1680 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 23, NO.
10, OCTOBER 2015
Reduction of Gaussian, Supergaussian, and Impulsive
Noise by Interpolation of the Binary Mask Residual
Marco Ruhland, Joerg Bitzer, Member, IEEE, Matthias Brandt, and Stefan Goetze, Member, IEEE
AbstractIn this paper, we present a new approach for noise re-
duction. A binary timefrequency (T-F) masking threshold crite-
rion is proposed and analyzed with respect to the average spectra of
music and noise disturbances. Modied autoregressive (AR) detec-
tion and AR interpolation are then applied to the residual signal of
the binary masking process. The proposed method is able to reduce
supergaussian and impulsive noise while ensuring preservation of
the desired signal, which is crucial for professional high-quality
audio restoration, and it is also suitable for Gaussian noise to a cer-
tain extent. The approach is compared to a state-of-the-art restora-
tion algorithm by means of the objective measures signal-to-noise
ratio (SNR) improvement and perceptual quality, and by subjec-
tive listening tests. The objective results as well as the listening
tests show that the proposed algorithm is especially suited for su-
pergaussian, grainy-sounding noise types, e.g., optical soundtrack
noise of celluloid movie footage, or rain noise.
Index TermsInterpolation, noise reduction, optical soundtrack
noise, timefrequency masking.
I. INTRODUCTION
T HE term noise reduction is often associated with the re-
moval or reduction of Gaussian noise disturbances. The
assumption of a Gaussian random process is common for pop-
ular denoising algorithms, like the well-known Wiener lter [1]
or the Ephraim-Malah method [2] developed in the 1980s. How-
ever, at the same time, Porter and Boll [3] showed that speech
signals are rather characterized by leptokurtic, respectively su-
pergaussian amplitude distributions, and that the error intro-
duced by the decient Gaussian assumption may be signicant.
This nding led to the development of more sophisticated ap-
proaches, e.g., [4] by Cohen, featuring two-sided Gamma- and
Laplace densities for the speech amplitude distributions, how-
ever still assuming a Gaussian distribution for the noise. Only
a few months later Martin [5] showed that car noise does rather
have a Laplacian distribution instead of a Gaussian distribution,
Manuscript received October 14, 2014; revised February 18, 2015; accepted
May 29, 2015. Date of publication June 11, 2015; date of current version June
19, 2015. This work was supported in part by the German Federal Ministry of
Education and Research (BMBF) under Grants 17N3008 and 03FH030PX2 and
in part by the EU-FP7 project EcoShopping under Grant 609180. The associate
editor coordinating the review of this manuscript and approving it for publica-
tion was Prof. DeLiang Wang.
M. Ruhland and S. Goetze are with the Project Group Hearing, Speech and
Audio Technology, Fraunhofer Institute for Digital Media Technology (IDMT),
D-26129 Oldenburg, Germany (e-mail: marco.ruhland@idmt.fraunhofer.de).
J. Bitzer and M. Brandt are with the Institute for Hearing Technology and
Audiology, Jade University of Applied Sciences, 26121 Oldenburg, Germany.
Color versions of one or more of the gures in this paper are available online
at http://ieeexplore.ieee.org.
Digital Object Identier 10.1109/TASLP.2015.2444664
2329-9290 2015 IEEE. Personal use is permitted, but republication/redistribution requires IEEE permission.
See http://www.ieee.org/publications_standards/publications/rights/index.html for more information.
and he showed how to employ this prior information within
the minimum mean square error (MMSE) estimator for a de-
noising algorithm. This approach works particularly well at low
signal-to-noise ratios (SNR) below 0 dB since the Laplacian
noise assumption yields less musical noise in that case. Fur-
ther renement of the estimators is presented in [6], [7]. Besides
these methods, several techniques exist to reduce the musical
noise phenomenon, e.g., spectral peak elimination [8], spectral
weighting [9], spectral domain smoothing [10], or smoothing of
the spectral gain function [11]. For speech signals, it has been
shown that cepstral smoothing is superior to spectral domain
smoothing, since speech-relevant information like pitch and for-
mant structure can be protected against smoothing within the
cepstral domain [12][14].
The previously described denoising methods usually work in
the frequency-domain. However, for the removal of impulsive
disturbances, e.g., clicks caused by dust and scratches on
a gramophone disc, time-domain interpolation methods are
preferable. Since click disturbances are mostly single, sparse
events in time, lasting only a few milliseconds, it is usually
sufcient to replace the corrupted samples by interpolated
values of the surrounding unaffected samples [15]. While
frequency-based methods usually affect all samples of a signal
block, better preservation of the desired signal is achieved
by time-domain interpolation, since only a few samples are
changed. Pioneering work on time-domain interpolation has
been done independently by Vaseghi [16] and Veldhuis [17] for
different applications.
However, in some cases, a strict distinction between impulse
disturbance and hiss cannot be made easily. Imagine the noise
of a heavy rainfall or applause which is the result of a vast
number of small impulsive events per time instant, and cannot
be regarded as single and sparse any more. Though the audible
sensation is stationary, a certain granularity is perceived that
allows the listener to identify the impulsive origin of the noise.
Similar noise can be observed when listening to the sound of
old celluloid movies of the optical soundtrack era. The footage
decomposes with time and suffers from dust and mould, badly
affecting the audio signal which is encoded in an optical sound-
track next to the picture information. The resulting noise is
grainy and of supergaussian amplitude distribution. Broadcast
media archives request for new restoration techniques to cope
with such kinds of degradation. For an overview of restoration
of optical soundtracks please cf. [18]. This kind of noise is prob-
lematic for both frequency- as well as time-domain approaches.
In this contribution we propose a new hybrid algorithm, per-
forming frequency-domain binary masking and time-domain
RUHLAND et al.: REDUCTION OF GAUSSIAN, SUPERGAUSSIAN AND IMPULSIVE NOISE 1681

