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Published in IET Signal Processing
Received on 7th October 2013
Revised on 17th July 2014
Accepted on 26th August 2014
doi: 10.1049/iet-spr.2013.0399

ISSN 1751-9675

Wavelet-domain audio watermarking using optimal

modification on low-frequency amplitude
Shuo-Tsung Chen1,2, Chih-Yu Hsu3, Hunag-Nan Huang2
1
Sustainability Research Center, Tunghai University, Taichung 40704, Taiwan (R.O.C)
2
Department of Applied Mathematics, Tunghai University, Taichung 40704, Taiwan (R.O.C)
3
Department of Information and Communication Engineering, Chaoyang University of Technology, Taichung 413, Taiwan (R.O.C)
E-mail: nhuang@thu.edu.tw

Abstract: On the basis of the Karush–Kuhn–Tucker (KKT) theorem, a novel digital audio watermarking scheme is proposed. To
guarantee the robustness of a watermark, this scheme embeds information into low-frequency coefficients of audio’s discrete
wavelet transform. For the modification of low-frequency amplitude, this study uses the KKT theorem to minimise the
difference between the original and the watermarked coefficients. Accordingly, embedding strength is increased to enhance
the most robustness of a watermarked audio under sufficient embedding capacity and audio quality. In addition, the proposed
watermarking scheme can extract the hidden data without the knowledge of original audio signal. Experimental results
indicate that the performance of the proposed scheme is mostly better than other amplitude modification methods.

1 Introduction embedding strength, the synchronisation codes are not

Digital watermarking is an efficient technique for copyright
protection and information hiding. According to the
application field, the digital watermarking can be classified
into three categories: digital image watermarking, digital
video watermarking and digital audio watermarking.
For the copyright protection of music, many audio
watermarking methods have been proposed recently. These
methods can be grouped into two categories: time-domain
[1–9] and frequency-domain [10–26]. In the time-domain,
different embedding techniques [1–3] are first applied to
digital watermarking. Lie and Chang [6] adopted
low-frequency amplitude modification to enhance
robustness. However, it has an extremely low capacity. This
is because they use three segments (length = 1020 points) to
present 1 bit. Xiang and Huang [7] proposed a
histogram-based audio watermarking which used the
invariant feature of audio histogram to design a robust
method against time scaling. Peng and Wang [9] proposed
an optimal audio watermarking scheme using genetic
optimisation with variable-length mechanism. The presented
genetic optimisation procedure can automatically determine
optimal embedding parameters for each audio frame of an
audio signal and the employed variable-length mechanism
can effectively search most suitable positions for watermark
embedding. However, the computing time is large since
generic algorithm is a kind of searching method and the
obtained optimal solution is not verified.
In the frequency-domain, Huang et al. [10] embeds the
watermark into discrete cosine transform (DCT)
coefficients, and hides the Bark code in the time-domain as
synchronisation codes. Since the time-domain has low
robust enough. However, if the synchronisation codes are
embedded in DCT, then the computation cost increases.
Wu et al. [11] used single-coefficient quantisation index
modulation method to embed information into
low-frequency coefficients in discrete wavelet transform
(DWT). Their method achieves robustness against common
signal processing procedure and noise corruption. However,
it is very vulnerable to re-sampling, amplitude and time
scaling. Some authors improved this method by using
multi-coefficient quantisation [12–14, 24, 25]. The methods
of multi-coefficient quantisation are still vulnerable to
amplitude and time scaling. According to the audio
statistics characteristics and synchronisation code technique,
Wang et al. [16, 17] gave a robust audio watermarking
scheme in wavelet domain. However, they only embed the
synchronisation code by modifying individual sample value,
which reduces the resisting ability greatly so that their
method is vulnerable to the attacks: re-sampling, amplitude
scaling and time scaling. To improve these two methods,
Wang et al. [23] proposed a robust digital audio
watermarking scheme using wavelet moment invariance.
Chen and Huang [18] and Chen et al. [19, 26] proposed a
novel audio watermarking that embeds information by
energy-proportion scheme. Their scheme can resist some
attacks, especially amplitude scaling. However, the
watermarked audio has low quality and is vulnerable to
time scaling. To enhance both signal-to-noise ratio (SNR)
and robustness of the single-coefficient quantisation
method, Chen et al. [20] proposed the wavelet-domain
audio watermarking scheme using optimisation-based
quantisation which is robust to common signal processing
such as re-sampling, MP3 compression, amplitude scaling etc.

