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""" CLAP Model | |
Adapted from CLIP: https://github.com/openai/CLIP. Originally MIT License, Copyright (c) 2021 OpenAI. | |
Adapted to the Audio Task. | |
""" | |
from collections import OrderedDict | |
from dataclasses import dataclass | |
from typing import Tuple, Union, Callable, Optional | |
import numpy as np | |
import torch | |
import torch.nn.functional as F | |
from torch import nn | |
import logging | |
from .utils import freeze_batch_norm_2d | |
from .pann_model import create_pann_model | |
from .htsat import create_htsat_model | |
from transformers import BertModel, RobertaModel, BartModel, RobertaConfig | |
class MLPLayers(nn.Module): | |
def __init__(self, units=[512, 512, 512], nonlin=nn.ReLU(), dropout=0.1): | |
super(MLPLayers, self).__init__() | |
self.nonlin = nonlin | |
self.dropout = dropout | |
sequence = [] | |
for u0, u1 in zip(units[:-1], units[1:]): | |
sequence.append(nn.Linear(u0, u1)) | |
sequence.append(self.nonlin) | |
sequence.append(nn.Dropout(self.dropout)) | |
sequence = sequence[:-2] | |
self.sequential = nn.Sequential(*sequence) | |
def forward(self, X): | |
X = self.sequential(X) | |
return X | |
class Bottleneck(nn.Module): | |
expansion = 4 | |
def __init__(self, inplanes, planes, stride=1): | |
super().__init__() | |
# all conv layers have stride 1. an avgpool is performed after the second convolution when stride > 1 | |
self.conv1 = nn.Conv2d(inplanes, planes, 1, bias=False) | |
self.bn1 = nn.BatchNorm2d(planes) | |
self.conv2 = nn.Conv2d(planes, planes, 3, padding=1, bias=False) | |
self.bn2 = nn.BatchNorm2d(planes) | |
self.avgpool = nn.AvgPool2d(stride) if stride > 1 else nn.Identity() | |
self.conv3 = nn.Conv2d(planes, planes * self.expansion, 1, bias=False) | |
self.bn3 = nn.BatchNorm2d(planes * self.expansion) | |
self.relu = nn.ReLU(inplace=True) | |
self.downsample = None | |
self.stride = stride | |
if stride > 1 or inplanes != planes * Bottleneck.expansion: | |
# downsampling layer is prepended with an avgpool, and the subsequent convolution has stride 1 | |
self.downsample = nn.Sequential( | |
OrderedDict( | |
[ | |
("-1", nn.AvgPool2d(stride)), | |
( | |
"0", | |
nn.Conv2d( | |
inplanes, | |
planes * self.expansion, | |
1, | |
stride=1, | |
bias=False, | |
), | |
), | |
("1", nn.BatchNorm2d(planes * self.expansion)), | |
] | |
) | |
) | |
def forward(self, x: torch.Tensor): | |
identity = x | |
out = self.relu(self.bn1(self.conv1(x))) | |
out = self.relu(self.bn2(self.conv2(out))) | |
out = self.avgpool(out) | |
out = self.bn3(self.conv3(out)) | |
if self.downsample is not None: | |
identity = self.downsample(x) | |
out += identity | |
out = self.relu(out) | |
return out | |
class AttentionPool2d(nn.Module): | |
def __init__( | |
self, spacial_dim: int, embed_dim: int, num_heads: int, output_dim: int = None | |
): | |
super().__init__() | |
self.positional_embedding = nn.Parameter( | |
torch.randn(spacial_dim**2 + 1, embed_dim) / embed_dim**0.5 | |
) | |
self.k_proj = nn.Linear(embed_dim, embed_dim) | |
self.q_proj = nn.Linear(embed_dim, embed_dim) | |
self.v_proj = nn.Linear(embed_dim, embed_dim) | |
self.c_proj = nn.Linear(embed_dim, output_dim or embed_dim) | |
self.num_heads = num_heads | |
def forward(self, x): | |
x = x.reshape(x.shape[0], x.shape[1], x.shape[2] * x.shape[3]).permute( | |
2, 0, 1 | |
) # NCHW -> (HW)NC | |
x = torch.cat([x.mean(dim=0, keepdim=True), x], dim=0) # (HW+1)NC | |
x = x + self.positional_embedding[:, None, :].to(x.dtype) # (HW+1)NC | |