interpolation to tackle the described problem. Comparing our samples at a sampling frequency Hz,
method with a state-of-the-art restoration algorithm will give with samples block shift, corresponding to 50%
new ideas on how to cope with supergaussian noise. overlap, by
The remainder of this contribution is organized as follows.
Section II introduces our approach, being a combination of time- (3)
frequency (T-F) masking and AR detection and interpolation.
In Section III, a brief summary of the theoretical analysis of the
proposed algorithm is given based on a former contribution of In Eq. (3), denotes the frequency index
the authors [19]. Then, the proposed algorithm is tested versus and the discrete frame index. The frequency-domain target
a state-of-the-art algorithm by means of objective quality mea- signal and the residual signal are initialized with
sures and subjective listening tests in Sections IV and V, respec- zeros. The absolute squared magnitude is compared
tively. Finally, conclusions are drawn in Section VI. to the binary masking threshold estimate for each fre-
quency bin . The threshold estimate can be initialized
II. THEORY with zeros before processing the rst block , or, for
faster initial response, with the squared magnitude of the rst
A. Binary Masking signal block, . If the squared magnitude for a certain
A degraded audio signal at discrete time index is given frequency bin is above , the respective fre-
as a set of sinusoidal basis functions plus a random process quency bin is copied into the target STFT signal
(cf. [20], [21]): . Otherwise, it is copied into the residual STFT
signal block . Finally, the inverse discrete Fourier
transform (IDFT) is used to obtain the time-domain signal
(1) blocks and of the target signal

For declicking applications, the task is to estimate the complex (4)

coefcients in Eq. (1) and the number of sinusoids, that
minimize the energy of the residual , e.g., in a least-squares
and the residual signal
sense (cf. [21], [20] or [22]). The residual is then treated
as an autoregressive (AR) process, and AR detection and inter-
polation is performed within to eliminate the clicks. After- (5)
wards, the interpolated residual is added back to the split-apart
sinusoids, to result in the restored audio signal . 1) Proposed BM threshold estimate: The above split-
An easier way to split the sinusoids from the residual can ting method has already been used in [24], however, with a
be achieved by T-F masking. The use of T-F masking for sepa- single-value threshold for all frequency bins . In order to obtain
ration tasks is quite young, though the idea behind it is quite a frequency dependent threshold for the binary mask, we use
simple and of low complexity. It has been successfully em-
ployed e.g., in computational auditory scene analysis (CASA), (6)
independent component analysis (ICA), improvement of au-
tomatic speech recognition (ASR), and cochlear implant (CI) where the power spectral density (PSD) estimate of the input
signal enhancement, amongst other elds of use. A list of con- signal is calculated by a recursively smoothed peri-
tributions on the use of T-F masking within these elds of re- odogram
search is given in [23], pg. 342. For a two-source separation (7)
task, like in our case, the separation of a desired target signal
from a noisy residual signal, the T-F masking process is also In Eq. (7), the smoothing vector is dened as
called binary masking (BM).
Let be the noisy music signal to be re- (8)
stored, as the sum of the undisturbed signal , and the noise
disturbance . The aim of the process is to nd a target signal with
and a residual signal that match the true unknown de-
sired signal and the noise signal as closely as possible. (9)
Although and are different from and , Eq. (2)
and
shall be satised,

(2) (10)

