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  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>Low-rank Adaptation Method for Wav2vec2-based Fake Audio Detection</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Chenglong Wang</string-name>
          <xref ref-type="aff" rid="aff3">3</xref>
          <xref ref-type="aff" rid="aff4">4</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Jiangyan Yi</string-name>
          <xref ref-type="aff" rid="aff1">1</xref>
          <xref ref-type="aff" rid="aff3">3</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Xiaohui Zhang</string-name>
          <xref ref-type="aff" rid="aff2">2</xref>
          <xref ref-type="aff" rid="aff3">3</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Jianhua Tao</string-name>
          <xref ref-type="aff" rid="aff0">0</xref>
          <xref ref-type="aff" rid="aff1">1</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Xinrui Yan</string-name>
          <xref ref-type="aff" rid="aff3">3</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Le Xu</string-name>
          <xref ref-type="aff" rid="aff3">3</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Ruibo Fu</string-name>
          <xref ref-type="aff" rid="aff1">1</xref>
          <xref ref-type="aff" rid="aff3">3</xref>
        </contrib>
        <aff id="aff0">
          <label>0</label>
          <institution>Department of Automation, Tsinghua University</institution>
        </aff>
        <aff id="aff1">
          <label>1</label>
          <institution>School of Artificial Intelligence, University of Chinese Academy of Sciences</institution>
          ,
          <country country="CN">China</country>
        </aff>
        <aff id="aff2">
          <label>2</label>
          <institution>School of Computer and Information Technology, Beijing Jiaotong University</institution>
        </aff>
        <aff id="aff3">
          <label>3</label>
          <institution>State Key Laboratory of Multimodal Artificial Intelligence Systems, Institute of Automation, Chinese Academy of Sciences</institution>
          ,
          <addr-line>Beijng</addr-line>
          ,
          <country country="CN">China</country>
        </aff>
        <aff id="aff4">
          <label>4</label>
          <institution>University of Science and Technology of China</institution>
          ,
          <addr-line>Hefei</addr-line>
          ,
          <country country="CN">China</country>
        </aff>
      </contrib-group>
      <pub-date>
        <year>2023</year>
      </pub-date>
      <fpage>101</fpage>
      <lpage>106</lpage>
      <abstract>
        <p>Self-supervised speech models are a rapidly developing research topic in fake audio detection. Many pre-trained models can serve as feature extractors, learning richer and higher-level speech features. However,when fine-tuning pre-trained models, there is often a challenge of excessively long training times and high memory consumption, and complete fine-tuning is also very expensive. To alleviate this problem, we apply low-rank adaptation(LoRA) to the wav2vec2 model, freezing the pre-trained model weights and injecting a trainable rank-decomposition matrix into each layer of the transformer architecture, greatly reducing the number of trainable parameters for downstream tasks. Compared with fine-tuning with Adam on the wav2vec2 model containing 317M training parameters, LoRA achieved similar performance by reducing the number of trainable parameters by 198 times.</p>
      </abstract>
      <kwd-group>
        <kwd>eol&gt;fake audio detection</kwd>
        <kwd>ASVspoof</kwd>
        <kwd>LoRA</kwd>
        <kwd>self-supervised</kwd>
      </kwd-group>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1. Introduction</title>
      <p>
        information for identifying fake audio. In addition to
wav2vec2, other studies have investigated various
selfSelf-supervised pre-training has become a popular re- supervised models as potential feature extractors for the
search topic in recent years and has already been applied spoof detection task [
        <xref ref-type="bibr" rid="ref9">9</xref>
        ].