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To achieve high robustness in time-domain, authors usually the Vk above. The orthogonal relations follow the existence
use large number of samples to represent a binary bit. of sequences h = {hk }k[Z and g = {gk }k[Z that satisfy the
However, such an embedding technique results in low following identities
capacity [6, 7]. This paper utilises low-frequency √


coefficients in DWT to embed the watermark. To modify hk = kw0, 0 , w−1, k l and w(t) = 2 hk w(2t − k)
these low-frequency coefficients, this paper uses the k[Z
Karush–Kuhn–Tucker (KKT) theorem to minimise the √


difference between the original and the watermarked gk = kc0, 0 , w−1, k l and c(t) = 2 gk w(2t − k)
coefficients in the matrix form. Accordingly, embedding k[Z
strength can be increased to enhance robustness under
sufficient audio quality and embedding capacity. In where h = {hk }k[Z and g = {gk }k[Z denote low-pass and
addition, the proposed watermarking scheme can extract the high-pass filters, respectively. Throughout this paper, the
hidden data without the help of original audio signal. host digital audio signal S(n), n [ N denotes the samples of
Finally, the performance is evaluated by embedding the original audio signal S(t) at the nth sample time. This
capacity, SNR, perceptual evaluation of audio quality paper adopts orthogonal Haar wavelet bases to implement
(PEAQ), mean opinion score and bit error rate (BER). DWT on the host digital audio signal S(n) through the filter
Simulation results verify the efficiency of the proposed bank [27].
scheme.
The rest of this paper is organised as follows. Overview of 2.2 Optimisation solver
DWT and mathematical clarification of the KKT theorem are
presented in Section 2. Section 3 uses the KKT theorem to To find the extreme of a matrix function, an optimisation
propose a novel scheme of watermark embedding and solver called the KKT theorem is summarised [28, 29].
detecting. Some experiments are used to test the Suppose that CN = [c1, c2, …, cN]T is an N × 1 matrix
performance of the proposed watermarking scheme in (vector). The problem of minimising a performance index f
Section 4. Conclusions are finally drawn in Section 5. (CN) subject to constraints gi (CN) ≤ 0, i = 1, 2,…, m can be
described as
2 Mathematical clarifications
minimise f (CN ) (3a)
To develop an optimisation-based scheme for audio
subject to gi (CN ) ≤ 0, i = 1, 2, . . . , m (3b)
watermarking, DWT and the KKT theorem for later use are
reviewed in this section.
To solve (3), the KKT theorem is introduced as follows.
KKT theorem: Let f and gi, i = 1, 2,…, m be convex and
2.1 DWT continuously differentiable
  functions and there exists a
point CN0 such that gi CN0 , 0, i = 1, 2, . . . , m. Then CN∗
Wavelets are obtained using a single prototype function, c(x), is a global optimal solution to minimise f(CN) subject to
which is regulated by the scaling and shift parameters. The constraints gi (CN) < 0, i = 1, 2,…, m, if and only if there
discrete normalised scaling and wavelet basis function are exists a m × 1 vector l = [l1, l2, …, lm] such that
defined as
    ∗
(1) ∇f CN∗ + m i=1 li ∇gi CN = 0,
w j, k (t) = 2 j/2 hj (2j t − k) (1) ∗
(2) gi CN ≤ 0, i = 1, 2, . . . , m,
(3) l
i ≥ 0, i = ∗1, 2, . . . , m and
c j, k (t) = 2 j/2 gj (2j t − k) (2) (4) m i=1 li gi CN = 0.

A construction of two subspaces In the above theorem, we refer to l as the KKT multiplier
vector and conditions (1)–(4) as the KKT condition.
Vj = span{w j, k : k [ Z} Specially, when m = 1 the KKT multiplier l is a scalar and
the KKT condition becomes
and        
∇f CN∗ + l∇g CN∗ = 0, g CN∗ ≤ 0, l ≥ 0, lg CN∗ = 0
Wj = span{c j, k : k [ Z}
And hence the global solution CN∗ can be calculated either by
follows where j and m refer to the dilation and translation (corresponding to the case l = 0)
parameters. Moreover, it is a necessity that the subspaces    
∇f CN∗ = 0 with g CN∗ ≤ 0 (4)
{0} , · · · , V1 , V0 , V−1 , · · · , L (R) 2
or (corresponding to another case l > 0)
2
form a multi-resolution analysis of L (R) and the subspaces      
…,W1, W0, W−1, …stand for the orthogonal differences of ∇f CN∗ + l∇g CN∗ = 0 with g CN∗ = 0 (5)

