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#!/usr/bin/env python3 | |
# -*- encoding: utf-8 -*- | |
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved. | |
# MIT License (https://opensource.org/licenses/MIT) | |
# Modified from https://github.com/ddlBoJack/emotion2vec/tree/main | |
import os | |
import time | |
import torch | |
import logging | |
import numpy as np | |
from functools import partial | |
from omegaconf import OmegaConf | |
import torch.nn.functional as F | |
from contextlib import contextmanager | |
from distutils.version import LooseVersion | |
from funasr_detach.register import tables | |
from funasr_detach.models.emotion2vec.modules import AltBlock | |
from funasr_detach.models.emotion2vec.audio import AudioEncoder | |
from funasr_detach.utils.load_utils import load_audio_text_image_video | |
logger = logging.getLogger(__name__) | |
if LooseVersion(torch.__version__) >= LooseVersion("1.6.0"): | |
from torch.cuda.amp import autocast | |
else: | |
# Nothing to do if torch<1.6.0 | |
def autocast(enabled=True): | |
yield | |
class Emotion2vec(torch.nn.Module): | |
""" | |
Author: Ziyang Ma, Zhisheng Zheng, Jiaxin Ye, Jinchao Li, Zhifu Gao, Shiliang Zhang, Xie Chen | |
emotion2vec: Self-Supervised Pre-Training for Speech Emotion Representation | |
https://arxiv.org/abs/2312.15185 | |
""" | |
def __init__(self, **kwargs): | |
super().__init__() | |
# import pdb; pdb.set_trace() | |
cfg = OmegaConf.create(kwargs["model_conf"]) | |
self.cfg = cfg | |
make_layer_norm = partial( | |
torch.nn.LayerNorm, | |
eps=cfg.get("norm_eps"), | |
elementwise_affine=cfg.get("norm_affine"), | |
) | |
def make_block(drop_path, dim=None, heads=None): | |
return AltBlock( | |
cfg.get("embed_dim") if dim is None else dim, | |
cfg.get("num_heads") if heads is None else heads, | |
cfg.get("mlp_ratio"), | |
qkv_bias=True, | |
drop=cfg.get("encoder_dropout"), | |
attn_drop=cfg.get("attention_dropout"), | |
mlp_drop=cfg.get("activation_dropout"), | |
post_mlp_drop=cfg.get("post_mlp_drop"), | |
drop_path=drop_path, | |
norm_layer=make_layer_norm, | |
layer_norm_first=cfg.get("layer_norm_first"), | |
ffn_targets=not cfg.get("end_of_block_targets"), | |
) | |
self.alibi_biases = {} | |
self.modality_encoders = torch.nn.ModuleDict() | |
enc = AudioEncoder( | |
cfg.modalities.audio, | |
cfg.get("embed_dim"), | |
make_block, | |
make_layer_norm, | |
cfg.get("layer_norm_first"), | |
self.alibi_biases, | |
) | |
self.modality_encoders["AUDIO"] = enc | |
self.ema = None | |
self.average_top_k_layers = cfg.get("average_top_k_layers") | |
self.loss_beta = cfg.get("loss_beta") | |
self.loss_scale = cfg.get("loss_scale") | |
self.dropout_input = torch.nn.Dropout(cfg.get("dropout_input")) | |
dpr = np.linspace( | |
cfg.get("start_drop_path_rate"), | |
cfg.get("end_drop_path_rate"), | |
cfg.get("depth"), | |
) | |
self.blocks = torch.nn.ModuleList( | |
[make_block(dpr[i]) for i in range(cfg.get("depth"))] | |
) | |
self.norm = None | |
if cfg.get("layer_norm_first"): | |
self.norm = make_layer_norm(cfg.get("embed_dim")) | |
vocab_size = kwargs.get("vocab_size", -1) | |
self.proj = None | |
if vocab_size > 0: | |
self.proj = torch.nn.Linear(cfg.get("embed_dim"), vocab_size) | |
def forward( | |
self, | |
source, | |
target=None, | |
id=None, | |
mode=None, | |
padding_mask=None, | |
mask=True, | |
features_only=False, | |
force_remove_masked=False, | |
remove_extra_tokens=True, | |
precomputed_mask=None, | |
**kwargs, | |
): | |
feature_extractor = self.modality_encoders["AUDIO"] | |
mask_seeds = None | |
extractor_out = feature_extractor( | |
source, | |
padding_mask, | |
mask, | |
remove_masked=not features_only or force_remove_masked, | |
clone_batch=self.cfg.get("clone_batch") if not features_only else 1, | |
mask_seeds=mask_seeds, | |
precomputed_mask=precomputed_mask, | |