in the fields of natural language processing and computer The method of using wav2vec2 as a feature extractor
vision. In the domain of speech processing, numerous typically requires fine-tuning on the training set. This
self-supervised speech models have been proposed, such process requires updating all parameters of the wav2vec2
as wav2vec [
        <xref ref-type="bibr" rid="ref1">1</xref>
        ], wav2vec2[
        <xref ref-type="bibr" rid="ref2">2</xref>
        ], and HuBERT [
        <xref ref-type="bibr" rid="ref3">3</xref>
        ], which en- model to create a new model that contains the same
paables learning of high-level representations of the speech rameters as the original one. This process requires a large
signal. These models have been successfully applied amount of computing resources for training as well as
in various areas, including speech recognition, speaker specialized graphics memory. For example, the Wav2vec2
recognition, and emotion recognition [
        <xref ref-type="bibr" rid="ref4 ref5">4, 5</xref>
        ]. However, in XLSR [
        <xref ref-type="bibr" rid="ref10">10</xref>
        ] contains 317M parameters. In addition,
finethe realm of fake audio detection, only a limited number tuning can easily sufer from catastrophic forgetting and
of studies have explored the use of pre-trained models memory-based overfitting when used with datasets
conas feature extractors. Specifically, some researchers have taining specific tasks [
        <xref ref-type="bibr" rid="ref11">11</xref>
        ].
employed the pre-trained wav2vec2 model, which has In recent years, the research results in the field of
natdemonstrated the ability to extract more robust deep fea- ural language processing (NLP) have developed rapidly,
tures [
        <xref ref-type="bibr" rid="ref6">6</xref>
        ], and has been applied in the recent ADD2022 and various eficient transfer learning methods have
challenge [
        <xref ref-type="bibr" rid="ref7 ref8">7, 8</xref>
        ]. This model takes the raw waveform as emerged, such as adapter-based [
        <xref ref-type="bibr" rid="ref12">12</xref>
        ], prefix-based [
        <xref ref-type="bibr" rid="ref13">13</xref>
        ],
input and learns speech information from a large amount and low-rank adaptation (LoRA) [
        <xref ref-type="bibr" rid="ref14">14</xref>
        ] techniques. These
of unlabeled speech data, potentially containing valuable methods can achieve the most advanced efects with
only partial model parameters trained, avoiding the
probIJCAI 2023 Workshop on Deepfake Audio Detection and Analysis lems brought by full-model tuning. In downstream tasks,
(DADA 2023), August 19, 2023, Macao, S.A.R maintaining static freezing of large-scale pre-trained
lan* Corresponding author. guage models (PLM) parameters, eficient parameter
op† These authors contributed equally. timization methods are used to train only a small part of
j"iancghyeanng.lyoin@gn.wlparn.iga.@acn.clpnr(.iJa..Yaci).c;n21(1C2.0W32a0n@gb);jtu.edu.cn (X. Zhang); additional task-specific parameters, thereby alleviating
jhtao@tsinghua.edu.cn (J. Tao) extreme forgetting [
        <xref ref-type="bibr" rid="ref15">15</xref>
        ] without requiring extra
mem© 2023 Copyright for this paper by its authors. Use permitted under Creative Commons License ory and computing resources. However, these eficient
CPWrEooUrckReshdoinpgs IhStpN:/c1e6u1r3-w-0s.o7r3g ACttEribUutRion W4.0oInrtekrnsahtioonpal (PCCroBYce4.0e).dings (CEUR-WS.org)
      </p>
      <sec id="sec-1-1">
        <title>Attention</title>
        <p>Q
+ LoRA
K
+
LoRA
K
V
+</p>
        <p>LoRA</p>
      </sec>
      <sec id="sec-1-2">
        <title>Hidden States</title>
        <sec id="sec-1-2-1">
          <title>Multi-Head Attention</title>
        </sec>
        <sec id="sec-1-2-2">
          <title>Hidden h</title>
        </sec>
        <sec id="sec-1-2-3">
          <title>Pretrained</title>
        </sec>
        <sec id="sec-1-2-4">
          <title>Weights</title>
          <p>×
∈ ℝ</p>
        </sec>
        <sec id="sec-1-2-5">
          <title>Input X</title>
        </sec>
        <sec id="sec-1-2-6">
          <title>LoRA</title>
          <p>+
d</p>
          <p>
            B
r
A
tuning methods are not systematically studied with self- were inspired by [
            <xref ref-type="bibr" rid="ref14">14</xref>
            ] to adopt a new technique called
supervised models in fake audio detection. Low-rank Adaptation (LoRA), which can reduce
com
          </p>
          <p>
            Inspired by Hu et al. [
            <xref ref-type="bibr" rid="ref14">14</xref>
            ], we propose a low-rank adap- putational resource consumption when fine-tuning the
tive method for pre-trained models used in fake audio de- wav2vec2 model.