Fig. 1 Structure of synchronisation codes and watermarks

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If we let
H(CN , l) = f (CN ) + lg(CN ) (6)
then solving (5) becomes to minimise the function H(CN, l)
without any constraint which leads to the optimal solution
computed by
∂H ∂H
=0 and =0 (7)
∂l ∂CN
3 Proposed watermarking scheme
To achieve high robustness in time-domain, method [11] uses
large number of samples to represent a binary bit. However,
such an embedding technique results in low capacity. To
guarantee the robustness of the watermark under high
capacity, this paper embeds information into low-frequency
coefficients in DWT. Generally, the quality of a
watermarked audio is measured by SNR. To have the best
quality under the embedding constraint, this section rewrites
SNR as a performance index. Then the difficulty of
modifying consecutive coefficients is converted into an
optimisation problem. Finally, the KKT theorem is applied
to solve this problem, and thus an optimisation-based
watermarking formula is obtained.
Fig. 2 Watermark embedding procedure
168
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3.1 Adaptive threshold and embedding process
Since the watermarked audio may suffer the attack of shifting
or cropping, it is necessary to embed the synchronisation
codes together. These synchronisation codes are used to
locate the positions where the watermark is embedded. The
structure is shown in Fig. 1. Before embedding, the
synchronisation codes and watermark are arranged into a
binary sequence, j = {ji|ji = 1 or 0}. To determine the
adaptive threshold for embedding and extracting the binary
sequence, original audio signal with length L is firstly cut
into I segments and then H-level DWT is performed on
each segment. Accordingly, the total number of
lowest-frequency coefficients in each segment is K =
L/(I·2H) and the mean of these coefficients is
1 K
m= |c | (8)
K i=0 i
The two thresholds are computed adaptively as
h1 = m + 1 (9)
and
h0 = m − 1 (10)
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doi: 10.1049/iet-spr.2013.0399

where ε > 0 is an embedding strength. Moreover, the
parameters m and ε for each segment are used as additional
keys for the watermark embedding and extraction.
In the proposed embedding process, the DWT
lowest-frequency coefficients of each audio segment are
partitioned into several groups and each group contains N
consecutive coefficients. Let
 T
CN = |c1 | |c2 | ··· |cN |
Then, the binary bit of value 0 or 1 is embedded by the
following rules
† AĈN ≥ h1 , if ji = 1
† AĈN ≤ h0 , if ji = 0
where ĈN is watermarked wavelet coefficient vector
corresponding to CN and A = [a1, …, aN] is the scaling
Fig. 3 Watermark extraction procedure
IET Signal Process., 2015, Vol. 9, Iss. 2, pp. 166–176
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vector whose entries are positive and can be arbitrarily
assigned by an encoder. To avoid the situation that the
value of some entries becomes arbitrarily large, without loss
of generality we may set the summation of all the scaling
factors
 equal to a  constant N. For example,
A = 0.9 1.2 1.2 0.7 is a suitable selection when
N = 4. With the suitable scaling vector A, we consider the
optimisation problem for watermarking is to select the
vector ĈN such that the SNR is maximised under the
(11)
constraint (12) or (13). The solution of the vector ĈN will
be derived optimally in next section.
3.2 Optimisation-based watermarking formula
(12) In general, the quality of the watermarked audio is evaluated
by SNR which is defined as follows
(13)
S(n) 2
SNR = 10 log10
2
(14)
Ŝ(n) − S(n) 2
2
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or


Ŝ(n) − S(n) 2
SNR = −10 log10 2 (15)
S(n) 2
2

where S(n) and Ŝ(n) denote the original and the modified
audio signal. Another consideration of SNR in (15) is


Ĉ − C 2
N N 2
SNR = −10 log10 2 (16)
C
N 2

since we use orthogonal wavelet basis. For optimisation, we

modify (16) as a performance index

Ĉ − C 2
N N
Fig. 4 Relationship between SNR and the scaling factor a2 for 2 (17)
C
popular N

or

(ĈN − CN )T (ĈN − CN )
(18)
CNT CN

Assuming we are embedding the watermark with the binary

bit of value 1, by connecting (17) with (12) we have the
following corresponding optimisation problem

(ĈN − CN )T (ĈN − CN )
minimise (19a)
CNT CN

subject to AĈN ≥ h1 (19b)

Fig. 5 Relationship between SNR and the scaling factor a2 for

piano

Fig. 7 Relationship between SNR and the scaling factor a2 for

symphony

Table 1 Subjective difference grades

SDG Qualities Impairments

0.0 excellent imperceptible

−1.0 good perceptible but not annoying
−2.0 fair slightly annoying
−3.0 poor annoying
Fig. 6 Relationship between SNR and the scaling factor a2 for −4.0 bad very annoying
singing