expressing Eq. (1) in a simpler fashion. In Eq. (2), repre- In Eqs. (9) and (10), we use time constants of s,
sents the so-called target signal, as the sum of sinusoids, and and s. The relatively short release time en-
the noisy residual signal. Several steps have to be per- sures that the threshold estimate follows the noise oor quickly,
formed to obtain these signals. First, the noisy signal is whereas the high attack time helps to preserve the threshold.
transformed into the frequency-domain, using a DFT of length This ensures that short transients, e.g., drum sounds, and other
1682 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 23, NO. 10, OCTOBER 2015
Fig. 1. Block diagram of the BM process. The noisy mixture
signal containing mainly the noise. By comparing the target and residual spectrograms on the right side with the clean and noise spectrograms on the left
side, it is obvious that the BM process does not separate the signals perfectly. Further processing is required for high quality audio restoration results.
short-time instrumental events will not change the threshold too
fast. The result is, that, after some seconds of adaptation time,
the threshold signal gives a robust estimate of the noise
PSD of the disturbed signal. By applying the binary masking
procedure, the target signal frames will contain the
main content of the music, and the residual will con-
tain the noise as visualized in Fig. 1. However, some low-level
parts of the music, that fall below the threshold , e.g.,
reverb tails, will remain in the residual signal. Finally, in
Eq. (6) is needed to improve the threshold estimation and serves
as a frequency dependent overestimation factor. This xed real
valued parameter compensates for the fact that music signals
follow a -characteristic [25], i.e., that the long-term power
spectrum of music shows a decaying slope of 3 dB per octave,
or 10 dB per frequency decade. Since most impulsive noise
disturbances are characterized by a at spectrum with high en-
ergy also in the upper frequency range, there is an increasing
offset in the estimate over frequency. The lack of energy
of music in the high frequency range will cause the threshold
estimate to be too low in that range. For compensation,
the decay parameter is needed, being
This invokes a slope of dB/dec. to be added to the
threshold estimate. Intuitively, if music entails a slope of
10 dB/dec., a value of about 5 dB/dec. for will help to
adjust the threshold properly (assuming 0 dB/dec. for the noise
disturbance).
Fig. 1 shows an example of a BM process. By comparing
the target and residual spectrograms on the right side with the
clean and noise spectrograms on the left side, it is obvious that
the BM process does not separate the signals perfectly. Musical
noise is present in the target signal, and the residual might still
contain some components of the desired signal. Please note, that
listening examples are available at [26]. Hence, for high quality
audio restoration, it is mandatory to apply further processing,
is split into a target signal holding the desired components of the signal, and a residual
preferably within the residual signal, and nally recombining
target and residual to form the restoration result.
2) Ideal Binary Mask: Besides other BM estimation methods
that exist in the literature (cf. e.g. [27]), the ideal binary mask
(IBM) was proposed in [28] and is formulated in [23] in detail.
As described in [27], it is obtained by comparing the true instan-
taneous SNR with a xed threshold (typically 0 dB) for each T-F
bin. According to [29] and others, the IBM offers the highest
SNR improvement considering the target signal versus the de-
graded signal. As a matter of course, the true instantaneous SNR
is unknown for real-world restoration tasks. For comparison we
will show IBM results as well in Section IV as a quality level
for the objective measurements.
B. Autoregressive Detection
As already stated in Section I, several noise types with super-
gaussian distribution, such as e.g. rain noise, can be seen as a
vast number of small impulses per time instant. The task of an
impulse detection algorithm is to localize impulse disturbances
in time with high accuracy since a low false alarm rate of such
an algorithm helps to preserve the unaffected parts within the
desired signal. As shown in [30], the AR method works best in
(11)
terms of missing detection rate and lowest false alarm rate com-
pared to other methods. This detection method originally has
been introduced in [31], [32] and is recommended in standard
audio restoration literature like [15], [20]. Here, we use AR de-
tection and interpolation within the residual signal. An AR coef-
cient estimate can be obtained from a signal block
using e.g. the Yule-Walker or the Burg method [33]. It is shown
in [30] that the AR model order usually can be quite low
(as long as is fullled) for reliable detection results.
We choose . An error signal is then calcu-
lated by applying the well-known AR prediction error formula
(12)
RUHLAND et al.: REDUCTION OF GAUSSIAN, SUPERGAUSSIAN AND IMPULSIVE NOISE 1683

on the residual block , with being the estimated

AR coefcients. For the case that no impulsive event is present
in , the prediction error will be low. But as
soon as an impulse occurs within , the mist in pre-
diction leads to an error signal of high magnitude. Thus, as sug-
gested in the literature mentioned above, impulsive disturbances
can be detected by thresholding the squared error ,
leading to a binary detection sequence . However,
instead of thresholding the squared error signal we
introduce a xed detection percentage to select a xed number
of samples in each processed block. If , no samples of the
signal block shall be detected as impulsive and, thus
no interpolation will follow. On contrary, if , all of the
samples within the block shall be treated as impulsive (this of
course makes no sense, since the subsequent AR interpolator
would not work without any non-impulsive samples remaining
in the block). If is set to a value between 0 and 1, e.g.,
, then only 10% of the samples of the block shall be
detected as impulsive, namely those 10% that have the highest
squared error . This is done by sorting the samples
of within the current block descending by energy,
whilst memorizing the sample indices, taking the rst percent
of the sorted index table, and assigning logical 1 to the detection
output sequence at the cropped index table positions.
The remaining values for are set to 0, indicating the Fig. 2. Block diagram showing the partitioning of signal vector into known
samples and unknown samples , and the reassembling of and the inter-
non-impulsive samples. polation result that replaces the discarded samples , to obtain the inter-
polated audio signal .
C. Autoregressive Interpolation
After detection, the impulsive samples have to be interpo-
lated. In the literature, several methods have been proposed for
this task, e.g., median ltering [34], neural network approaches
like in [24], or frequency-domain based approaches as described
in [35]. The latter is well-suited for the restoration of clipped
audio signals. However, according to [22] and others, the time-
domain least-squares AR-based interpolator, proposed in [36]
and [17], is a very effective tool for audio restoration. To ob-
tain the interpolation rule, Eq. (12) is written in a matrix/vector
form, and minimized in the least-squares sense [17], [20].
Fig. 3. Partitioning of the AR coefcient matrix into matrices and ,
(13) according to the positions of known and unknown samples in a signal block.

Please note, that the dependency of the block index is
omitted in Eq. (13), for reasons of readability. In Eq. (13),
(14)
is a vector consisting of the non-impulsive
or known samples within the audio signal (the operator
calculates the nearest integer), i.e., is a subset of the vector
(15)
of length that the detection algorithm identied as clean, non-
impulsive samples, in order of their appearance in . is the
interpolator solution for the impulsive or unknown samples
within in the least-squares sense. The residual signal block is
interpolated, using the block AR interpolator given by Eq. (13).
The detected unknown samples in are replaced by the inter-
polation result , to form a processed residual signal , spectral representation
resp. . The separation and reassembling proce-
dure is shown in Fig. 2. The matrices and from Eq. (13)
are obtained by column-wise partitioning of a matrix of size
, consisting of the AR coefcients, according to
the positions of known and unknown samples and (in-
dicated by ) within a signal block, as visualized in
Fig. 3. The AR order is chosen to be , according to
the ndings in [21]. Multiple iterations of detection and interpo-
lation could be calculated to improve the AR estimation. How-
ever, the overall improvement for the restored signal would be
rather small (cf. [21]).
D. Spectral Correction and Recombining
The penultimate step is a spectral correction of the in-
terpolated residual. After the time-domain interpolation of
, it may happen that some frequency bins in the
, which have been set to 0 by
1684 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 23, NO. 10, OCTOBER 2015