tection. By freezing the weights of the pre-trained model To fine-tune the pre-trained weight matrix  ∈ R× ,
and injecting a trainable rank decomposition matrix, we we utilized a low-rank decomposition method to limit its
reduce the number of trainable parameters to address update range. We express the update as  =  + ∆ ,
issues during fine-tuning. We conducted extensive exper- where ∆  = ,  ∈ R× ,  ∈ R× and  is the
iments on the ASVspoof2019 dataset, which showed that rank of the decomposition matrices with  ≪ . During
compared to global fine-tuning, our approach reduced the training process, the parameters of  were fixed and
the number of training parameters by 99.49% with only a did not receive any gradient updates, only the weight
0.17% decrease in performance. Additionally, our method matrices of  and  were trained. It should be noted
reduces the hardware training threshold and increases that both  and ∆  have the same input vector. For
training speed. the original output ℎ =  , we obtained the refined
          </p>
          <p>The main contributions of this study can be summa- output after low-rank decomposition as follows:
rized as follows:
• To our best knowledge, we are the first to apply
the LoRA method to pre-trained models for fake
audio detection.
• The study found that our proposed method can
significantly reduce the number of model
parameters while maintaining high performance and
improving training eficiency.</p>
        </sec>
      </sec>
    </sec>
    <sec id="sec-2">
      <title>2. Proposed Methods</title>
      <p>Fine-tuning a pre-trained model is a common method
used to further improve its performance on a specific
task. However, the requirement that the newly fine-tuned
model has the same number of parameters as the
original model results in a significant increase in computing
resources and higher costs. To address this issue, we
ℎ = ℎ + ∆ 
= ℎ + 
(1)</p>
      <p>
        In this study, we applied the LoRA technique at three
diferent locations of the multi-head attention layer in
the wav2vec2 transformer, specifically at the query, key,
and value vectors in the self-attention module, as shown
in Figure 1. Unlike the fully fine-tuned model, LoRA
introduces no inference latency and roughly converges to
training the original model’s performance within
approximately the same number of training iterations [
        <xref ref-type="bibr" rid="ref14">14</xref>
        ]. This
means that we can significantly reduce the computational
resources consumed during the fine-tuning process by
applying the LoRA technique, while maintaining stable
model performance.
r
      </p>
    </sec>
    <sec id="sec-3">
      <title>3. Experiments</title>
      <p>
        3.1. Dataset
ASVspoof 2019 LA [
        <xref ref-type="bibr" rid="ref16">16</xref>
        ] mainly has 19 spoofing attack
algorithms (A01-A19), including two types of spoofing
attacks: text to speech (TTS) and voice conversion (VC).
      </p>
      <p>
        The LA data set contains three subsets: the training
set, the development set, and the evaluation set.
Table 1 details the number of real and fake audio of the
ASVspoof2019 LA dataset. The training set and
development set mainly include four TTS and two VC algorithms,
namely A01-A06. To better evaluate the performance of
the system, unseen spoofing attacks were added to the
evaluation set, including two known spoofing attacks
(A16 and A19) and 11 unseen spoofing attacks (A07-A15,
A17, and A18).