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Table 2 Embedding domain, embedding capacity, SNR and PEAQ
Each audio’s Each audio’s Average SNR, dB and the Average PEAQ and the
embedding domain embedding capacity corresponding variance corresponding variance

reference paper time-domain low 500 bits/11.6 s 22.1|2.62 (popular) −1.03|0.66 (popular)
[6] frequency 15.7|1.43 (piano) −2.53|0.56 (piano)
17.4|1.42 (singing) −1.68|0.52 (singing)
20.3|2.53 (symphony) −1.22|0.62 (symphony)
reference paper DWT 7-level 1333 bits/11.6 s 20.5|2.83 (popular) −2.26|2.77 (popular)
[19] 17.1|2.27 (piano) −2.47|1.78 (piano)
17.6|2.14 (singing) −2.15|0.73 (singing)
17.2|2.68 (symphony) −2.04|1.89 (symphony)
proposed DWT 7-level 2000 bits/11.6 s 24.1|0.13 (popular) −1.01|0.24 (popular)
method (N = 2) 20.3|0.08 (piano) −1.56|0.22 (piano)
21.8|0.04 (singing) −1.02|0.18 (singing)
20.2|0.06 (symphony) −1.26|0.13 (symphony)
proposed DWT 7-level 1000 bits/11.6 s 23.5|0.12 (popular) −1.03|0.09 (popular)
method (N = 4) 19.8|0.06 (piano) −1.13|0.06 (piano)
21.3|0.04 (singing) −1.06|0.06 (singing)
20.2|0.04 (symphony) −1.22|0.08 (symphony)

whose optimal solution can be computed via the KKT Since CNT CN is a constant, H(ĈN , l) can be rewritten as
theorem with m = 1. Thus, there are two possible solutions follows
by using (4) or (7). From (4), one of the optimal solution is  
given by ĈN∗ = CN such that AĈN∗ = ACN ≥ h1 , that is, no H(ĈN , l) = (ĈN − CN )T (ĈN − CN ) + lCNT CN AĈN − h1
coefficient adjustment is needed to embedding the binary (21)
bit of value ‘1’. If the condition ACN ≥ h1 does not hold,
then we seek for another solution via (7). Set l denote the The necessary conditions for existence of the minimum of
multiplier to combine (19a) and (19b) to form an H(ĈN , l) are
unconstrained matrix function
∂H
= 2(ĈN − CN ) + lAT CNT CN = 0 (22)
∂ĈN
(ĈN − CN ) (ĈN − CN )
T  
H(ĈN , l) = + l AĈN − h1 (20) ∂H
CNT CN = AĈN − h1 = 0 (23)
∂l

Table 3 Re-sampling
Music types Re-sampling rate, Hz 96 000 22 050 11 025 8000 6000

popular reference paper [6] BER, % mean 0.28 0.32 0.34 0.41 0.29
variance 0.02 0.03 0.02 0.06 0.08
reference paper [19] BER, % mean 4.22 3.12 4.52 5.13 5.61
variance 0.12 0.04 0.22 0.62 0.44
proposed method (N = 2) BER, % mean 0.04 0 0 0.08 0.21
variance 0.01 0 0 0 0.03
proposed method (N = 4) BER, % mean 0.03 0 0 0 0.06
variance 0 0 0 0 0
piano reference paper [6] BER, % mean 0.80 0.31 0.92 1.62 2.13
variance 0.06 0.04 0.04 0.06 0.04
reference paper [19] BER, % mean 5.26 5.26 6.81 7.43 7.26
variance 0.22 0.12 0.38 0.46 0.42
proposed method (N = 2) BER, % mean 0.29 0.23 0.65 1.14 1.68
variance 0.04 0.01 0.02 0.04 0.04
proposed method (N = 4) BER, % mean 0.12 0 0 0.15 0.42
variance 0.01 0 0 0.02 0.02
singing reference paper [6] BER, % mean 0.22 0.28 0.30 0.54 0.56
variance 0.01 0.02 0.02 0.04 0.04
reference paper [19] BER, % mean 6.93 6.92 11.3 12.3 12.5
variance 0.14 0.06 0.06 0.24 0.32
proposed method (N = 2) BER, % mean 0.02 0 0 0.12 0.02
variance 0 0 0 0.02 0.01
proposed method (N = 4) BER, % mean 0 0 0 0 0
variance 0 0 0 0 0
symphony reference paper [6] BER, % mean 0.12 0.28 0.31 0.48 0.50
variance 0.03 0.01 0.01 0.02 0.01
reference paper [19] BER, % mean 5.74 4.81 9.92 9.98 10.7
variance 0.16 0.32 0.29 0.58 0.62
proposed method (N = 2) BER, % mean 0.12 0.14 0.42 0.44 0.46
variance 0.04 0.01 0.02 0.02 0.02
proposed method (N = 4) BER, % mean 0.04 0 0.05 0.07 0.13
variance 0 0 0 0.01 0