Fig. 4. Zoomed sections of the squared magnitude spectrum, (a) of the binary
masking residual , (b) after interpolation, and (c) after interpolation and
spectral correction . Arrows indicate audible energy regain caused by
interpolation.

the BM process, regain energy (as visualized in Fig. 4(b), by

arrows). This phenomenon is perceivable, especially in the
low-frequency range, as rumbling. Therefore, we suggest to set Fig. 5. Block diagram of the algorithm. Grey circles indicate the algorithms
such bins back to zero again in , the spectral repre- fundamental parameters (decay), and detection percentage .
sentation of . Although this procedure has slight
inuence on the whole time-domain signal block again, the function (ICDF) of the noise, which is given in Table I,
interpolated sections remain reduced in power. The nal step the cropping borders and of the histogram can be
of the proposed algorithm is to recombine target and residual, calculated from as (cf. [19])
i.e., adding back to the target signal ,
to obtain the restored audio block (17)

(16) Then, the variance , and thereby the power of the noisy
residual is dened by
Finally, applying a Hann-window and overlap-add procedure
[37] results in the full-length restored audio signal .
Fig. 5 shows a block diagram of the whole algorithm. (18)
The following section summarizes briey the ndings of [19]
about the analytic prediction of the noise reduction level, i.e.
about the choice of the parameter , that can be obtained by with being the statistical power density function (PDF) of
binary masking in combination with AR interpolation. For more the noise process (also listed in Table I). Finally, the interpolated
details, the interested reader is referred to [19]. noise level in dB is expressed as