3.2. Evaluation Metrics
In this work, in order to evaluate the results of
diferent fake audio detection systems, the equal error rate
(EER) is used as the evaluation metrics. Previously, EER
is used in the ASVspoof challenges and ADD 2022
challenge [
        <xref ref-type="bibr" rid="ref16 ref17 ref7">7, 16, 17</xref>
        ]. A real-valued, finite numerical value
is assigned to each trial. It reflects the support for two
competing hypotheses, namely that the trial is a bona
ifde audio or a manipulated one. But we do not optimize
a decision threshold, and thus nor do we produce hard
decisions. High detection score should indicate a genuine
utterance and low score should indicate a manipulated
utterance. The metric in this paper is the ’threshold-free’
EER, defined as follows. Let ( ) and ( ) denote
the false alarm and miss rates at threshold  .
      </p>
      <p>( ) =
( ) =
#{  ℎ  &gt;  }
#{  }</p>
      <p>(2)
#{  ℎ  &lt;  }
#{  }
3.3. Experimental Setup
We first focus on the efect of rank r on model
performance. We found that the number of model parameters is
positively correlated with the value of R, indicating that
the model’s parameter count increases with increasing
r. Surprisingly, the model achieved the best EER of 1.30
when R was set to 4, outperforming the results of the
other three ranks. It is worth noting that even with a very
low value of r, r=2, the model’s EER was still better than
when the model’s parameters were fixed,
demonstrating the efectiveness of combining LoRA with Wav2vec2.</p>
      <p>However, when the rank of LoRA was set to 8 and 16, the
EER results were slightly worse than the rank of 4. This
may be due to injecting a rank that was too large, leading
to an increase in the number of model parameters and
an increase in overfitting.</p>
      <p>
        Table 3 shows the efect of diferent weight types on the
results. We found that, for all weight types and low-rank
matrix ranks, the LoRA model outperforms the baseline
model in terms of EER, indicating that LoRA’s low-rank
matrix adaptation technique can efectively improve the
performance of ASVspoof detection. Furthermore, for
each weight type, as the low-rank matrix rank r increases,
the performance of the LoRA model slightly improves,
but the improvement gradually decreases. This suggests
the paper[
        <xref ref-type="bibr" rid="ref9">9</xref>
        ] presented that “a good self-supervised front that the performance of LoRA is limited by the low-rank
end should be trained with diverse speech data.” The matrix rank, and therefore using a higher rank does not
model’s downsampling factor is 320. Thus, there is a significantly improve performance. Finally, among each
1024-dimensional vector for every 10 ms of speech. weight type, Wq,wv and Wq,wk,wv perform the best,
      </p>
      <p>
        For the backend classifier, we chose light convolution indicating that applying LoRA to the query and value
neural network (LCNN), which is the baseline system for matrices of the Transformer can lead to better
perforASVspoof2019 and 2021. Other settings refer to the paper mance.
[
        <xref ref-type="bibr" rid="ref19">19</xref>
        ]. Table 4 presents the impact of input audio length on
      </p>
      <p>To train the model, we use the Adam optimizer with a model performance. As expected, the ASVspoof
deteclearning rate of 1 × 10−5 . Due to the limitation of GPU tion system exhibits a continuous decrease in EER
performemory, we set the batch size to 16. The model is trained mance with an increase in audio length. Longer speech
for 50 epochs. The training set is used to train the model, signals provide more information, facilitating the
diferthe development set is used to select the model with the entiation of genuine from spoofed speech. The poorest
best performance, and finally, the evaluation set is used EER performance is observed at the shortest input length
for evaluation. of 1 second, with a score of 10.27%. This highlights the
challenge of the ASVspoof detection task in short audio
scenes. In contrast, the EER performance of the ASVspoof
4. Results and Discussion detection system drops to 1.18% when the input length
increases to 8 seconds, approaching the optimal
performance. Combining the findings in Table 6, we conclude
that increasing the audio length can improve the
performance of the ASVspoof detection system, but it also
leads to higher memory requirements during training.</p>
      <p>Therefore, it is crucial to strike a balance between model
performance and training eficiency.</p>
      <p>
        Based on the aforementioned experimental results, we
set the rank r to 2, the length of the input audio to 4
seconds, and apply the LoRA method to the  and 
weight matrices. Adapter [
        <xref ref-type="bibr" rid="ref12">12</xref>
        ] is a method in the field
of Natural Language Processing (NLP) that reduces the
number of trainable parameters during the fine-tuning
process. This paper reproduces the adapter method in
ASVspoof. Table 5 presents the comparison of our
proposed method with fixed parameters, global fine-tuning,
and other local fine-tuning methods. Based on the data in
the table, the following conclusions can be drawn. Firstly,
both global and local fine-tuning can improve the EER
performance of ASVspoof detection task compared to the
method with fixed pre-trained model parameters,
indicating the benefits of task-specific fine-tuning. Secondly,
compared with global and local fine-tuning methods, our
proposed LoRA method significantly reduces the
number of trainable parameters while achieving performance of the Transformer can lead to better performance.