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Table 4 MP3 compression Table 5 Low-pass filter
Bit rate, kbps 128 112 96 80 Cut-off frequency, kHz 3 5

popular reference mean 0.25 0.31 1.56 4.12 popular reference paper [6] BER, % mean 17.4 16.8
paper [6] BER, variance 0.01 0.02 0.02 0.04 variance 0.02 0.02
% reference paper [19] BER, % mean 25.1 24.3
reference mean 0.38 1.68 2.46 5.21 variance 0.64 0.52
paper [19] variance 0.04 0.12 0.12 0.34 proposed method (N = 2) mean 5.96 2.26
BER, % BER, % variance 0.01 0.02
proposed mean 0.06 0.28 1.74 3.12 proposed method (N = 4) mean 6.96 2.32
method (N = 2) variance 0 0.02 0.04 0.03 BER, % variance 0.01 0.01
BER, % piano reference paper [6] BER, % mean 19.9 19.3
proposed mean 0.04 0.24 1.39 2.86 variance 0.03 0.02
method (N = 4) variance 0 0.01 0.01 0.02 reference paper [19] BER, % mean 24.6 24.2
BER, % variance 0.44 0.46
piano reference mean 0.28 0.43 2.92 4.33 proposed method (N = 2) mean 6.95 1.78
paper [6] BER, variance 0.02 0.02 0.04 0.12 BER, % variance 0.02 0.02
% proposed method (N = 4) mean 6.84 1.98
reference mean 0.45 1.92 2.88 5.43 BER, % variance 0.02 0.01
paper [19] variance 0.04 0.22 0.24 0.21 singing reference paper [6] BER, % mean 18.2 17.3
BER, % variance 0.01 0.02
proposed mean 0.08 0.42 1.92 3.56 reference paper [19] BER, % mean 31.9 29.8
method (N = 2) variance 0.02 0.04 0.02 0.02 variance 0.46 0.44
BER, % proposed method (N = 2) mean 5.12 1.98
proposed mean 0.07 0.31 1.86 3.32 BER, % variance 0.01 0.01
method (N = 4) variance 0 0.01 0.02 0.01 proposed method (N = 4) mean 5.06 2.04
BER, % BER, % variance 0.02 0.01
singing reference mean 0.08 0.11 2.01 2.88 symphony reference paper [6] BER, % mean 20.3 20.1
paper [6] BER, variance 0 0.02 0.02 0.04 variance 0.02 0.04
% reference paper [19] BER, % mean 28.2 27.5
reference mean 0.32 0.98 1.42 2.24 variance 0.32 0.31
paper [19] variance 0.04 0.04 0.05 0.08 proposed method (N = 2) mean 7.16 2.04
BER, % BER, % variance 0.01 0.01
proposed mean 0.06 0.18 1.49 2.11 proposed method (N = 4) mean 7.34 2.28
method (N = 2) variance 0 0.01 0.02 0.02 BER, % variance 0.01 0.01
BER, %
proposed mean 0.03 0.12 1.18 2.36
method (N = 4) variance 0 0.01 0.02 0.02
BER, % coefficients is
symphony reference mean 0.22 0.28 0.96 1.12
paper [6] BER, variance 0.01 0.03 0.04 0.04
% ĈN∗ = CN − AT (AAT )−1 (ACN − h1 ) (28)
reference mean 0.37 2.37 3.32 5.79
paper [19] variance 0.02 0.26 0.54 0.52
BER, % where the superscript * denotes the optimal result with respect
proposed mean 0.08 0.24 0.58 0.58 to the corresponding variable. On the basis of (28), the binary
method (N = 2) variance 0.01 0.01 0.02 0.01 bit ‘1’ can be embedded by the optimal modified coefficients
BER, % ĈN∗ . In other words, the binary bit ‘0’ is embedded by using h0
proposed mean 0.01 0.10 0.12 0.26
method (N = 4) variance 0 0.01 0.02 0.02
instead of h1
BER, %
ĈN∗ = CN − AT (AAT )−1 (ACN − h0 )

Multiply (22) by A, we observe that

Fig. 2 shows the proposed embedding procedure.
 