x, _ = F.multi_head_attention_forward( | |
query=x, | |
key=x, | |
value=x, | |
embed_dim_to_check=x.shape[-1], | |
num_heads=self.num_heads, | |
q_proj_weight=self.q_proj.weight, | |
k_proj_weight=self.k_proj.weight, | |
v_proj_weight=self.v_proj.weight, | |
in_proj_weight=None, | |
in_proj_bias=torch.cat( | |
[self.q_proj.bias, self.k_proj.bias, self.v_proj.bias] | |
), | |
bias_k=None, | |
bias_v=None, | |
add_zero_attn=False, | |
dropout_p=0, | |
out_proj_weight=self.c_proj.weight, | |
out_proj_bias=self.c_proj.bias, | |
use_separate_proj_weight=True, | |
training=self.training, | |
need_weights=False, | |
) | |
return x[0] | |
class ModifiedResNet(nn.Module): | |
""" | |
A ResNet class that is similar to torchvision's but contains the following changes: | |
- There are now 3 "stem" convolutions as opposed to 1, with an average pool instead of a max pool. | |
- Performs anti-aliasing strided convolutions, where an avgpool is prepended to convolutions with stride > 1 | |
- The final pooling layer is a QKV attention instead of an average pool | |
""" | |
def __init__(self, layers, output_dim, heads, image_size=224, width=64): | |
super().__init__() | |
self.output_dim = output_dim | |
self.image_size = image_size | |
# the 3-layer stem | |
self.conv1 = nn.Conv2d( | |
3, width // 2, kernel_size=3, stride=2, padding=1, bias=False | |
) | |
self.bn1 = nn.BatchNorm2d(width // 2) | |
self.conv2 = nn.Conv2d( | |
width // 2, width // 2, kernel_size=3, padding=1, bias=False | |
) | |
self.bn2 = nn.BatchNorm2d(width // 2) | |
self.conv3 = nn.Conv2d(width // 2, width, kernel_size=3, padding=1, bias=False) | |
self.bn3 = nn.BatchNorm2d(width) | |
self.avgpool = nn.AvgPool2d(2) | |
self.relu = nn.ReLU(inplace=True) | |
# residual layers | |
self._inplanes = width # this is a *mutable* variable used during construction | |
self.layer1 = self._make_layer(width, layers[0]) | |
self.layer2 = self._make_layer(width * 2, layers[1], stride=2) | |
self.layer3 = self._make_layer(width * 4, layers[2], stride=2) | |
self.layer4 = self._make_layer(width * 8, layers[3], stride=2) | |
embed_dim = width * 32 # the ResNet feature dimension | |
self.attnpool = AttentionPool2d(image_size // 32, embed_dim, heads, output_dim) | |
self.init_parameters() | |
def _make_layer(self, planes, blocks, stride=1): | |
layers = [Bottleneck(self._inplanes, planes, stride)] | |
self._inplanes = planes * Bottleneck.expansion | |
for _ in range(1, blocks): | |
layers.append(Bottleneck(self._inplanes, planes)) | |
return nn.Sequential(*layers) | |
def init_parameters(self): | |
if self.attnpool is not None: | |
std = self.attnpool.c_proj.in_features**-0.5 | |
nn.init.normal_(self.attnpool.q_proj.weight, std=std) | |
nn.init.normal_(self.attnpool.k_proj.weight, std=std) | |
nn.init.normal_(self.attnpool.v_proj.weight, std=std) | |
nn.init.normal_(self.attnpool.c_proj.weight, std=std) | |
for resnet_block in [self.layer1, self.layer2, self.layer3, self.layer4]: | |
for name, param in resnet_block.named_parameters(): | |
if name.endswith("bn3.weight"): | |
nn.init.zeros_(param) | |
def lock(self, unlocked_groups=0, freeze_bn_stats=False): | |
assert ( | |
unlocked_groups == 0 | |
), "partial locking not currently supported for this model" | |
for param in self.parameters(): | |
param.requires_grad = False | |
if freeze_bn_stats: | |
freeze_batch_norm_2d(self) | |
def stem(self, x): | |
for conv, bn in [ | |
(self.conv1, self.bn1), | |
(self.conv2, self.bn2), | |
