time_pooling | |
self.mask_ratio = mask_ratio | |
self.use_reg = regularization | |
self.audiomae = Vanilla_AudioMAE() | |
self.audiomae.eval() | |
for p in self.audiomae.parameters(): | |
p.requires_grad = False | |
# Required | |
def get_unconditional_condition(self, batchsize): | |
param = next(self.audiomae.parameters()) | |
assert param.requires_grad == False | |
device = param.device | |
# time_pool, freq_pool = max(self.time_pooling_factors), max(self.freq_pooling_factors) | |
time_pool, freq_pool = min(self.eval_time_pooling, 64), min( | |
self.eval_freq_pooling, 8 | |
) | |
# time_pool = self.time_pooling_factors[np.random.choice(list(range(len(self.time_pooling_factors))))] | |
# freq_pool = self.freq_pooling_factors[np.random.choice(list(range(len(self.freq_pooling_factors))))] | |
token_num = int(512 / (time_pool * freq_pool)) | |
return [ | |
torch.zeros((batchsize, token_num, 768)).to(device).float(), | |
torch.ones((batchsize, token_num)).to(device).float(), | |
] | |
def pool(self, representation, time_pool=None, freq_pool=None): | |
assert representation.size(-1) == 768 | |
representation = representation[:, 1:, :].transpose(1, 2) | |
bs, embedding_dim, token_num = representation.size() | |
representation = representation.reshape(bs, embedding_dim, 64, 8) | |
if self.training: | |
if time_pool is None and freq_pool is None: | |
time_pool = min( | |
64, | |
self.time_pooling_factors[ | |
np.random.choice(list(range(len(self.time_pooling_factors)))) | |
], | |
) | |
# freq_pool = self.freq_pooling_factors[np.random.choice(list(range(len(self.freq_pooling_factors))))] | |
freq_pool = min(8, time_pool) # TODO here I make some modification. | |
else: | |
time_pool, freq_pool = min(self.eval_time_pooling, 64), min( | |
self.eval_freq_pooling, 8 | |
) | |
self.avgpooling = nn.AvgPool2d( | |
kernel_size=(time_pool, freq_pool), stride=(time_pool, freq_pool) | |
) | |
self.maxpooling = nn.MaxPool2d( | |
kernel_size=(time_pool, freq_pool), stride=(time_pool, freq_pool) | |
) | |
pooled = ( | |
self.avgpooling(representation) + self.maxpooling(representation) | |
) / 2 # [bs, embedding_dim, time_token_num, freq_token_num] | |
pooled = pooled.flatten(2).transpose(1, 2) | |
return pooled # [bs, token_num, embedding_dim] | |
def regularization(self, x): | |
assert x.size(-1) == 768 | |
x = F.normalize(x, p=2, dim=-1) | |
return x | |
# Required | |
def forward(self, batch, time_pool=None, freq_pool=None): | |
assert batch.size(-2) == 1024 and batch.size(-1) == 128 | |
if self.device is None: | |
self.device = next(self.audiomae.parameters()).device | |
batch = batch.unsqueeze(1).to(self.device) | |
with torch.no_grad(): | |
representation = self.audiomae( | |
batch, | |
mask_ratio=self.mask_ratio, | |
no_mask=self.no_audiomae_mask, | |
no_average=self.no_audiomae_average, | |
) | |
representation = self.pool(representation, time_pool, freq_pool) | |
if self.use_reg: | |
representation = self.regularization(representation) | |
return [ | |
representation, | |
torch.ones((representation.size(0), representation.size(1))) | |
.to(representation.device) | |
.float(), | |
] | |
class CLAPAudioEmbeddingClassifierFreev2(nn.Module): | |
def __init__( | |
self, | |
pretrained_path="", | |
enable_cuda=False, | |
sampling_rate=16000, | |
embed_mode="audio", | |
amodel="HTSAT-base", | |
unconditional_prob=0.1, | |
random_mute=False, | |
max_random_mute_portion=0.5, | |
training_mode=True, | |
): | |
super().__init__() | |
self.device = "cpu" # The model itself is on cpu | |
self.cuda = enable_cuda | |
self.precision = "fp32" | |
self.amodel = amodel # or 'PANN-14' | |
self.tmodel = "roberta" # the best text encoder in our training | |
self.enable_fusion = False # False if you do not want to use the fusion model | |
self.fusion_type = "aff_2d" | |
self.pretrained = pretrained_path | |
self.embed_mode = embed_mode | |
self.embed_mode_orig = embed_mode | |
self.sampling_rate = sampling_rate | |
self.unconditional_prob = unconditional_prob | |
self.random_mute = random_mute | |
self.tokenize = RobertaTokenizer.from_pretrained("roberta-base") | |
self.max_random_mute_portion = max_random_mute_portion | |
self.training_mode = training_mode | |
self.model, self.model_cfg = create_model( | |
self.amodel, | |
self.tmodel, | |
self.pretrained, | |
precision=self.precision, | |
device=self.device, | |
enable_fusion=self.enable_fusion, | |
fusion_type=self.fusion_type, | |
) | |
self.model = self.model.to(self.device) | |
audio_cfg = self.model_cfg["audio_cfg"] | |
self.mel_transform = torchaudio.transforms.MelSpectrogram( | |
sample_rate=audio_cfg["sample_rate"], | |
n_fft=audio_cfg["window_size"], | |
win_length=audio_cfg["window_size"], | |
hop_length=audio_cfg["hop_size"], | |
center=True, | |
pad_mode="reflect", | |
