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  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>TranssionADD: A multi-frame reinforcement based sequence tagging model for audio deepfake detection</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Jie Liu</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Zhiba Su</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Hui Huang</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Caiyan Wan</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Quanxiu Wang</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Jiangli Hong</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Benlai Tang</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Fengjie Zhu</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <aff id="aff0">
          <label>0</label>
          <institution>Department of AI Technology</institution>
          ,
          <addr-line>Transsion</addr-line>
        </aff>
      </contrib-group>
      <pub-date>
        <year>2023</year>
      </pub-date>
      <fpage>113</fpage>
      <lpage>118</lpage>
      <abstract>
        <p>Thanks to recent advancements in end-to-end speech modeling technology, it has become increasingly feasible to imitate and clone a user's voice. This leads to a significant challenge in diferentiating between authentic and fabricated audio segments. To address the issue of user voice abuse and misuse, the second Audio Deepfake Detection Challenge (ADD 2023) aims to detect and analyze deepfake speech utterances. Specifically, Track 2, named the Manipulation Region Location (RL), aims to pinpoint the location of manipulated regions in audio, which can be present in both real and generated audio segments. We propose our novel TranssionADD system as a solution to the challenging problem of model robustness and audio segment outliers in the trace competition. Our system provides three unique contributions: 1) we adapt sequence tagging task for audio deepfake detection; 2) we improve model generalization by various data augmentation techniques; 3) we incorporate multi-frame detection (MFD) module to overcome limited representation provided by a single frame and use isolated-frame penalty (IFP) loss to handle outliers in segments. Our best submission achieved 2nd place in Track 2, demonstrating the efectiveness and robustness of our proposed system.</p>
      </abstract>
      <kwd-group>
        <kwd>eol&gt;Audio Deepfake Detection</kwd>
        <kwd>Manipulation Region Location</kwd>
        <kwd>Anti-spoofing</kwd>
        <kwd>Audio Synthesis Detection</kwd>
      </kwd-group>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1. Introduction</title>
      <p>
        hashing. Neither of these two works is suitable for sit- convolution with filter size and kernel size set to be 256
uations where the forged region is extremely similar to and 3 respectively. After each convolutional layer, batch
its neighbor, such as a segment forged by noise addition normalization and Relu activation are applied.
only. Wu et al. [
        <xref ref-type="bibr" rid="ref24">24</xref>
        ] introduce a location model similar Next, the output of final convolutional layer is fed into
to Question-Answering strategy, which is to make the a single bidirectional LSTM layer with 256 units (128 in
model answer "where are the start and end points" of each direction) to generate classification representations.
anomalous clips. But it makes the problem more am- At the end of the model, we use a linear layer to output
biguous because apparently it is not a natural language the classification probabilities of each frame of audio.
understanding task.
      </p>
      <p>
        To solve these problems, we propose TranssionADD 2.2. Data Augmentation
system, which specifically addresses poor generalization
and outlier problems of the location model. Firstly, we In this study, there are two major dificulties: 1) the data
innovatively convert the task into a sequence tagging is very limited and imbalanced; 2) the training and test
problem, predicting the label of each frame and merg- sets have huge disparities in terms of their domain. The
ing frames with the same label to locate audio segments. model, therefore, sufers from underperformance. To
Secondly, we use RCNN-BLSTM to extract spatial and address this issue, we propose a two-stage data
augmentemporal features frame by frame. To avoid the network’s tation method.
poor learning of contextual information, we use multi- Before-training augmentation. Both the training
frame detection (MFD) module to compress multi-frame set and dev set are insuficient and lack diversity.
Thereinformation. Then we address data insuficiency by apply- fore, we propose three data augmentation methods:
ing data augmentation including voice conversion, pitch
shift, etc. Finally, to solve outliers from isolated frames,
we introduce a penalty strategy called isolated-frame
penalty (IFP) to constrain this situation. Our submission
won second place in the ADD 2023. More explicitly, our
system mainly contributes to the following parts:
• Voice conversion. Use VC [
        <xref ref-type="bibr" rid="ref8">8</xref>
        ] model trained on
given dataset to transform segments of audio and
then annotate adjusted segments as "fake".
• Noise adding. Add noise from MUSAN [
        <xref ref-type="bibr" rid="ref26">26</xref>
        ]
cor
      </p>
      <p>pus to entire audio segments.
• Insertion. Insert a real audio clip with diferent
lengths randomly selected from other audio or
the audio itself.
• We design a novel location model that draws
inspiration from the sequence tagging task,
specifically the Name Entity Recognition (NER) task.
• We utilize data augmentation method both before</p>
      <p>training and during training.
• We apply MFD module to improve robustness and</p>
      <p>add IFP loss to deal with outliers.