(self.conv3, self.bn3), | |
]: | |
x = self.relu(bn(conv(x))) | |
x = self.avgpool(x) | |
return x | |
def forward(self, x): | |
x = self.stem(x) | |
x = self.layer1(x) | |
x = self.layer2(x) | |
x = self.layer3(x) | |
x = self.layer4(x) | |
x = self.attnpool(x) | |
return x | |
class LayerNorm(nn.LayerNorm): | |
"""Subclass torch's LayerNorm to handle fp16.""" | |
def forward(self, x: torch.Tensor): | |
orig_type = x.dtype | |
x = F.layer_norm(x, self.normalized_shape, self.weight, self.bias, self.eps) | |
return x.to(orig_type) | |
class QuickGELU(nn.Module): | |
# NOTE This is slower than nn.GELU or nn.SiLU and uses more GPU memory | |
def forward(self, x: torch.Tensor): | |
return x * torch.sigmoid(1.702 * x) | |
class ResidualAttentionBlock(nn.Module): | |
def __init__(self, d_model: int, n_head: int, act_layer: Callable = nn.GELU): | |
super().__init__() | |
self.attn = nn.MultiheadAttention(d_model, n_head) | |
self.ln_1 = LayerNorm(d_model) | |
self.mlp = nn.Sequential( | |
OrderedDict( | |
[ | |
("c_fc", nn.Linear(d_model, d_model * 4)), | |
("gelu", act_layer()), | |
("c_proj", nn.Linear(d_model * 4, d_model)), | |
] | |
) | |
) | |
self.ln_2 = LayerNorm(d_model) | |
def attention(self, x: torch.Tensor, attn_mask: Optional[torch.Tensor] = None): | |
return self.attn(x, x, x, need_weights=False, attn_mask=attn_mask)[0] | |
def forward(self, x: torch.Tensor, attn_mask: Optional[torch.Tensor] = None): | |
x = x + self.attention(self.ln_1(x), attn_mask=attn_mask) | |
x = x + self.mlp(self.ln_2(x)) | |
return x | |
class Transformer(nn.Module): | |
def __init__( | |
self, width: int, layers: int, heads: int, act_layer: Callable = nn.GELU | |
): | |
super().__init__() | |
self.width = width | |
self.layers = layers | |
self.resblocks = nn.ModuleList( | |
[ | |
ResidualAttentionBlock(width, heads, act_layer=act_layer) | |
for _ in range(layers) | |
] | |
) | |
def forward(self, x: torch.Tensor, attn_mask: Optional[torch.Tensor] = None): | |
for r in self.resblocks: | |
x = r(x, attn_mask=attn_mask) | |
return x | |
class VisualTransformer(nn.Module): | |
def __init__( | |
self, | |
image_size: int, | |
patch_size: int, | |
width: int, | |
layers: int, | |
heads: int, | |
output_dim: int, | |
act_layer: Callable = nn.GELU, | |
): | |
super().__init__() | |
self.image_size = image_size | |
self.output_dim = output_dim | |
self.conv1 = nn.Conv2d( | |
in_channels=3, | |
out_channels=width, | |
kernel_size=patch_size, | |
stride=patch_size, | |
bias=False, | |
) | |
scale = width**-0.5 | |
self.class_embedding = nn.Parameter(scale * torch.randn(width)) | |
self.positional_embedding = nn.Parameter( | |
scale * torch.randn((image_size // patch_size) ** 2 + 1, width) | |
) | |
self.ln_pre = LayerNorm(width) | |
self.text_branch = Transformer(width, layers, heads, act_layer=act_layer) | |
self.ln_post = LayerNorm(width) | |
self.proj = nn.Parameter(scale * torch.randn(width, output_dim)) | |
def lock(self, unlocked_groups=0, freeze_bn_stats=False): | |
assert ( | |
unlocked_groups == 0 | |
), "partial locking not currently supported for this model" | |
for param in self.parameters(): | |
param.requires_grad = False | |
def forward(self, x: torch.Tensor): | |
x = self.conv1(x) # shape = [*, width, grid, grid] | |
x = x.reshape(x.shape[0], x.shape[1], -1) # shape = [*, width, grid ** 2] | |
x = x.permute(0, 2, 1) # shape = [*, grid ** 2, width] | |
x = torch.cat( | |
[ | |
self.class_embedding.to(x.dtype) | |
+ torch.zeros( | |