2 AĈN − ACN + lAAT CNT CN = 0 (24)
3.3 Extraction process
Since When extracting the hidden information, we split the test
audio into several segments and then perform DWT on
AĈN = h1 (25) each segment in the same manner as in the embedding
process. Let ĈN∗ be the optimally watermarked coefficient
vector
 ∗ which has N consecutive absolute values
from (23), and CNT CN is a constant, (24) is rewritten as
2(h1 − ACN ) + lCNT CN AAT = 0
Hence the optimal solution for parameter l is
ĉ , i = 1, . . . , N . We extract the binary sequence j by
i
using the following rules:
(26)
† If AĈN∗ ≥ h1 , the extracted bit j̃i = 1.
† If AĈN∗ ≤ h0 , the extracted bit j̃i = 0.

where h0 and h1 are obtained by using (9) and (10) with two
2
l∗ = T (AAT )−1 (ACN − h1 ) (27) stored keys m and ε for each audio segment. After finding the
CN CN synchronisation code, we continue to apply the above
extraction rules to detect the watermark. The extraction
Replacing (27) to (22), the optimal solution of modified procedure is shown in Fig. 3.

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3.4 Relationship between SNR and scaling factors
To understand the effect of adjusting scaling factors on the
SNR, we consider the case with N = 4 and the scaling
vector to be A = a1 a2 1 1 , that is, a1 + a2 = 2.
Figs. 4–7 show the relationship between the SNR and the
scaling factor a2 for audio popular and piano. As the
scaling factor a2 is decreasing from one, the SNR still
maintains about 21 and 15 dB. It indicates that the variance
of the scaling factors has no obvious effect on SNR. Hence,
we  randomly set  the scaling vector to
A = 1.4 0.6 1.6 0.4 in the experiments in next
section.
4 Experimental results
This section applies the proposed watermarking scheme to
four types of 16 bit mono audios, namely popular, piano,
singing and symphony music, sampled at 44 100 Hz. Each
type has 12 audio files. The length of each audio is about
11.6 s. In the embedding process, the embedding strength ε
is set to 2000 and 4000 for N = 2 and N = 4. The
performance of the proposed watermarking scheme is
evaluated by embedding capacity, SNR, PEAQ and BER.
4.1 Embedding domain and capacity
Before embedding the watermark, each audio is cut into four
segments. By applying 7-level DWT to each segment of the
original audio, we embed the synchronisation code and
watermark into the lowest-frequency sub-band. Since the
watermark is embedded into DWT level seven, the
embedding capacity of each segment is calculated as
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44 100 × 11.609 ÷ 4 ÷ 27 ÷ 2 = 500 bits for N = 2 and 44
100 × 11.609 ÷ 4 ÷ 27 ÷ 4 = 250 bits for N = 4. Accordingly,
the audio’s embedding capacity is 2000 and 1000 bits for
N = 2 and N = 4.
4.2 SNR and PEAQ
The watermarked audio quality is evaluated mathematically
by SNR defined in (15) or (16). To evaluate the
watermarked audio quality practically, we performed an
objective listening test by using PEAQ which is a
standardised algorithm for objectively measuring perceived
audio quality [30–32]. The analyses of the algorithm results
are based on the subjective difference grade (SDG). The
output values of the algorithm are mapped to SDG by an
artificial neural network with one hidden layer. It compares
the audio signal under test with the original reference audio
signal. The SDG indicates the measured basic audio quality
of the signal under test on a continuous scale from −4
(very annoying impairment) to 0 (imperceptible
impairment). The meaning of each score in SDG is listed in
Table 1. The results are listed in Table 2. We can see that
SNR of our method is mostly 20 dB under high embedding
capacity. On the basis of the sufficient quality of the
watermarked audio, the proposed optimisation-based
scheme has higher embedding capacity than the methods
proposed by Lie and Chang [6] and Chen et al. [19].
4.3 Robustness tests
Various attacks were used to test the robustness, classified
into seven types: (i) re-sampling (ii) MP3 compression (iii)
low-pass filtering (iv) amplitude scaling (v) time scaling

Table 6 Amplitude scaling

Amplification, % 0.8 0.9 1.1 1.2

popular reference paper [6] BER, % mean 0.54 0.54 0.42 0.58
variance 0.02 0.01 0.03 0.03
reference paper [19] BER, % mean 0.45 0.43 0.34 0.38
variance 0.01 0 0 0
proposed method (N = 2) BER, % mean 7.98 6.17 2.88 1.96
variance 0.12 0.04 0.04 0.03
proposed method (N = 4) BER, % mean 29.6 25.4 0.36 0.36
variance 0.21 0.28 0.02 0.01
piano reference paper [6] BER, % mean 0.79 0.79 0.63 0.65
variance 0.03 0.04 0.04 0.04
reference paper [19] BER, % mean 0.43 0.35 0.32 0.38
variance 0.01 0 0 0
proposed method (N = 2) BER, % mean 15.8 12.6 3.22 2.52
variance 0.33 0.22 0.16 0.06
proposed method (N = 4) BER, % mean 34.7 29.4 0.43 0.45
variance 0.22 0.24 0.12 0.08
singing reference paper [6] BER, % mean 0.62 0.64 0.54 0.55
variance 0.01 0.01 0 0
reference paper [19] BER, % mean 0.07 0.07 0.07 0.07
variance 0 0 0 0
proposed method (N = 2) BER, % mean 8.22 6.33 3.11 2.09
variance 0.22 0.08 0.12 0.08
proposed method (N = 4) BER, % mean 29.6 23.2 0.18 0.18
variance 0.36 0.29 0.05 0.04
symphony reference paper [6] BER, % mean 0.72 0.71 0.58 0.55
variance 0.13 0.02 0.01 0.01
reference paper [19] BER, % mean 0.41 0.32 0.24 0.33
variance 0.01 0 0 0.01
proposed method (N = 2) BER, % mean 16.1 14.3 2.92 2.77
variance 0.24 0.31 0.26 0.22
proposed method (N = 4) BER, % mean 31.2 27.5 0.44 0.58
variance 0.44 0.46 0.02 0.02