• Experiments show our proposed method
per</p>
      <p>forms better than the baseline.</p>
    </sec>
    <sec id="sec-2">
      <title>2. Method</title>
      <p>In this section, we introduce details and rationales of our
proposed method, which mainly consists of four parts: 1)
RCNN-BLSTM backbone; 2) data augmentation; 3)
multiframe detection module; and 4) isolated-frame penalty
loss.</p>
      <sec id="sec-2-1">
        <title>2.1. RCNN-BLSTM Backbone</title>
        <p>
          As shown in Figure 1, we use mel-spectrograms extracted
from audio as input representations, and then we build a
basic model based on residual convolutional neural
network (RCNN) and bi-directional long short-term
memory (BLSTM). Some of our structures refer to the
design of RawNet2 [
          <xref ref-type="bibr" rid="ref21">21</xref>
          ] and reference encoder [
          <xref ref-type="bibr" rid="ref25">25</xref>
          ].
Melspectrograms are passed through 7 layers of 1D residual
In-training augmentation. Due to the fixed nature
of data augmentation before training, it limits the
randomness of the data. Therefore, we propose dynamic
data augmentation during training to ensure suficient
diversity and randomness in the training data.
• Pitch shift. Randomly adjust the pitch of
segments from audio and then label the adjusted
segments as "fake".
• Gaussian noise. Add Gaussian noise to
segments from audio and then label the adjusted
segments as "fake".
        </p>
      </sec>
      <sec id="sec-2-2">
        <title>2.3. Multi-frame Detection Module</title>
        <p>
          RCNN primarily focuses on single-frame information but
lacks a broader context. To further extract more
efective contextual information, we draw inspiration from
wav2vec2.0 [
          <xref ref-type="bibr" rid="ref27">27</xref>
          ] and enhance current single-frame
detection with an additional multi-frame detection (MFD)
module trained in a supervised manner. Multiple
audio frames are passed through a downsampling network
to get their multi-frame feature. The logits of the
average label values corresponding to these multiple audio
frames are calculated to get an estimated label of this
segment. The multi-frame feature is then passed through
        </p>
        <p>Linear
+
repeat</p>
        <p>Linear
BLSTM</p>
        <p>MFD</p>
        <p>X N
x0
x1
x2
x3
x4
...</p>
        <p>x8
x9</p>
        <p>... xn
RCNN encoder</p>
        <p>+
BatchNorm</p>
        <p>Relu
Conv1D</p>
        <p>BatchNorm</p>
        <p>Relu
Conv1D
(stride=2)
BatchNorm</p>
        <p>Relu
Conv1D
(stride=5)
(a) Our model framework
(b) RCNN encoder
(c) MFD
a classification network using the estimated label as the simultaneously calculating the diference between
curtarget. rent frame and its surrounding frames (1 to 3 before and</p>
        <p>Precisely, as shown in Figure 1 , we use two convo- after). Particularly, for frame , its predicted
probabillutional layers to build the downsampling network and ity is ˆ, and we consider the previous and following 
one dense layer to form the classification network. The frames,  ∈ {1, 2, 3} . Correspondingly, we can obtain
iflter size of both convolutional layers is set to 128. The the regularization term .
stride values are set to be 5 and 2; the kernel sizes are
sMetFDto ibsea7ddaendd t3o, rtheseptercatiinvienlyg. lTohsse. cFrionsasl-leyn,twroepryepl oliscsaotef () = ||ˆ − ∑︀=1(ˆ2− + ˆ+) || (1)
and add the outputs of the MFD module to the outputs Finally, we calculate the regularity constraint of total
of BLSTM for each frame, to enhance the classification N frames and divide the sum by 3 as the final regularity
prediction accuracy for each frame. constraint term . The formula is as follows:</p>
      </sec>
      <sec id="sec-2-3">
        <title>2.4. Isolated-frame Penalty Loss</title>
        <p>In traditional NER tasks, one entity corresponds to one =1
or several word(s); while in ADD tasks, one fake/real seg- Then, our total loss involves losses of multiple modules,
ment corresponds to tens or hundreds of frames. Conse- and its expression is as follows:
quently, when applying regular NER to audio frame-level
predictions, a challenge arises wherein the model pre-  =  +   +  *  (3)
dicts labels that appear on a significantly higher number
of discontinuous frames, which we refer to as outliers. Where  and   represent the cross-entropy
Therefore, in order to constrain the case where the pre- loss for the single frame and each MFD module,
respecdicted value is diferent from the surrounding values, we tively, while  serves as the constraint applied to isolated
design an additional loss function called isolated-frame small fragments. It is important to note that the weight
penalty (IFP) loss.  assigned to the IFP loss is a hyperparameter.</p>
        <p>Specifically, we devise a penalty calculation method:

 = ∑︁((1) + (2) + (3))/3
(2)</p>
      </sec>
    </sec>
    <sec id="sec-3">
      <title>3. Experiments</title>
      <sec id="sec-3-1">