x.shape[0], 1, x.shape[-1], dtype=x.dtype, device=x.device | |
), | |
x, | |
], | |
dim=1, | |
) # shape = [*, grid ** 2 + 1, width] | |
x = x + self.positional_embedding.to(x.dtype) | |
x = self.ln_pre(x) | |
x = x.permute(1, 0, 2) # NLD -> LND | |
x = self.text_branch(x) | |
x = x.permute(1, 0, 2) # LND -> NLD | |
x = self.ln_post(x[:, 0, :]) | |
if self.proj is not None: | |
x = x @ self.proj | |
return x | |
class CLAPVisionCfg: | |
layers: Union[Tuple[int, int, int, int], int] = 12 | |
width: int = 768 | |
patch_size: int = 16 | |
image_size: Union[Tuple[int, int], int] = 224 | |
timm_model_name: str = ( | |
None # a valid model name overrides layers, width, patch_size | |
) | |
timm_model_pretrained: bool = ( | |
False # use (imagenet) pretrained weights for named model | |
) | |
timm_pool: str = ( | |
"avg" # feature pooling for timm model ('abs_attn', 'rot_attn', 'avg', '') | |
) | |
timm_proj: str = ( | |
"linear" # linear projection for timm model output ('linear', 'mlp', '') | |
) | |
# Audio Config Class | |
class CLAPAudioCfp: | |
model_type: str = "PANN" | |
model_name: str = "Cnn14" | |
sample_rate: int = 48000 | |
# Param | |
audio_length: int = 1024 | |
window_size: int = 1024 | |
hop_size: int = 1024 | |
fmin: int = 50 | |
fmax: int = 14000 | |
class_num: int = 527 | |
mel_bins: int = 64 | |
clip_samples: int = 480000 | |
class CLAPTextCfg: | |
context_length: int | |
vocab_size: int | |
width: int | |
heads: int | |
layers: int | |
model_type: str | |
class CLAP(nn.Module): | |
def __init__( | |
self, | |
embed_dim: int, | |
audio_cfg: CLAPAudioCfp, | |
text_cfg: CLAPTextCfg, | |
quick_gelu: bool = False, | |
enable_fusion: bool = False, | |
fusion_type: str = "None", | |
joint_embed_shape: int = 512, | |
mlp_act: str = "relu", | |
): | |
super().__init__() | |
if isinstance(audio_cfg, dict): | |
audio_cfg = CLAPAudioCfp(**audio_cfg) | |
if isinstance(text_cfg, dict): | |
text_cfg = CLAPTextCfg(**text_cfg) | |
self.audio_cfg = audio_cfg | |
self.text_cfg = text_cfg | |
self.enable_fusion = enable_fusion | |
self.fusion_type = fusion_type | |
self.joint_embed_shape = joint_embed_shape | |
self.mlp_act = mlp_act | |
self.context_length = text_cfg.context_length | |
# OpenAI models are pretrained w/ QuickGELU but native nn.GELU is both faster and more | |
# memory efficient in recent PyTorch releases (>= 1.10). | |
# NOTE: timm models always use native GELU regardless of quick_gelu flag. | |
act_layer = QuickGELU if quick_gelu else nn.GELU | |
if mlp_act == "relu": | |
mlp_act_layer = nn.ReLU() | |
elif mlp_act == "gelu": | |
mlp_act_layer = nn.GELU() | |
else: | |
raise NotImplementedError | |
# audio branch | |
# audio branch parameters | |
if audio_cfg.model_type == "PANN": | |
self.audio_branch = create_pann_model(audio_cfg, enable_fusion, fusion_type) | |
elif audio_cfg.model_type == "HTSAT": | |
self.audio_branch = create_htsat_model( | |
audio_cfg, enable_fusion, fusion_type | |
) | |
else: | |
logging.error(f"Model config for {audio_cfg.model_type} not found") | |
raise RuntimeError(f"Model config for {audio_cfg.model_type} not found.") | |
# text branch | |
# text branch parameters | |
if text_cfg.model_type == "transformer": | |
self.text_branch = Transformer( | |
width=text_cfg.width, | |
layers=text_cfg.layers, | |
heads=text_cfg.heads, | |
act_layer=act_layer, | |
) | |
self.vocab_size = text_cfg.vocab_size | |
self.token_embedding = nn.Embedding(text_cfg.vocab_size, text_cfg.width) | |
self.positional_embedding = nn.Parameter( | |