power=2.0, | |
norm=None, | |
onesided=True, | |
n_mels=64, | |
f_min=audio_cfg["fmin"], | |
f_max=audio_cfg["fmax"], | |
) | |
for p in self.model.parameters(): | |
p.requires_grad = False | |
self.unconditional_token = None | |
self.model.eval() | |
def get_unconditional_condition(self, batchsize): | |
self.unconditional_token = self.model.get_text_embedding( | |
self.tokenizer(["", ""]) | |
)[0:1] | |
return torch.cat([self.unconditional_token.unsqueeze(0)] * batchsize, dim=0) | |
def batch_to_list(self, batch): | |
ret = [] | |
for i in range(batch.size(0)): | |
ret.append(batch[i]) | |
return ret | |
def make_decision(self, probability): | |
if float(torch.rand(1)) < probability: | |
return True | |
else: | |
return False | |
def random_uniform(self, start, end): | |
val = torch.rand(1).item() | |
return start + (end - start) * val | |
def _random_mute(self, waveform): | |
# waveform: [bs, t-steps] | |
t_steps = waveform.size(-1) | |
for i in range(waveform.size(0)): | |
mute_size = int( | |
self.random_uniform(0, end=int(t_steps * self.max_random_mute_portion)) | |
) | |
mute_start = int(self.random_uniform(0, t_steps - mute_size)) | |
waveform[i, mute_start : mute_start + mute_size] = 0 | |
return waveform | |
def cos_similarity(self, waveform, text): | |
# waveform: [bs, t_steps] | |
original_embed_mode = self.embed_mode | |
with torch.no_grad(): | |
self.embed_mode = "audio" | |
# MPS currently does not support ComplexFloat dtype and operator 'aten::_fft_r2c' | |
if self.cuda: | |
audio_emb = self(waveform.cuda()) | |
else: | |
audio_emb = self(waveform.to("cpu")) | |
self.embed_mode = "text" | |
text_emb = self(text) | |
similarity = F.cosine_similarity(audio_emb, text_emb, dim=2) | |
self.embed_mode = original_embed_mode | |
return similarity.squeeze() | |
def build_unconditional_emb(self): | |
self.unconditional_token = self.model.get_text_embedding( | |
self.tokenizer(["", ""]) | |
)[0:1] | |
def forward(self, batch): | |
# If you want this conditioner to be unconditional, set self.unconditional_prob = 1.0 | |
# If you want this conditioner to be fully conditional, set self.unconditional_prob = 0.0 | |
if self.model.training == True and not self.training_mode: | |
print( | |
"The pretrained CLAP model should always be in eval mode. Reloading model just in case you change the parameters." | |
) | |
self.model, self.model_cfg = create_model( | |
self.amodel, | |
self.tmodel, | |
self.pretrained, | |
precision=self.precision, | |
device="cuda" if self.cuda else "cpu", | |
enable_fusion=self.enable_fusion, | |
fusion_type=self.fusion_type, | |
) | |
for p in self.model.parameters(): | |
p.requires_grad = False | |
self.model.eval() | |
if self.unconditional_token is None: | |
self.build_unconditional_emb() | |
# if(self.training_mode): | |
# assert self.model.training == True | |
# else: | |
# assert self.model.training == False | |
# the 'fusion' truncate mode can be changed to 'rand_trunc' if run in unfusion mode | |
if self.embed_mode == "audio": | |
if not self.training: | |
print("INFO: clap model calculate the audio embedding as condition") | |
with torch.no_grad(): | |
# assert ( | |
# self.sampling_rate == 16000 | |
# ), "We only support 16000 sampling rate" | |
# if self.random_mute: | |
# batch = self._random_mute(batch) | |
# batch: [bs, 1, t-samples] | |
if self.sampling_rate != 48000: | |
batch = torchaudio.functional.resample( | |
batch, orig_freq=self.sampling_rate, new_freq=48000 | |
) | |
audio_data = batch.squeeze(1).to("cpu") | |
self.mel_transform = self.mel_transform.to(audio_data.device) | |
mel = self.mel_transform(audio_data) | |
audio_dict = get_audio_features( | |
audio_data, | |
mel, | |
480000, | |
data_truncating="fusion", | |
data_filling="repeatpad", | |
audio_cfg=self.model_cfg["audio_cfg"], | |
) | |
# [bs, 512] | |
embed = self.model.get_audio_embedding(audio_dict) | |
elif self.embed_mode == "text": | |
with torch.no_grad(): | |
# the 'fusion' truncate mode can be changed to 'rand_trunc' if run in unfusion mode | |
text_data = self.tokenizer(batch) | |
if isinstance(batch, str) or ( | |
isinstance(batch, list) and len(batch) == 1 | |
): | |
for key in text_data.keys(): | |
text_data[key] = text_data[key].unsqueeze(0) | |
embed = self.model.get_text_embedding(text_data) | |
embed = embed.unsqueeze(1) | |
for i in range(embed.size(0)): | |
if self.make_decision(self.unconditional_prob): | |
embed[i] = self.unconditional_token | |
# embed = torch.randn((batch.size(0), 1, 512)).type_as(batch) | |
return embed.detach() | |
def tokenizer(self, text): | |
result = self.tokenize( | |
text, | |
padding="max_length", | |
truncation=True, | |
max_length=512, | |
return_tensors="pt", | |
) | |
return {k: v.squeeze(0) for k, v in result.items()} | |