III. ANALYTIC PREDICTION OF NOISE REDUCTION
Some of the theoretical aspects of this paper were addressed
in an earlier contribution by the authors [19]. Assuming a BM
scheme as introduced in the previous section, and assuming the
split-apart residual signal having a white spectrum (which in
fact is a patchy spectrum, since multiple T-F bins are set to zero
during the masking process), it was shown that the proposed AR
detection and interpolation process equals to setting the highest
energy time domain samples of the residual noise signal to zero.
This, furthermore, means that the time sample histogram of the
residual noise gets truncated from both sides, depending on the
detection percentage . Knowing the inverse cumulative density
(19)
Summarized, the noise power after AR interpolation can be cal-
culated by using the well-known statistic variance calculation
integral with the PDF, and taking the concatenation borders as
the variance integrals delimiters from the ICDF. This theory
was veried by simulations in [19], using Gaussian, Laplacian,
and modied Cauchy noise with a variable density parameter .
These noises offer distinct sound characteristics and different
values of the sample kurtosis , which can be seen as a measure
for the peakiness or impulsiveness of the noise (cf. Table I).
The simulation results of [19] showed that the level of noise
reduction increases with growing detection percentage , and
RUHLAND et al.: REDUCTION OF GAUSSIAN, SUPERGAUSSIAN AND IMPULSIVE NOISE
TABLE I
THREE DIFFERENT NOISE TYPES AND THEIR PDFS/ICDFS, SAMPLE KURTOSIS , AND PERCEIVED SOUND IMPRESSION.
GAUSSIAN AND LAPLACE DENSITY FORMULAE ARE OF ZERO MEAN VALUE AND VARIANCE ONE. THE MODIFIED
CAUCHY DISTRIBUTION OFFERS A DENSITY PARAMETER .
Fig. 6. Analytic prediction of the achievable SNR improvement over the de-
tection percentage for white Cauchy noise (solid) and white Gaussian noise
(dashed).
that the amount of noise reduction is also dependent on the
sample kurtosis of the noise signal. The higher the sample
kurtosis, i.e., the more peaky or impulsive the noise is, the
more noise reduction can be obtained at a xed detection
percentage . Fig. 6 shows the analytic prediction of the achiev-
able SNR improvement over , for white Gaussian and white
Cauchy noise , derived from Eqs. (17)(19). Please
note, that although Fig. 6 depicts the whole range of from 0 to
100%, only values below 50% should be used, since for proper
AR interpolation the number of unknown samples should not
exceed the number of known samples. Additionally, using
high values can reduce the residual signal level too much,
so that musical noise might become audible in the restoration
result.
In the following section, these ndings will be veried by ob-
jective measures on degraded audio signals, that are processed
with the proposed implementation of the algorithm described
in Section II. Furthermore, a state-of-the-art algorithm will be
tested versus the proposed algorithm. The analytic prediction
will help to adjust the performance of the proposed algorithm to
the performance of the state-of-the-art algorithm.
IV. OBJECTIVE EVALUATION
Three musical genres were used for the following evalua-
tions: Classical music, pop music, and speech. From each genre,
ten audio tracks of different artists were selected. All pieces are
1685
DENOTES THE INVERSE ERROR FUNCTION
under Creative Commons licence or otherwise allowed for re-
search. Some examples are located on the website to this article
[26], namely:
Symphony No. 5 by Gustav Mahler (genre classic)
An industrial synth-pop song (genre pop)
A spoken-word recording of the fable The North Wind
and the Sun [38] in the German version (genre speech).
Each clean audio signal is of length 30 seconds and mixed to
mono. Two types of noise disturbances are added to the clean
audio signals. The rst noise signal is Gaussian white noise
having a kurtosis of . The second noise signal is a super-
gaussian white noise generated by the modied Cauchy noise
generator published in [39], having a kurtosis of (den-
sity parameter ). Although the proposed BMRI algo-
rithm also works for colored noise, only white noise signals will
be tested here to be in accordance to the analytic predictions of
Section III. Both white noise signals are mixed to the clean audio
signals to form a disturbed signal at different input SNRs. Two
restoration algorithms will be compared:
The least squares autoregressive (LSAR) method, pro-
posed by Vaseghi [15]. This algorithm is a well-known
standard in professional audio restoration. We use the
source code provided on the webpage of Nuzman [21].
The proposed algorithm, in the following termed as binary
mask residual interpolation method (BMRI).
For both algorithms, the detection AR order and AR in-
terpolation order were set to 32 samples, according to the
ndings in [21]. It is hardly possible for the LSAR algorithm to
predict the SNR improvement in dB a priori, so we decided to
use default parameters (threshold , fatness , one iteration
only, cf. [21] for further information). The audio material was
processed using the given setting, and the SNR improvement
was measured afterwards. Based on this, the BMRI algorithms
detection percentage was set accordingly, using the analytic
prediction results from Fig. 6, to achieve similar SNR improve-
ment as the LSAR algorithm.
A. Objective Quality Measures
Two objective quality measures are applied to evaluate the
performance of the algorithms. After introducing the measures
in the following, the measurement results will be presented.
From the great variety of objective measures [40] that, in gen-
eral, can be used to assess the performance of noise reduction
algorithms, we chose the SNR improvement measure and the
overall Perceptual Similarity Measure (PSMt) of the Oldenburg
Perception Model (PEMO) [41] based on the auditory model by
1686 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 23, NO. 10, OCTOBER 2015
Dau et al. [42]. The SNR measure is a frequently used quality
measure that is easy to calculate, but as stated in [43], not always
