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IJSRD - International Journal for Scientific Research & Development| Vol. 1, Issue 5, 2013 | ISSN (online): 2321-0613
All rights reserved by www.ijsrd.com 1144
Abstract-- Digital watermarking is the process of hiding
information in a carrier signal. An audio watermarking is a
kind digital watermarking over which the carrier signal is
audio. Here in this paper, an advanced audio watermarking
scheme called additive interpolation-error expansion
technique is proposed. The paper proposes a successful
method for embedding and extraction of binary bits and text
in audio wav file.
Keywords: Multi-Embedding, Input-Flexibility
I. INTRODUCTION
Reverse data hiding is the process of hiding a data which is
normally called as a watermark. The extraction process
should recover the original cover signal along with the
watermark. The watermark may be perceptible or
imperceptible in nature. For an audio signal, watermarking
is typically done in order to identify ownership of copyright
for that audio. The watermark should do in a way that it is
difficult to remove. The watermark information is also
copied as the audio is being copied.
A practically useful reversible watermark scheme should
meet the requirements [1]: (1) Robustness, (2)
Imperceptibility, (3) Readily Embedding and Retrieving.
Also it has to be (a) blind, (b) Higher embedding capacity.
The reversible watermarking techniques can of three classes
[2]: (I) Reversible watermarking scheme by applying data
compression, (II) Reversible watermarking scheme by using
histogram bin shifting, (III) Reversible watermarking
schemes by using difference expansion.
Interpolation is done in order to increase the size. Here in
this paper, a reversible audio watermarking algorithm using
additive interpolation-error expansion is proposed for the
embedding and extraction of binary bits and text.
II. ADDITIVE INTERPOLATION-ERROR EXPANSION
APPROACH
This section recalls the additive interpolation-error
expansion presented in [3].
A. Embedding Algorithm
The audio wav file has to quantize into q bits and hence the
sample values must be in the range [0,2q
– 1].
The proposed algorithm was implemented in MATLAB.
First, read any audio wav file. By using any interpolator,
find the interpolation of the wav file. Let x be the original
wav file and x‟ be its interpolation values. Then the
interpolation-errors are calculated using:
e=x-x‟ (1)
Next is to calculate LM and RM using:
LM= arg eєE max hist(e) (2)
RM= arg eєE-[LM] max hist(e) (3)
Where LM and RM denote the corresponding values of the
two highest points of interpolation-error histogram, arg
denotes the position and E denotes the set of interpolation-
errors. The explanation for (2) and (3) is given as follows.
For calculating LM, first find the histogram of (1). The
proposed way is to find the unique values of e and then find
the position having the maximum histogram value. The
value in that position in the unique value array of e gives
LM. For finding RM value, first delete all the LM values
from (1) and then precede the same procedure for LM to get
(3). Any other methods can also be used for calculating the
histogram. In most of the cases LM is 0 and RM is equal to
1.
The next step is to split the error array into two different
arrays. The splitting is purely based on the LM and RM
values. The splitting is as follows.
Left interpolation-errors (LE): e<=LM (4)
Right interpolation-errors (RE): e>=RM (5)
From the newly developed arrays, find two more parameters
LN and RN which are defined as
LN=arg eєLE min hist(e) (6)
RN=arg eєRE min hist(e) (7)
The procedure for finding LN and RN can be done by
following the same way for finding (2) and (3). Usually LN
is a smaller integer that with no interpolation-error
satisfying e = LN and RN is a larger integer with no
interpolation-error satisfying e = RN.
After finding LM, RM, LN, RN it is possible to define
additive interpolation-error expansion.
E‟={
( )
( ) ( ) ( ) (8)
e‟ is the expanded interpolation-error, b is the embedded bit
and sign(e) is sign function which is given as
sign (e)={ (9)
The actual bit embedding process is carried out in this stage.
While implementing, (9) should be defined before (8).
The actual watermarked wav sample x” is given as
x” = x‟+ e‟ (10)
In order to overcome the overflow/underflow issue, the
below given algorithm should be used. Overflow/underflow
issue arises when samples are changed from 2q
-1 to 2q
or
from 0 to -1.