        <title>3.1. Datasets</title>
        <p>Our dataset is a publicly available Mandarin corpus
provided by organizers of ADD 2023. Some samples are all
real or all fake, and some are partially fake or partially
real.</p>
      </sec>
      <sec id="sec-3-2">
        <title>3.2. Training Setup</title>
        <sec id="sec-3-2-1">
          <title>3.2.1. Input representations</title>
          <p>We compare mel-spectrograms and MFCCs, and finally
we choose mel-spectrograms as low-level acoustic
representation due to the better performance. We use a 16
kHz sampling rate for all experiments. And window size
of fast Fourier transform (FFT), hop size, and number of
output bins are set to 800, 160, and 80 respectively.</p>
        </sec>
        <sec id="sec-3-2-2">
          <title>3.2.2. Data augmentation</title>
          <p>Before training, we augment the training set and dev
set using techniques such as VC, noise addition, and
insertion. During training, we use a 0.2 random rate to
add Gaussian noise with less than 15 dB signal-to-noise
ratios (SNR) and change the audio pitch by pitch shift.</p>
        </sec>
        <sec id="sec-3-2-3">
          <title>3.2.3. Implementation details</title>
          <p>We first train the model on one NVIDIA 3090 GPU, with
64 batch sizes and 100 epochs. The models are optimized
by Adam with a learning rate of 0.0001 and weight decay
of 10−5 . The hyper-parameter weight  , in the
isolatedframe penalty (IFP) loss, is set to be 0.1 as default, based
on simple tuning experiments.
3.2.4. Metrics
As follow,  is used to measure the model’s
ability to correctly distinguish between genuine and fake
audio, and 1_ is used to measure the model’s
ability to correctly identify fake areas from fake audios.
The final score is the weighted sum of  and
1_.</p>
          <p>= 0.3 *  + 0.7 * 1_
(4)</p>
          <p>Moreover, we consider audio should consist of long
enough consecutive frames with the same label and
isolated frames are possible to be outliers. Therefore, relying
solely on  and 1_ may not adequately
capture the occurrence of fragmented and isolated
segments in audio frame predictions. To evaluate the eficacy
of dealing with outliers, we introduce a simple auxiliary
evaluation metric called "iso-rate," as defined in Eq. 5. If
a segment is shorter than 6 frames (60ms), we classify it
as an isolated segment (), and then compute the
proportion of these isolated segments appearing across
the entire dataset. A lower iso-rate indicates that the
evaluated system has fewer isolated segments and addresses
outliers better.</p>
          <p>iso-rate =


(5)</p>
        </sec>
      </sec>
      <sec id="sec-3-3">
        <title>3.3. Results</title>
        <p>From Table 1 above, our system achieves significantly
higher scores on the development set than on the test
set which may contains more variable and complicated
samples. To address this problem, we introduce the
augmented development set (aug-dev) as an essential
reference to evaluate the model’s performance.</p>
        <sec id="sec-3-3-1">
          <title>3.3.1. Performance</title>
        </sec>
        <sec id="sec-3-3-2">
          <title>3.3.2. Ablation studies</title>
          <p>We also conduct ablation studies to demonstrate the
effectiveness of each part in our method. Specifically, we
conduct individual experiments by removing diferent
components, including before-training augmentation
(BefAug), in-training augmentation (InAug), multi-frame
detection module (MFD), and isolated-frame penalty (IFP).</p>
          <p>From the multi-metric comparison in Table 2, we
observe that the IFP significantly reduces iso-rate while
improving the correct detection score, as evident from
the second and third rows. Additionally, the removal of
MFD module, as indicated in the fourth row, results in
a significant decrease in correct detection score,
indicating its notable contribution, despite limited impact on
iso-rate. Moreover, the fifth and sixth rows demonstrate
that using data augmentation both before-training and
in-training (BefAug &amp; InAug) leads to a significant
improvement in performance, especially on more various
augmented development set and test set.</p>
        </sec>
      </sec>
    </sec>
    <sec id="sec-4">
      <title>4. Conclusion</title>
      <p>In summary, we propose a basic RCNN-BLSTM
backbone to classify each frame of audio. Then we enhance
the stability and generalization ability of our base model
through diverse data augmentation techniques. We also
innovatively design a multi-frame detection (MFD)
module and isolated-frame penalty (IFP) loss to improve the
representation of local features and constrain outliers,
resulting in improved model performance. These three
aspects are crucial for our success in manipulated region
detection challenge. In the future, to further improve
the detection capabilities of our system, we will focus on
enhancing the model’s robustness against noise.</p>
    </sec>
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