shows high correlation with subjective results. Perceptual mea-
sures are better suited to determine the perceived signal quality
[44]. Since the commonly used perceptual evaluation of speech
quality scores (PESQ) [45] is limited to 16 kHz sampling fre-
quency and speech only, we decided to supplement the eval-
uation by the PEMO-based perceptual measure PSMt. Overall
trends between results for PESQ and PSMt are similar. How-
ever, PEMO allows the use of audio material of higher sampling
frequencies and is not limited to speech signals. Furthermore, it
has been shown that PEMO is well-suited for the evaluation of
noise reduction tasks [44].
1) SNR measure: The input and output signal-to-noise ratios
( and ) can be determined in case the clean audio
signal is available. The can be set by mixing the normal-
ized clean audio signal and the normalized noise disturbance
signal with adjusted levels. After processing by the denoising
algorithm, the (in dB) can be calculated as the average
SNR over all discrete time frames by [40]
(20)
where is the clean audio signal without noise,
is the restored audio signal after processing,
is the set of frames with speech activity and its cardinality
[46]. The SNR improvement is then obtained by
(21)
It is common practice to plot the over the to
visualize the input-output behavior of the investigated system.
2) Overall Perceptual Similarity Measure (PSMt) using
PEMO: Since the SNR measures do not incorporate any
knowledge about the human auditory system [43] and it has
been shown that taking such information into account in objec-
tive quality assessment [44] is of importance, we furthermore
calculate the Perceptual Similarity Measure (PSM) which
was originally developed to predict quality degradations of
broadband audio signals and which is based on the linear cross
correlation coefcient of internal representations of signal
pairs in blocks of 10 ms length [47]. The measure PSMt is the
5th percentile of the PSM output, calculating the perceptual
distance between a test signal and a reference signal in a
range between zero and one. A PSMt value of zero means
no similarity, whereas a value of one stands for both test and
reference signal being perceptually identical. PSMt showed
high correlation with subjective ratings [41], [44]. For the input
PSMt measure , the clean signal serves as reference,
and a degraded signal at a given SNR is used as test signal.
The output PSMt measure is calculated using the
clean audio signal as reference again, and the restored audio
signal as test signal. As before for the SNR measures, the PSMt
improvement is calculated by subtracting
from , and nally, plotted over . target signal is practically free of white Cauchy noise, but yet
B. Measurement Results
The top row of Fig. 7 shows the SNR improvement
over the input SNR, , for the LSAR algorithm (left panel),
the proposed BMRI algorithm with a matched setting (middle
panel), and again the BMRI with a high setting (right panel). The
bottom row of the plots shows the perceptual measure
over the input SNR. Each panel contains three curves for the
genres pop, classical music, and speech in white Cauchy noise
(black lines), and three for the genres in Gaussian noise (dark-
grey lines), plus the neutral line (dashed, no improvement). Fur-
thermore, two dashed light-grey lines indicate the performance
of the IBM (diamond markers) and the proposed BM (triangle
markers) on classical music and white Cauchy noise as best
performing genre, both without interpolation and recombina-
tion of target and residual, i.e. measured on the extracted target
signals alone. They dene an example for oor and ceiling of
performance for standalone binary masking, highlighted as a
light-grey corridor in Fig. 7.
The matched setting (middle panels) means that the SNR im-
provement of the BMRI is set equal to the measured SNR im-
provement of the LSAR algorithm at its highest point (speech at
0 dB input SNR, white Cauchy noise). The reason for choosing
that point is, that speech, with its inherent pauses, at a low input
SNR, comes closest to the condition of a theoretically white
BM residual signal with no tonal components. This is the pre-
liminary assumption that has to be satised in order to pre-
dict the SNR improvement of the BMRI algorithm in depen-
dence of the detection percentage parameter and vice versa
(cf. Section III). The LSAR algorithm offers a dB
at that point, so according to Fig. 6, the BMRI detection per-
centage was set to , resulting in a very close match of
BMRI and LSAR for speech at 0 dB SNR in white Cauchy noise.
For the high setting of the BMRI in the right panel, a detection
percentage with a theoretical dB was
chosen, to investigate possible degradations of the desired signal
by interpolation artefacts at higher values.
C. Discussion
The results for the LSAR algorithm (left panels in Fig. 7)
show a good SNR improvement for the white Cauchy noise at
low input SNR. Towards higher input SNR, the improvement
drops, and even reaches negative values for pop music at 20 dB
input SNR, as indicated by the standard deviation bars. Here,
the desired signal gets affected by the LSAR algorithm. Since
at 20 dB input SNR the white Cauchy noise is quite low com-
pared to the desired signal, it is obvious that transients like e.g.
drum sounds cause high AR error signals that trigger the inter-
polation threshold and thus get degraded. The speech and clas-
sical music signals achieve the best SNR improvement, since
its inherent transient sounds (like e.g. plosives) are lower in
energy than transients in pop music. For the Gaussian noise
there is no SNR improvement at all for low SNR, and nally,
some negative improvement at 15 and 20 dB input SNR, for
the same reasons mentioned above. Compared to the standalone
performance of the proposed BM, the LSAR seems to be worse
at low input SNR. However, this is an erroneous belief, as the
PSMt corridor in the lower panel conrms. The proposed BMs
RUHLAND et al.: REDUCTION OF GAUSSIAN, SUPERGAUSSIAN AND IMPULSIVE NOISE 1687