Algorithm 1
Map = [];
For first_sample to last_sample
if x”=0 V x”=2q
-1
Map= [Map; 0];
Reversible Digital Watermarking of Audio Wav Signal Using Additive
Interpolation-Error Expansion
Anand Baby Alias1
Minu Kuriakose2
1
Student, M. Tech 2
Assistant Professor
1,2
Electronics and Communication Department, FISAT
Reversible Digital Watermarking of Audio Wav Signal Using Additive Interpolation-Error Expansion
(IJSRD/Vol. 1/Issue 5/2013/027)
All rights reserved by www.ijsrd.com 1145
elseif x”= -1
Map = [Map; 1];
x”= 0;
elseif x” = 2q
Map = [Map; 1];
x” = 2q
-1
end
end
The map values will only be generated in case of
overflow/underflow happens.
B. Extraction Algorithm
Before the actual data extraction steps begins a reverse of
the above given algorithm has to be implemented in order to
reverse the truncation produced by that algorithm
Algorithm 2
Map_counter=1;
for first_sample to last_sample
if x”=0 V x”=2q
-1
if Map(Map_counter)=0
Map_counter=Map_counter+1;
elseif Map(Map_counter)=1
If x”=0
x”=-1
elseif x”=2q
-1
x”=2q
end
Map_counter=Map_counter + 1;
end
end
end
Algorithm 1 and Algorithm 2 are given above directly in the
MATLAB format.
Now compute the interpolation values of x” by using the
same interpolator used in the embedded stage. The newly
obtained interpolation values is x‟. Then compute e‟using:
e‟=x”-x‟ (11)
For extracting the embedded bits
b = { (12)
For get back the original interpolation-errors,
e= {
( ) [ ] [ ]
( ) [ ) ( ]
(13)
It should be checked that (13) and (1) be same.
For getting back the original audio samples
x = x‟+ e (14)
It should be noted that the above embedding algorithm is to
embed a single binary bit, either 1 or 0. So for embedding
multiple binary bits, multi-embedding approach is explored.
This can be done by make use of „for loop‟.
III. INPUT FLEXIBILITY APPROACH
The above explained algorithm works only if the number of
bits which is to be embedding should be equal to or greater
than the number of input wav array. So in order to overcome
this limitation, Input Flexibility Approach is used. The idea
is that, after embedding the exact input bits, default values-
NaN are given to all the other embedding spaces.
IV. EMBEDDING OF TEXT
The approach is that before the text files are being
embedding to the actual implementation code, all the
characters are converted into its respective binary values and
then after extracting, all the binary values are converted
back to characters. The above idea can be adapted by using
„de2bin‟and „bi2de‟command.
V. RESULTS
A null difference between the original and restored wav is
obtained. A sample wav is taken and the results obtained are
shown below:
Fig. 1: Original wav and wav after bit embedding
Fig. 2: wav after bit extraction
The null-difference between the embedded wav and the
original wav is obtained as:
Fig. 3: Original wav – Recovered wav
The null-difference between the bits embedded and
extracted bits is obtained as:
Reversible Digital Watermarking of Audio Wav Signal Using Additive Interpolation-Error Expansion
(IJSRD/Vol. 1/Issue 5/2013/027)
All rights reserved by www.ijsrd.com 1146
Fig. 4: Embedded bits – Extracted bits
VI. CONCLUSION
In this paper, a reversible audio watermarking scheme by
exploring additive interpolation-error expansion is proposed.
Method for generation of compact boundary maps is
proposed. Multi-embedding alone with Input Flexibility
Approach is also explained. The same implementation
procedure is then expanded for the embedding of text. The
generated MATLAB code was undergone test for around
hundreds of input wav file. So it is clear that more research
is needed to understand the exact characteristics of the input
wav file for multi-embedding approach.
ACKNOWLEDGEMENT
We would like to extend heartful and sincere gratitude to
Jose Juan Garcia-Hernandez, the author of “Using additive
interpolation-error expansion for reversible digital
watermarking in audio signals”, 2012 IEEE, for his
technical and moral guidance. His paper on “Using additive
interpolation-error expansion for reversible digital
watermarking in audio signals” was very insightful which
acted as the backbone structure for this paper.