torch.empty(self.context_length, text_cfg.width) | |
) | |
self.ln_final = LayerNorm(text_cfg.width) | |
self.text_transform = MLPLayers( | |
units=[ | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
], | |
dropout=0.1, | |
) | |
self.text_projection = nn.Sequential( | |
nn.Linear(text_cfg.width, self.joint_embed_shape), | |
mlp_act_layer, | |
nn.Linear(self.joint_embed_shape, self.joint_embed_shape), | |
) | |
elif text_cfg.model_type == "bert": | |
self.text_branch = BertModel.from_pretrained("bert-base-uncased") | |
self.text_transform = MLPLayers( | |
units=[ | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
], | |
dropout=0.1, | |
) | |
self.text_projection = nn.Sequential( | |
nn.Linear(768, self.joint_embed_shape), | |
mlp_act_layer, | |
nn.Linear(self.joint_embed_shape, self.joint_embed_shape), | |
) | |
elif text_cfg.model_type == "roberta": | |
self.text_branch = RobertaModel( | |
RobertaConfig.from_pretrained("roberta-base") | |
) | |
self.text_transform = MLPLayers( | |
units=[ | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
], | |
dropout=0.1, | |
) | |
self.text_projection = nn.Sequential( | |
nn.Linear(768, self.joint_embed_shape), | |
mlp_act_layer, | |
nn.Linear(self.joint_embed_shape, self.joint_embed_shape), | |
) | |
elif text_cfg.model_type == "bart": | |
self.text_branch = BartModel.from_pretrained("facebook/bart-base") | |
self.text_transform = MLPLayers( | |
units=[ | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
], | |
dropout=0.1, | |
) | |
self.text_projection = nn.Sequential( | |
nn.Linear(768, self.joint_embed_shape), | |
mlp_act_layer, | |
nn.Linear(self.joint_embed_shape, self.joint_embed_shape), | |
) | |
else: | |
logging.error(f"Model config for {text_cfg.model_type} not found") | |
raise RuntimeError(f"Model config for {text_cfg.model_type} not found.") | |
self.text_branch_type = text_cfg.model_type | |
# text branch parameters | |
# audio branch parameters | |
self.audio_transform = MLPLayers( | |
units=[ | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
self.joint_embed_shape, | |
], | |
dropout=0.1, | |
) | |
# below here is text branch parameters | |
# ============================================================================================================ | |
self.audio_projection = nn.Sequential( | |
nn.Linear(embed_dim, self.joint_embed_shape), | |
mlp_act_layer, | |
nn.Linear(self.joint_embed_shape, self.joint_embed_shape), | |
) | |
self.logit_scale_a = nn.Parameter(torch.ones([]) * np.log(1 / 0.07)) | |
self.logit_scale_t = nn.Parameter(torch.ones([]) * np.log(1 / 0.07)) | |
self.register_buffer("attn_mask", self.build_attention_mask(), persistent=False) | |
self.init_text_branch_parameters() | |
def init_text_branch_parameters(self): | |
if self.text_branch_type == "transformer": | |
nn.init.normal_(self.token_embedding.weight, std=0.02) | |
nn.init.normal_(self.positional_embedding, std=0.01) | |
proj_std = (self.text_branch.width**-0.5) * ( | |
(2 * self.text_branch.layers) ** -0.5 | |
) | |
attn_std = self.text_branch.width**-0.5 | |
fc_std = (2 * self.text_branch.width) ** -0.5 | |
for block in self.text_branch.resblocks: | |
nn.init.normal_(block.attn.in_proj_weight, std=attn_std) | |
nn.init.normal_(block.attn.out_proj.weight, std=proj_std) | |
nn.init.normal_(block.mlp.c_fc.weight, std=fc_std) | |
nn.init.normal_(block.mlp.c_proj.weight, std=proj_std) | |
if self.text_branch_type == "bert" or self.text_branch_type == "roberta": | |