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Table 7 Time scaling other two approaches by Lie and Chang [6] and Chen et al.
[19]. The results of testing these seven attacks are shown as
Time scaling −5 −2 2 5
amount, % follows:

popular reference paper mean 38.9 40.3 38.7 38.8 (1) Re-sampling: The sampling rate of the watermarked audio
[6] BER, % variance 0.24 0.23 0.05 0.04 was down-sampled from 44 100 to 22050 Hz, 8000 and
reference paper mean 41.4 41.1 40.3 41.4 6000 Hz and then back to 44 100 Hz using interpolation.
[19] BER, % variance 0.08 0.04 0.06 0.05 Similarly, the sampling rate of the watermarked audio was
proposed mean 41.9 40.2 39.4 43.4
method (N = 2) variance 0.12 0.03 0.02 0.04 up-sampled from 44 100 to 96 000 Hz and then back to 44
BER, % 100 Hz. Table 3 shows the results of these re-sampling
proposed mean 40.7 40.1 39.2 41.4 processes for three approaches, which indicate that the
method (N = 4) variance 0.06 0.03 0.02 0.04 proposed scheme is more robust than the methods proposed
BER, %
piano reference paper mean 43.9 43.5 45.2 45.8
by Lie and Chang [6] and Chen et al. [19].
[6] BER, % variance 0.24 0.22 0.03 0.13 (2) MP3 compression: MP3 compression is the
reference paper mean 46.2 45.1 44.2 46.3 conventionally adopted audio compression method. Table 4
[19] BER, % variance 0.32 0.21 0.13 0.23 presents the results of applying MP3 compression at
proposed mean 42.8 43.2 42.4 44.7 different bit rates to the watermarked audio. The proposed
method (N = 2) variance 0.14 0.05 0.06 0.07
BER, % method was found to have better robustness MP3
proposed mean 42.1 42.1 42.3 42.4 compression attack than the methods proposed by Lie and
method (N = 4) variance 0.11 0.03 0.03 0.06 Chang [6] and Chen et al. [19].
BER, % (3) Low-pass filtering: Table 5 shows the effect of using a
singing reference paper mean 41.6 40.1 39.2 39.8
[6] BER, % variance 0.05 0.04 0.02 0.03
low-pass filter with a cutoff frequency of 3 and 5 kHz. The
reference paper mean 45.6 43.7 42.5 43.1 proposed method has higher robustness against the low-pass
[19] BER, % variance 0.22 0.02 0.04 0.29 filter attack than the methods proposed by Lie and Chang
proposed mean 42.8 41.2 39.9 44.5 [6] and Chen et al. [19].
method (N = 2) variance 0.02 0.01 0.01 0.02 (4) Amplitude scaling: An amplitude scaling attack was
BER, %
proposed mean 41.4 40.1 39.4 41.5 performed on the watermarked audio with scaling factors
method (N = 4) variance 0.01 0.01 0.01 0.02 0.5, 0.8, 1.1 and 1.2. Table 6 shows the experimental
BER, % results of this attack. Since the method by Lie and Chang
symphony reference paper mean 42.9 42.3 40.6 41.9 [6] uses the embedding technique that modify the audio
[6] BER, % variance 0.21 0.12 0.05 0.07
reference paper mean 42.4 40.3 40.2 41.3
samples or DWT coefficients by the same scales in a
[19] BER, % variance 0.43 0.32 0.21 0.29 section of a group. Hence, their method is more robust
proposed mean 43.4 42.4 40.6 42.1 against amplitude scaling attacks than ours. In addition, the
method (N = 2) variance 0.02 0.01 0.01 0.02 method proposed by Chen et al. [19] has high robustness
BER, % against amplitude scaling attacks.
proposed mean 42.7 41.1 39.7 40.4
method (N = 4) variance 0.02 0.01 0.01 0.01 (5) Time scaling: The watermarked audios were scaled by
BER, % −5, −2, 2 and 5%. Table 7 presents the analytical results of
this scaling. The proposed method was found to have similar
robustness with the methods proposed by Lie and Chang [6]
and Chen et al. [19].
(vi) echo and (vii) Gaussian noise. The robustness is (6) Echo: An echo was performed on the watermarked audios
measured by BER which is defined as with delay 0.5 ms, volume −6 dB and feedback − 60 dB.
The proposed method has BER about 10–15% which is
Berror
BER = × 100% much better than methods proposed by Lie and Chang [6]
Btotal and Chen et al. [19]. Table 8 shows the experimental
results of this attack.
where Berror and Btotal denote the number of error bits and the (7) Gaussian noise: Table 9 lists the experimental results of
number of total bits. Since we adopt 12 different songs in each adding Gaussian noise to the audio signal. Our method has
category, as piano, popular, symphony and singing, the test better performance than the methods proposed by Lie and
results for robustness are illustrated in their mean and Chang [6] and Chen et al. [19].
variance. We also compare the proposed scheme with the