VII. REFERENCES
[1]. Jen-Bang Feng, Iuon-Chang Lin, Chwei-Shyong Tsai,
and Yen-Ping Chu, “Reversible Watermarking:
Current Status and Key Issues”, International Journal
of Network Security, Vol.2, No.3, PP.161-171, May
2006
[2]. Mikko Loytnoja, Nedeljko Cvejic, Tapio Seppanen,
“Audio Protection with Removable Watermarking”,
2007 IEEE
[3]. Jose Juan Garcia-Hernandez and Lorenzo Delgado-
Guillen, “Using additive interpolation-error
expansion for reversible digital watermarking in
audio signals”, 2012 IEEE

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Reversible Digital Watermarking of Audio Wav Signal Using Additive Interpolation-Error Expansion

  • 1. IJSRD - International Journal for Scientific Research & Development| Vol. 1, Issue 5, 2013 | ISSN (online): 2321-0613 All rights reserved by www.ijsrd.com 1144 Abstract-- Digital watermarking is the process of hiding information in a carrier signal. An audio watermarking is a kind digital watermarking over which the carrier signal is audio. Here in this paper, an advanced audio watermarking scheme called additive interpolation-error expansion technique is proposed. The paper proposes a successful method for embedding and extraction of binary bits and text in audio wav file. Keywords: Multi-Embedding, Input-Flexibility I. INTRODUCTION Reverse data hiding is the process of hiding a data which is normally called as a watermark. The extraction process should recover the original cover signal along with the watermark. The watermark may be perceptible or imperceptible in nature. For an audio signal, watermarking is typically done in order to identify ownership of copyright for that audio. The watermark should do in a way that it is difficult to remove. The watermark information is also copied as the audio is being copied. A practically useful reversible watermark scheme should meet the requirements [1]: (1) Robustness, (2) Imperceptibility, (3) Readily Embedding and Retrieving. Also it has to be (a) blind, (b) Higher embedding capacity. The reversible watermarking techniques can of three classes [2]: (I) Reversible watermarking scheme by applying data compression, (II) Reversible watermarking scheme by using histogram bin shifting, (III) Reversible watermarking schemes by using difference expansion. Interpolation is done in order to increase the size. Here in this paper, a reversible audio watermarking algorithm using additive interpolation-error expansion is proposed for the embedding and extraction of binary bits and text. II. ADDITIVE INTERPOLATION-ERROR EXPANSION APPROACH This section recalls the additive interpolation-error expansion presented in [3]. A. Embedding Algorithm The audio wav file has to quantize into q bits and hence the sample values must be in the range [0,2q – 1]. The proposed algorithm was implemented in MATLAB. First, read any audio wav file. By using any interpolator, find the interpolation of the wav file. Let x be the original wav file and x‟ be its interpolation values. Then the interpolation-errors are calculated using: e=x-x‟ (1) Next is to calculate LM and RM using: LM= arg eєE max hist(e) (2) RM= arg eєE-[LM] max hist(e) (3) Where LM and RM denote the corresponding values of the two highest points of interpolation-error histogram, arg denotes the position and E denotes the set of interpolation- errors. The explanation for (2) and (3) is given as follows. For calculating LM, first find the histogram of (1). The proposed way is to find the unique values of e and then find the position having the maximum histogram value. The value in that position in the unique value array of e gives LM. For finding RM value, first delete all the LM values from (1) and then precede the same procedure for LM to get (3). Any other methods can also be used for calculating the histogram. In most of the cases LM is 0 and RM is equal to 1. The next step is to split the error array into two different arrays. The splitting is purely based on the LM and RM values. The splitting is as follows. Left interpolation-errors (LE): e<=LM (4) Right interpolation-errors (RE): e>=RM (5) From the newly developed arrays, find two more parameters LN and RN which are defined as LN=arg eєLE min hist(e) (6) RN=arg eєRE min hist(e) (7) The procedure for finding LN and RN can be done by following the same way for finding (2) and (3). Usually LN is a smaller integer that with no interpolation-error satisfying e = LN and RN is a larger integer with no interpolation-error satisfying e = RN. After finding LM, RM, LN, RN it is possible to define additive interpolation-error expansion. E‟={ ( ) ( ) ( ) ( ) (8) e‟ is the expanded interpolation-error, b is the embedded bit and sign(e) is sign function which is given as sign (e)={ (9) The actual bit embedding process is carried out in this stage. While implementing, (9) should be defined before (8). The actual watermarked wav sample x” is given as x” = x‟+ e‟ (10) In order to overcome the overflow/underflow issue, the below given algorithm should be used. Overflow/underflow issue arises when samples are changed from 2q -1 to 2q or from 0 to -1. Algorithm 1 Map = []; For first_sample to last_sample if x”=0 V x”=2q -1 Map= [Map; 0]; Reversible Digital Watermarking of Audio Wav Signal Using Additive Interpolation-Error Expansion Anand Baby Alias1 Minu Kuriakose2 1 Student, M. Tech 2 Assistant Professor 1,2 Electronics and Communication Department, FISAT