self.text_branch.embeddings.word_embeddings.weight.shape[-1] | |
elif self.text_branch_type == "bart": | |
self.text_branch.shared.weight.shape[-1] | |
else: | |
self.text_branch.width | |
nn.init.constant_(self.logit_scale_a, np.log(1 / 0.07)) | |
nn.init.constant_(self.logit_scale_t, np.log(1 / 0.07)) | |
# deprecated | |
# if hasattr(self.visual, 'init_parameters'): | |
# self.visual.init_parameters() | |
# if self.text_projection is not None: | |
# nn.init.normal_(self.text_projection, std=width**-0.5) | |
def build_attention_mask(self): | |
# lazily create causal attention mask, with full attention between the vision tokens | |
# pytorch uses additive attention mask; fill with -inf | |
mask = torch.empty(self.context_length, self.context_length) | |
mask.fill_(float("-inf")) | |
mask.triu_(1) # zero out the lower diagonal | |
return mask | |
def encode_audio(self, audio, device): | |
return self.audio_branch( | |
audio, mixup_lambda=None, device=device | |
) # mix lambda needs to add | |
# def list_of_dict_of_tensor2dict_of_tensor(self, x, device): | |
# tmp = {} | |
# for k in x[0].keys(): | |
# tmp[k] = [] | |
# for i in range(len(x)): | |
# tmp[k].append(x[i][k][:77]) | |
# for k in x[0].keys(): | |
# tmp[k] = torch.tensor(tmp[k]).to(device=device, non_blocking=True) | |
# return tmp | |
def encode_text(self, text, device): | |
if self.text_branch_type == "transformer": | |
text = text.to(device=device, non_blocking=True) | |
x = self.token_embedding(text) # [batch_size, n_ctx, d_model] | |
x = x + self.positional_embedding | |
x = x.permute(1, 0, 2) # NLD -> LND | |
x = self.text_branch(x, attn_mask=self.attn_mask) | |
x = x.permute(1, 0, 2) # LND -> NLD | |
x = self.ln_final(x) | |
# x.shape = [batch_size, n_ctx, transformer.width] | |
# take features from the eot embedding (eot_token is the highest number in each sequence) | |
x = self.text_projection(x[torch.arange(x.shape[0]), text.argmax(dim=-1)]) | |
elif self.text_branch_type == "bert": | |
# text = self.list_of_dict_of_tensor2dict_of_tensor(text, device) | |
# text = BatchEncoding(text) | |
x = self.text_branch( | |
input_ids=text["input_ids"].to(device=device, non_blocking=True), | |
attention_mask=text["attention_mask"].to( | |
device=device, non_blocking=True | |
), | |
token_type_ids=text["token_type_ids"].to( | |
device=device, non_blocking=True | |
), | |
)["pooler_output"] | |
x = self.text_projection(x) | |
elif self.text_branch_type == "roberta": | |
x = self.text_branch( | |
input_ids=text["input_ids"].to(device=device, non_blocking=True), | |
attention_mask=text["attention_mask"].to( | |
device=device, non_blocking=True | |
), | |
)["pooler_output"] | |
x = self.text_projection(x) | |
elif self.text_branch_type == "bart": | |
x = torch.mean( | |
self.text_branch( | |
input_ids=text["input_ids"].to(device=device, non_blocking=True), | |
attention_mask=text["attention_mask"].to( | |
device=device, non_blocking=True | |
), | |
)["encoder_last_hidden_state"], | |
axis=1, | |
) | |
x = self.text_projection(x) | |
else: | |
logging.error(f"Model type {self.text_branch_type} not found") | |
raise RuntimeError(f"Model type {self.text_branch_type} not found.") | |
return x | |
def forward(self, audio, text, device=None): | |
"""Forward audio and text into the CLAP | |
Parameters | |
---------- | |
audio: torch.Tensor (batch_size, audio_length) | |
the time-domain audio input / the batch of mel_spec and longer list. | |
text: torch.Tensor () // need to add | |
the text token input | |
""" | |