Fig. 7. (top row) and (bottom row) over , for LSAR (left column) and for BMRI (matched setting, middle column, and high setting, right
column). Speech (circles), classic (crosses) and pop (squares) are degraded by the supergaussian white Cauchy noise (black lines) and Gaussian noise (dark-grey
lines). The dashed, light-grey lines show the performance of the IBM (diamonds) and the proposed BM (triangles) on classical music and white Cauchy noise (best
performer) obtained from the respective target signals (no recombination of target and residual), and thereby dene an area of performance for standalone binary
masking without interpolation (light-grey area). Remark: Marker positions are shifted on the x-axis, for better visibility.

dominated by a lot of musical noise. This perceptual mist is not
reected within the SNR measure. Therefore, the PSMt curves
in the bottom left panel of Fig. 7 show a different behavior.
The highest PSMt improvement for white Cauchy noise at low
input SNR is now obtained for pop and classical music and not
for speech, as for the measures. By listening to the de-
graded input audio les [26], it can be noticed that the white
Cauchy noise is more annoying within speech as it is within the
wide-band pop and classical music, although the mixing of clean
audio and noise at distinct SNRs was carried out carefully. This
is an effect of auditory masking [48]. If a single sine tone and
a stationary narrow-band noise masker of the same center fre-
quency is presented to a human listener at the same time, the lis-
tener will not be able to detect the tone within the noise at a cer-
tain signal-to-masker ratio (SMR) [1]. This works vice versa,
meaning that noise can be masked by a tone, however, needing
a higher SMR [49]. The auditory masking effects are taken in ac-
count within modern audio codecs, and they are incorporated as
well in the auditory model [42] of PEMO-Q. Now, considering
wide-band classical or pop music, more noise is masked within
the auditory system than it is by the sparsely lled spectrum of
speech. This nally leads to the lower PSMt improvement for
speech in the PSMt plots, and might also explain the decline in
PSMt improvement towards higher SNRs for pop and classical
music. The grey curves for Gaussian noise in the bottom left
panel show no PSMt improvement for that noise type, whether
positive nor negative, although there is some negative at
15 and 20 dB input SNR for mainly pop music in the upper left
panel. Here as well, auditory masking effects might hide that the
desired signal gets affected by the restoration algorithm.
The middle panels of Fig. 7 show the results for the BMRI
algorithm, congured with the matched setting. Looking at the
upper middle panel, the 0 dB input SNR point for speech in
white Cauchy noise offers the same amount of SNR improve-
ment like the LSAR algorithm in the upper left panel. How-
ever, the proposed BMRI algorithm yields higher for
the upper input SNRs, giving no negative any more,
and overall higher values for all SNR conditions. Even
for Gaussian noise, SNR improvement is visible. The ability of
the BMRI algorithm to reduce Gaussian noise disturbances has
already been shown in [19]. Best is obtained for speech
and classical music, both for Gaussian and supergaussian dis-
turbance. The considerably better results for BMRI especially at
high input SNR are due to the carefully elaborated BM threshold
criterion in combination with the constraint to only interpolate
a xed amount of samples per signal block.
In the right column of Fig. 7 the results for the BMRI algo-
rithm with the high setting at are plotted. Even at this
fairly high level of noise reduction, there are almost no negative
values at input SNR 15 and 20 dB for both noise types
than for the LSAR algorithm, meaning that there is less affection
of the desired signal by the BMRI at high input SNR compared
to LSAR, although the amount of noise reduction is higher. In
the bottom right panel, the black curves for white Cauchy noise
show the same tendency over input SNR as in the middle panel
for the matched setting, but at a clearly higher . Finally,
also the dark-grey curves show more perceptual improvement
now, conrming the ability of the proposed BMRI algorithm to
reduce Gaussian noise as well.
It is worth mentioning that both LSAR and BMRI outper-
forms the ideal binary mask for high input SNR, in terms of per-
ceptual measurement. Informal listening conrms that the IBM
produces musical noise and artefacts even at an SNR of 15 and
20 dB and therefore gives poor perceptional performance.
1688 IEEE/ACM TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 23, NO. 10, OCTOBER 2015
V. SUBJECTIVE EVALUATION
The previously described objective quality measures already
reveal valuable quantitative properties of the examined audio al-
gorithms. However, objective methods for predicting the overall
quality perceived by the human listeners may not fully correlate
to the subjective opinion of test listeners [44], [50]. Therefore,
it is often advantageous to measure the perceived quality also
by means of subjective listening tests. In this contribution, we
assessed the subjective perceived quality of the processed audio
samples based by means of subjective ratings of ten normal
hearing persons with musical background. A subset of the audio
samples from Section IV was used (pop music, classical music,
and speech), again degraded by adding white Cauchy noise of
density ( ), at 10 dB and 20 dB SNR. LSAR
and the proposed BMRI algorithm were applied for restoration.
For the LSAR algorithm, the default setting was used, and for
BMRI, the matched setting (cf. Section IV for the parameters of
the settings).
The objective quality measures calculated in Section IV-B,
as well as informal listening tests, reveal just minor improve-
ment for the standard Gaussian noise. Higher settings beyond
12.5% would be necessary to achieve similar noise reduction
B. Results of Listening Tests
performance as in the white Cauchy case. If is raised, more and
more samples of the residual get interpolated and the residual
signal loses energy, and thus, the musical noise of the target
signal would become audible in the restoration result. In contrast
to speech enhancement for, e.g., hands-free systems in cars or as
pre-processing for automatic speech recognition (ASR), here it
is crucial for high quality audio restoration to not introduce any
artefacts, be it at the cost of obtaining less noise reduction than
possible. Besides that, the objective measures showed that the
common LSAR approach hardly improves Gaussian disturbed
signals at all. Informal listening to the example signals on the
website [26] conrm this. For these reasons, we will focus on
the white Cauchy noise only and omit the Gaussian noise type
in the following subjective evaluations.
A. Evaluation Procedure
Two different subjective evaluation methods are used to ana-
lyze different aspects of the processed signals:
1) Pairwise comparison by a two-alternative forced choice
test (2-AFC) [51]: The noisy test signals (classic, pop,
speech) have either been processed by BMRI or LSAR or
left unprocessed. For both noise conditions (10 and 20 dB
SNR) and for each genre, three possible algorithm combi-
nations had to be tested, namely BMRI vs. LSAR, BMRI
vs. unprocessed, and LSAR vs. unprocessed, resulting in a
total of 18 pairs. These pairs were presented in six runs
at 36 random pair trials per run. The subjects were in-
structed to choose the audio sample with the preferred
overall quality in each trial. To determine the rank order
of the corresponding algorithms, the Bradley-Terry-Luce
(BTL) model [52], [53] has been applied with a minimum
required signicance of 99% to declare a test valid.
2) A Mean Opinion Score (MOS) test procedure was used
to evaluate the effect of the denoising algorithms on the
residual noise and the target signal individually. It has
been adopted from the ITU-T recommendations P.835
[54], commonly used to evaluate the performance of
noise reduction algorithms in speech communication
systems. For this test, the subjects were asked to assess
the quality dimensions degradation of desired signal,
annoyance of residual noise, and overall quality on a
quasi-continuous MOS scale, ranging from one to ve
[44], [55] (1=bad or very annoying, 2=poor or annoying,
3=fair or slightly annoying, 4=good or perceptible but
not annoying, 5=excellent or imperceptible) while they
could switch seamlessly between a restored audio signal
and a clean, undisturbed version of the signal. Besides the