) | |
x = extractor_out["x"] | |
encoder_mask = extractor_out["encoder_mask"] | |
masked_padding_mask = extractor_out["padding_mask"] | |
masked_alibi_bias = extractor_out.get("alibi_bias", None) | |
alibi_scale = extractor_out.get("alibi_scale", None) | |
if self.dropout_input is not None: | |
x = self.dropout_input(x) | |
layer_results = [] | |
for i, blk in enumerate(self.blocks): | |
if ( | |
not self.training | |
or self.cfg.get("layerdrop", 0) == 0 | |
or (np.random.random() > self.cfg.get("layerdrop", 0)) | |
): | |
ab = masked_alibi_bias | |
if ab is not None and alibi_scale is not None: | |
scale = ( | |
alibi_scale[i] | |
if alibi_scale.size(0) > 1 | |
else alibi_scale.squeeze(0) | |
) | |
ab = ab * scale.type_as(ab) | |
x, lr = blk( | |
x, | |
padding_mask=masked_padding_mask, | |
alibi_bias=ab, | |
) | |
if features_only: | |
layer_results.append(lr) | |
if self.norm is not None: | |
x = self.norm(x) | |
if features_only: | |
if remove_extra_tokens: | |
x = x[:, feature_extractor.modality_cfg.num_extra_tokens :] | |
if masked_padding_mask is not None: | |
masked_padding_mask = masked_padding_mask[ | |
:, feature_extractor.modality_cfg.num_extra_tokens : | |
] | |
return { | |
"x": x, | |
"padding_mask": masked_padding_mask, | |
"layer_results": layer_results, | |
"mask": encoder_mask, | |
} | |
def extract_features( | |
self, source, mode=None, padding_mask=None, mask=False, remove_extra_tokens=True | |
): | |
res = self.forward( | |
source, | |
mode=mode, | |
padding_mask=padding_mask, | |
mask=mask, | |
features_only=True, | |
remove_extra_tokens=remove_extra_tokens, | |
) | |
return res | |
def inference( | |
self, | |
data_in, | |
data_lengths=None, | |
key: list = None, | |
tokenizer=None, | |
frontend=None, | |
**kwargs, | |
): | |
# if source_file.endswith('.wav'): | |
# wav, sr = sf.read(source_file) | |
# channel = sf.info(source_file).channels | |
# assert sr == 16e3, "Sample rate should be 16kHz, but got {}in file {}".format(sr, source_file) | |
# assert channel == 1, "Channel should be 1, but got {} in file {}".format(channel, source_file) | |
granularity = kwargs.get("granularity", "utterance") | |
extract_embedding = kwargs.get("extract_embedding", True) | |
if self.proj is None: | |
extract_embedding = True | |
meta_data = {} | |
# extract fbank feats | |
time1 = time.perf_counter() | |
audio_sample_list = load_audio_text_image_video( | |
data_in, | |
fs=16000, | |
audio_fs=kwargs.get("fs", 16000), | |
data_type=kwargs.get("data_type", "sound"), | |
tokenizer=tokenizer, | |
) | |
time2 = time.perf_counter() | |
meta_data["load_data"] = f"{time2 - time1:0.3f}" | |
meta_data["batch_data_time"] = len(audio_sample_list[0]) / kwargs.get( | |
"fs", 16000 | |
) | |
results = [] | |
output_dir = kwargs.get("output_dir") | |
if output_dir: | |
os.makedirs(output_dir, exist_ok=True) | |
for i, wav in enumerate(audio_sample_list): | |
source = wav.to(device=kwargs["device"]) | |
if self.cfg.normalize: | |
source = F.layer_norm(source, source.shape) | |
source = source.view(1, -1) | |
feats = self.extract_features(source, padding_mask=None) | |
x = feats["x"] | |
feats = feats["x"].squeeze(0).cpu().numpy() | |
if granularity == "frame": | |
feats = feats | |
elif granularity == "utterance": | |
feats = np.mean(feats, axis=0) | |
if output_dir and extract_embedding: | |
np.save(os.path.join(output_dir, "{}.npy".format(key[i])), feats) | |
labels = tokenizer.token_list if tokenizer is not None else [] | |
scores = [] | |
if self.proj: | |
x = x.mean(dim=1) | |
x = self.proj(x) | |
x = torch.softmax(x, dim=-1) | |
scores = x[0].tolist() | |
result_i = {"key": key[i], "labels": labels, "scores": scores} | |
if extract_embedding: | |
result_i["feats"] = feats | |
results.append(result_i) | |
return results, meta_data | |