if device is None: | |
if audio is not None: | |
device = audio.device | |
elif text is not None: | |
device = text.device | |
if audio is None and text is None: | |
# a hack to get the logit scale | |
return self.logit_scale_a.exp(), self.logit_scale_t.exp() | |
elif audio is None: | |
return self.encode_text(text, device=device) | |
elif text is None: | |
return self.audio_projection( | |
self.encode_audio(audio, device=device)["embedding"] | |
) | |
audio_features = self.audio_projection( | |
self.encode_audio(audio, device=device)["embedding"] | |
) | |
audio_features = F.normalize(audio_features, dim=-1) | |
text_features = self.encode_text(text, device=device) | |
# print("text_features", text_features) | |
# print("text_features.shape", text_features.shape) | |
# print("text_features.type", type(text_features)) | |
text_features = F.normalize(text_features, dim=-1) | |
audio_features_mlp = self.audio_transform(audio_features) | |
text_features_mlp = self.text_transform(text_features) | |
# Four outputs: audio features (basic & MLP), text features (basic & MLP) | |
return ( | |
audio_features, | |
text_features, | |
audio_features_mlp, | |
text_features_mlp, | |
self.logit_scale_a.exp(), | |
self.logit_scale_t.exp(), | |
) | |
def get_logit_scale(self): | |
return self.logit_scale_a.exp(), self.logit_scale_t.exp() | |
def get_text_embedding(self, data): | |
"""Get the text embedding from the model | |
Parameters | |
---------- | |
data: torch.Tensor | |
a tensor of text embedding | |
Returns | |
---------- | |
text_embed: torch.Tensor | |
a tensor of text_embeds (N, D) | |
""" | |
device = next(self.parameters()).device | |
for k in data: | |
data[k] = data[k].to(device) | |
text_embeds = self.encode_text(data, device=device) | |
text_embeds = F.normalize(text_embeds, dim=-1) | |
return text_embeds | |
def get_audio_embedding(self, data): | |
"""Get the audio embedding from the model | |
Parameters | |
---------- | |
data: a list of dict | |
the audio input dict list from 'get_audio_feature' method | |
Returns | |
---------- | |
audio_embed: torch.Tensor | |
a tensor of audio_embeds (N, D) | |
""" | |
device = next(self.parameters()).device | |
# input_dict = {} | |
# keys = data[0].keys() | |
# for k in keys: | |
# input_dict[k] = torch.cat([d[k].unsqueeze(0) for d in data], dim=0).to( | |
# device | |
# ) | |
audio_embeds = self.audio_projection( | |
self.encode_audio(data, device=device)["embedding"] | |
) | |
audio_embeds = F.normalize(audio_embeds, dim=-1) | |
return audio_embeds | |
def audio_infer(self, audio, hopsize=None, device=None): | |
"""Forward one audio and produce the audio embedding | |
Parameters | |
---------- | |
audio: (audio_length) | |
the time-domain audio input, notice that it must be only one input | |
hopsize: int | |
the overlap hopsize as the sliding window | |
Returns | |
---------- | |
output_dict: { | |
key: [n, (embedding_shape)] if "HTS-AT" | |
or | |
key: [(embedding_shape)] if "PANN" | |
} | |
the list of key values of the audio branch | |
""" | |
assert not self.training, "the inference mode must be run at eval stage" | |
output_dict = {} | |
# PANN | |
if self.audio_cfg.model_type == "PANN": | |
audio_input = audio.unsqueeze(dim=0) | |
output_dict[key] = self.encode_audio(audio_input, device=device)[ | |
key | |
].squeeze(dim=0) | |
elif self.audio_cfg.model_type == "HTSAT": | |
# repeat | |
audio_len = len(audio) | |
k = self.audio_cfg.clip_samples // audio_len | |
if k > 1: | |
audio = audio.repeat(k) | |