restored signals from the LSAR and BMRI algorithm, a
third set of test signals was generated by mixing the clean
signals with noise at a 15 dB better SNR than the SNR of
the LSAR and BMRI input signals. This additional signal
set serves as the imitation of an imaginary artefact-free
denoising algorithm with 15 dB noise reduction, as quality
reference (REF). Six runs were performed over all test
les and subjects, where runs one and two were discarded
(they served as training phase for the subjects). Results
from run three to six were averaged, as in [43].
1) 2-AFC test: The results of the AFC test evaluation are
shown in Table II for an input SNR of 10 dB and in
Table III for an input SNR of 20 dB, respectively. The
proposed BMRI algorithm is preferred for both SNR con-
ditions and all three musical genres. For all test scenarios,
the unprocessed signal is rated worst in terms of overall
signal quality. The BTL-model scale differences towards
the rst rank are given in parenthesis for each algorithm
[53]. Mostly, the distances to the third rank (unprocessed
signal) are rather large, whereas the distances from the
second rank (LSAR) against the BMRI tend to be smaller.
This shows that both, BMRI and LSAR achieve major
improvement in quality towards the unprocessed signal,
but nevertheless, BMRI outperforms the common LSAR
approach. For pop music at 10 dB SNR, a large BTL
distance of the LSAR algorithm towards the rst rank can
be observed. This indicates the advantage of the proposed
BMRI algorithm over the common approaches, e.g. in
terms of being able to preserve desired transients like
snare drums etc., as already explained in Section II-A.
The signicance level for all AFC tests is 99%. The con-
sistency in Tables II and III is a measure for the amount
of inconsistent subject ratings [56]. A consistency of zero
means that the subjects answers are completely contra-
dictory, whereas a value of one means that there are no
contradicting statements. All consistency values show
that there were no contradictions amongst the subjects
individual ratings except for classical music at SNR 10 dB
(consistency 0.9). This small deviation could arise from
the fact that classical music offers a high dynamic range,
and therefore, it might be more difcult to distinguish the
restoration results of very silent parts in the music at that
rather low SNR.
2) MOS test: Fig. 8 to 10 show the results of the conducted
MOS rating regarding target signal degradation (cf. Fig. 8),
RUHLAND et al.: REDUCTION OF GAUSSIAN, SUPERGAUSSIAN AND IMPULSIVE NOISE
Fig. 8. Results of the MOS test: Individual rating of the degradation of the
target signal. The input SNR of the test signals was20 dB. The REF signal imi-
tates an artefact-free denoising algorithm with 15 dB reduction. Rating 5 = the
target signal is not audibly degraded, 1 = unacceptable degradation.
Fig. 9. Results of the MOS test: Individual rating of the annoyance of the
residual noise signal. The input SNR of the test signals was 20 dB. The REF
signal imitates an artefact-free denoising algorithm with 15 dB reduction. Rating
5 = the residual noise is not annoying / not audible at all, 1 = unacceptable an-
noyance by the residual noise.
Fig. 10. Results of the MOS test: Individual rating of the overall output signal
quality. The input SNR of the test signals was 20 dB. The REF signal imitates an
artefact-free denoising algorithm with 15 dB reduction. Rating 5 = best overall
quality, 1 = unacceptable overall quality of the restored signal.
1689
TABLE II
RESULTS OF THE 2-AFC LISTENING TESTS AT 10 dB SNR
TABLE III
RESULTS OF THE 2-AFC LISTENING TESTS AT 20 dB SNR
residual noise annoyance (cf. Fig. 9), and overall quality
(cf. Fig. 10). It can be seen that the preferred choice of
algorithm depends on the type of target signal that is pro-
cessed. The medians reveal that the BMRI algorithm is pre-
ferred in comparison to the LSAR algorithm for all test
signals over all three quality dimensions, as already indi-
cated by the results of the 2-AFC test (cf. Tables II and III).
Especially in terms of the annoyance of the residual noise
signal (Fig. 9), the proposed BMRI algorithm outperforms
the LSAR algorithm clearly. However, the ratings for the
simulated artefact-free denoising (REF) indicate that there
is still room for improvement of the BMRI algorithm.
VI. CONCLUSION
In this paper we presented an approach for the reduction of
noise disturbances of Gaussian, supergaussian and impulsive
characteristics. Although the approach is especially suited to
reduce supergaussian and impulsive noise, also Gaussian dis-
turbances can be reduced to a certain extent. High preservation
of the desired signal is achieved by a carefully elaborated BM
threshold criterion in combination with a modied AR detection
and interpolation stage in the time-domain. We showed by an-
alytical treatment that introducing an AR detection percentage
parameter allows for precise prediction of the noise reduction
level for white disturbances. We tied in with these ndings and
veried the suitability of the approach for high quality audio
restoration by means of objective and subjective quality mea-
sures. The objective measures conrm the analytic prediction,
and, together with the subjective listening tests, it is shown that
the proposed approach is superior to the common LSAR detec-
tion and interpolation methods in audio restoration, especially
for supergaussian noise types.
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Marco Ruhland studied electrical engineering at the
Cooperative State University of Mosbach, Germany.
He received his Dipl.-Ing. (BA) degree in 2001.
After ve years in the electrical construction group
of the Michael Weinig AG, Tauberbischofsheim,
he studied hearing technology and audiology at
the Jade University of Applied Sciences of Old-
enburg, Germany, graduating with the B.Eng. in
2010 and continuing with the master studies at
the Carl-von-Ossietzy University of Oldenburg,
Germany. He received his M.Sc. in 2012 and is now
with the Fraunhofer Institute for Digital Media Technology (IDMT), project
group Audio Technology for Assistive Systems, in Oldenburg, Germany. His
main research interests are on speech enhancement, speech recognition, event
detection, and audio restoration algorithms.
Joerg Bitzer received his diploma in 1995 and his
doctorate in electrical engineering in 2002 from the
University of Bremen where he also was a Research
Assistant until 1999. From 2000 to 2003 he was
Head of the algorithm development team at Houpert
Digital Audio, a company specialized in audio signal
processing. Since September 2003, he has been a
Professor for audio signal processing at the Jade
University of Applied Sciences Wilhelmshaven/Old-
enburg/Elseth. Additionally, in 2010, he joined the
Fraunhofer project group for hearing, speech, and
1691
audio technology in Oldenburg as a Scientic Supervisor and has been the
Deputy Head of the Transfer Center for User-Oriented Assistance Systems
since 2013. His current research interests include all forms of single- and
multichannel speech enhancement, audio restoration, audio effects for musical
applications, and information retrieval for large media archives.
Matthias Brandt was born in Bremen in 1980.
He received his diploma in electrical engineering
in 2008 from the University of Bremen, Germany.
From 2009 to 2012 he was employed at the Jade Uni-
versity of Applied Sciences Oldenburg, Germany.
Since 2012, he has been a Ph.D. student at the Uni-
versity of Oldenburg, Germany, in the eld of audio
restoration. His research focus is on the processing
of music signalsfrom developing methods to extract
parameters required for automatic denoising to
creating electronic music in his spare time.
Stefan Goetze is Head of Audio System Tech-
nology for Assistive Systems at the Fraunhofer
Institute for Digital Media Technology (IDMT),
Project group Hearing, Speech and Audio (HSA)
in Oldenburg, Germany. He received his Dipl.-Ing.
in 2004 and Dr.-Ing. in 2013 at the University of
Bremen, Germany, where he was a Research Engi-
neer from 2004 to 2008. His research interests are
sound pick/up, processing and enhancement, such
as noise reduction, acoustic echo cancellation and
dereverberation, as well as assistive technologies,
humanmachine-interaction, detection and classication of acoustic events and
automatic speech recognition. He is a Lecturer at the University of Bremen
and Project Leader of national and international research projects in the eld
of acoustics for ambient assisted living (AAL) technologies. He is member of
IEEE.