  • 2. Reversible Digital Watermarking of Audio Wav Signal Using Additive Interpolation-Error Expansion (IJSRD/Vol. 1/Issue 5/2013/027) All rights reserved by www.ijsrd.com 1145 elseif x”= -1 Map = [Map; 1]; x”= 0; elseif x” = 2q Map = [Map; 1]; x” = 2q -1 end end The map values will only be generated in case of overflow/underflow happens. B. Extraction Algorithm Before the actual data extraction steps begins a reverse of the above given algorithm has to be implemented in order to reverse the truncation produced by that algorithm Algorithm 2 Map_counter=1; for first_sample to last_sample if x”=0 V x”=2q -1 if Map(Map_counter)=0 Map_counter=Map_counter+1; elseif Map(Map_counter)=1 If x”=0 x”=-1 elseif x”=2q -1 x”=2q end Map_counter=Map_counter + 1; end end end Algorithm 1 and Algorithm 2 are given above directly in the MATLAB format. Now compute the interpolation values of x” by using the same interpolator used in the embedded stage. The newly obtained interpolation values is x‟. Then compute e‟using: e‟=x”-x‟ (11) For extracting the embedded bits b = { (12) For get back the original interpolation-errors, e= { ( ) [ ] [ ] ( ) [ ) ( ] (13) It should be checked that (13) and (1) be same. For getting back the original audio samples x = x‟+ e (14) It should be noted that the above embedding algorithm is to embed a single binary bit, either 1 or 0. So for embedding multiple binary bits, multi-embedding approach is explored. This can be done by make use of „for loop‟. III. INPUT FLEXIBILITY APPROACH The above explained algorithm works only if the number of bits which is to be embedding should be equal to or greater than the number of input wav array. So in order to overcome this limitation, Input Flexibility Approach is used. The idea is that, after embedding the exact input bits, default values- NaN are given to all the other embedding spaces. IV. EMBEDDING OF TEXT The approach is that before the text files are being embedding to the actual implementation code, all the characters are converted into its respective binary values and then after extracting, all the binary values are converted back to characters. The above idea can be adapted by using „de2bin‟and „bi2de‟command. V. RESULTS A null difference between the original and restored wav is obtained. A sample wav is taken and the results obtained are shown below: Fig. 1: Original wav and wav after bit embedding Fig. 2: wav after bit extraction The null-difference between the embedded wav and the original wav is obtained as: Fig. 3: Original wav – Recovered wav The null-difference between the bits embedded and extracted bits is obtained as:
  • 3. Reversible Digital Watermarking of Audio Wav Signal Using Additive Interpolation-Error Expansion (IJSRD/Vol. 1/Issue 5/2013/027) All rights reserved by www.ijsrd.com 1146 Fig. 4: Embedded bits – Extracted bits VI. CONCLUSION In this paper, a reversible audio watermarking scheme by exploring additive interpolation-error expansion is proposed. Method for generation of compact boundary maps is proposed. Multi-embedding alone with Input Flexibility Approach is also explained. The same implementation procedure is then expanded for the embedding of text. The generated MATLAB code was undergone test for around hundreds of input wav file. So it is clear that more research is needed to understand the exact characteristics of the input wav file for multi-embedding approach. ACKNOWLEDGEMENT We would like to extend heartful and sincere gratitude to Jose Juan Garcia-Hernandez, the author of “Using additive interpolation-error expansion for reversible digital watermarking in audio signals”, 2012 IEEE, for his technical and moral guidance. His paper on “Using additive interpolation-error expansion for reversible digital watermarking in audio signals” was very insightful which acted as the backbone structure for this paper. VII. REFERENCES [1]. Jen-Bang Feng, Iuon-Chang Lin, Chwei-Shyong Tsai, and Yen-Ping Chu, “Reversible Watermarking: Current Status and Key Issues”, International Journal of Network Security, Vol.2, No.3, PP.161-171, May 2006 [2]. Mikko Loytnoja, Nedeljko Cvejic, Tapio Seppanen, “Audio Protection with Removable Watermarking”, 2007 IEEE [3]. Jose Juan Garcia-Hernandez and Lorenzo Delgado- Guillen, “Using additive interpolation-error expansion for reversible digital watermarking in audio signals”, 2012 IEEE