audio_len = len(audio) | |
if hopsize is None: | |
hopsize = min(hopsize, audio_len) | |
if audio_len > self.audio_cfg.clip_samples: | |
audio_input = [ | |
audio[pos : pos + self.audio_cfg.clip_samples].clone() | |
for pos in range( | |
0, audio_len - self.audio_cfg.clip_samples, hopsize | |
) | |
] | |
audio_input.append(audio[-self.audio_cfg.clip_samples :].clone()) | |
audio_input = torch.stack(audio_input) | |
output_dict[key] = self.encode_audio(audio_input, device=device)[key] | |
else: | |
audio_input = audio.unsqueeze(dim=0) | |
output_dict[key] = self.encode_audio(audio_input, device=device)[ | |
key | |
].squeeze(dim=0) | |
return output_dict | |
def convert_weights_to_fp16(model: nn.Module): | |
"""Convert applicable model parameters to fp16""" | |
def _convert_weights_to_fp16(l): | |
if isinstance(l, (nn.Conv1d, nn.Conv2d, nn.Linear)): | |
l.weight.data = l.weight.data.half() | |
if l.bias is not None: | |
l.bias.data = l.bias.data.half() | |
if isinstance(l, nn.MultiheadAttention): | |
for attr in [ | |
*[f"{s}_proj_weight" for s in ["in", "q", "k", "v"]], | |
"in_proj_bias", | |
"bias_k", | |
"bias_v", | |
]: | |
tensor = getattr(l, attr) | |
if tensor is not None: | |
tensor.data = tensor.data.half() | |
for name in ["text_projection", "proj"]: | |
if hasattr(l, name): | |
attr = getattr(l, name) | |
if attr is not None: | |
attr.data = attr.data.half() | |
model.apply(_convert_weights_to_fp16) | |
# Ignore the state dict of the vision part | |
def build_model_from_openai_state_dict( | |
state_dict: dict, model_cfg, enable_fusion: bool = False, fusion_type: str = "None" | |
): | |
embed_dim = model_cfg["embed_dim"] | |
audio_cfg = model_cfg["audio_cfg"] | |
text_cfg = model_cfg["text_cfg"] | |
state_dict["positional_embedding"].shape[0] | |
state_dict["token_embedding.weight"].shape[0] | |
transformer_width = state_dict["ln_final.weight"].shape[0] | |
transformer_width // 64 | |
transformer_layers = len( | |
set( | |
k.split(".")[2] | |
for k in state_dict | |
if k.startswith(f"transformer.resblocks") | |
) | |
) | |
audio_cfg = CLAPAudioCfp(**audio_cfg) | |
text_cfg = CLAPTextCfg(**text_cfg) | |
model = CLAP( | |
embed_dim, | |
audio_cfg=audio_cfg, | |
text_cfg=text_cfg, | |
quick_gelu=True, # OpenAI models were trained with QuickGELU | |
enable_fusion=enable_fusion, | |
fusion_type=fusion_type, | |
) | |
state_dict["logit_scale_a"] = state_dict["logit_scale"] | |
state_dict["logit_scale_t"] = state_dict["logit_scale"] | |
pop_keys = list(state_dict.keys())[::] | |
# pop the visual branch saved weights | |
for key in pop_keys: | |
if key.startswith("visual."): | |
state_dict.pop(key, None) | |
for key in ["logit_scale", "input_resolution", "context_length", "vocab_size"]: | |
state_dict.pop(key, None) | |
# not use fp16 | |
# convert_weights_to_fp16(model) | |
model.load_state_dict(state_dict, strict=False) | |
return model.eval() | |
def trace_model(model, batch_size=256, device=torch.device("cpu")): | |
model.eval() | |
audio_length = model.audio_cfg.audio_length | |
example_audio = torch.ones((batch_size, audio_length), device=device) | |
example_text = torch.zeros( | |
(batch_size, model.context_length), dtype=torch.int, device=device | |
) | |
model = torch.jit.trace_module( | |
model, | |
inputs=dict( | |
forward=(example_audio, example_text), | |
encode_text=(example_text,), | |
encode_image=(example_audio,), | |
), | |
) | |
model.audio_cfg.audio_length = audio_length # Question: what does this do? | |
return model | |