Furcomparable to global fine-tuning on the ASVspoof de- thermore, increasing the audio length can improve the
tection task. This indicates that the LoRA method can performance of the ASVspoof detection system, but it also
maintain high performance while reducing the parame- leads to higher memory requirements during training.
ter count. Finally, compared with global fine-tuning and Finally, LoRA outperforms global fine-tuning in terms of
LoRA, the performance of the Adapter method is slightly training time and memory eficiency, making it a
promislower. This may be because Adapter only updates the ing approach to improve the eficiency and performance
weights of specific layers without updating the weights of the wav2vec2 model for the fake audio detection task.
of other layers, which limits its ability to fully leverage
the advantages of the pre-trained model, and therefore,
performs worse than global fine-tuning and LoRA on the 6. Acknowledgments
ASVspoof detection task. Therefore, we can conclude that
the LoRA method can significantly reduce the number This work is supported by the National Key Research
of model parameters by freezing the pre-trained model’s and Development Plan of China (No.2020AAA0140003),
parameters and injecting trainable low-rank decomposi- the National Natural Science Foundation of China
tion matrices, while maintaining high performance and (NSFC) (No.61831022, No.U21B2010, No.62101553,
improving training eficiency on the ASVspoof detection No.61971419, No.62006223, No.62276259, No.62201572,
task. No. 62206278), Beijing Municipal Science&amp;Technology
      </p>
      <p>Table 6 presents the results of our proposed method Commission,Administrative Commission of
Zhongguanon improving training eficiency, showing that LoRA out- cun Science Park No.Z211100004821013, Open Research
performs global fine-tuning in terms of training time Projects of Zhejiang Lab (NO. 2021KH0AB06).
and memory eficiency. Specifically, for all audio lengths,
LoRA’s training time is significantly shorter than global References
ifne-tuning, and this improvement becomes more
pronounced with increasing batch sizes. For instance, when
the audio length is 1 second, LoRA’s training time only
takes 46 seconds with a batch size of 16 samples, while
global fine-tuning requires 1 minute and 37 seconds.
Additionally, compared to global fine-tuning, LoRA can
handle larger batch sizes without running out of memory.</p>
      <p>For example, when the audio length is 4 seconds, LoRA
can handle a maximum batch size of 8, while global
finetuning runs out of memory with a batch size of 4. Overall,
these results suggest that LoRA is a promising approach
that can improve the training eficiency of the wav2vec2
model and lower the training threshold of hardware.</p>
    </sec>
    <sec id="sec-4">
      <title>5. Conclusion</title>
      <p>In conclusion, the paper proposes a novel low-rank
adaptation method (LoRA) to improve the eficiency and
performance of the wav2vec2 model for the fake audio
detection task. The experimental results show that both global
and local fine-tuning can improve the performance
compared to the fixed pre-trained model parameters.
However, the proposed LoRA method significantly reduces
the number of trainable parameters while achieving
performance comparable to global fine-tuning, indicating
that LoRA can maintain high performance while
reducing the parameter count. Additionally, the paper finds
that the performance of LoRA is limited by the low-rank
matrix rank, and therefore using a higher rank does not
significantly improve performance. Among each weight
type, applying LoRA to the query and value matrices
(2019).</p>
    </sec>
  </body>
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