Table 8 Echo The results in Tables 3–9 also indicate that the proposed
watermarking method is robust against most attacks under
Audio type Popular Piano Singing Symphony
higher embedding capacity.
reference mean 59.3 61.1 54.2 56.8
paper [6] variance 0.78 0.75 0.56 0.63
BER, % 5 Conclusions
reference mean 32.4 32.7 33.9 30.4
paper [19] variance 0.33 0.29 0.35 0.32 This paper presents an optimisation-based watermarking
BER, %
proposed mean 13.9 14.9 12.8 14.6 scheme in wavelet domain. We apply the KKT theorem to
method (N variance 0.26 0.04 0.02 0.23 minimise the difference between the original and the
= 2) BER, % watermarked coefficients. Accordingly, the
proposed mean 10.6 12.4 10.3 12.8 lowest-frequency amplitude can be modified optimally.
method (N variance 0.17 0.05 0.08 0.16
= 4) BER, %
Moreover, we also illustrate the relationship between SNR
and scaling factor. The relationship indicates that the

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& The Institution of Engineering and Technology 2015 doi: 10.1049/iet-spr.2013.0399
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Table 9 Gaussian noise
Amplitude of noise, dB −40 dB −30 dB −20 dB −15 dB

popular reference paper [6] BER, % mean 9.32 16.2 38.8 41.4
variance 0.22 0.43 0.46 0.53
reference paper [19] BER, % mean 4.95 8.16 33.4 37.4
variance 0.21 0.34 0.56 0.63
proposed method (N = 2) BER, % mean 0 0 26.4 28.1
variance 0 0 0.24 0.21
proposed method (N = 4) BER, % mean 0 0 23.5 24.2
variance 0 0 0.18 0.22
piano reference paper [6] BER, % mean 8.34 15.6 39.2 42.5
variance 0.12 0.34 0.34 0.32
reference paper [19] BER, % mean 4.11 7.23 31.8 34.2
variance 0.12 0.14 0.23 0.36
proposed method (N = 2) BER, % mean 0 0 24.5 25.2
variance 0 0 0.22 0.24
proposed method (N = 4) BER, % mean 0 0 21.4 23.6
variance 0 0 0.11 0.22
singing reference paper [6] BER, % mean 9.74 18.3 38.5 42.1
variance 0.24 0.24 0.36 0.34
reference paper [19] BER, % mean 3.26 6.83 29.1 31.6
variance 0.16 0.32 0.44 0.54
proposed method (N = 2) BER, % mean 0 0 20.9 21.3
variance 0 0 0.31 0.32
proposed method (N = 4) BER, % mean 0 0 20.4 21.2
variance 0 0 0.24 0.21
symphony reference paper [6] BER, % mean 11.3 19.4 40.1 42.8
variance 0.56 0.45 0.52 0.51
reference paper [19] BER, % mean 4.83 8.47 34.2 38.9
variance 0.08 0.12 0.35 0.72
proposed method (N = 2) BER, % mean 0 0 22.9 23.3
variance 0 0 0.27 0.32
proposed method (N = 4) BER, % mean 0 0 21.4 22.2
variance 0 0 0.33 0.28

variance of the scaling factors has no obvious effect on SNR.
The experimental results show that the embedded data are
robust against most signal processing and attacks, except
for amplitude scaling, under sufficient audio quality and
embedding capacity. In the future work, we will focus on
the improvement of robustness to amplitude scaling by
improving the embedding rules or adding a corresponding
constraint in the optimisation problem.
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