An Apparatus and a Method for Generating Bandwidth
Extension Output Data
Description
The present invention relates to an apparatus and a method
for generating bandwidth extension (BWE) output data, ar.
audio encoder and an audio decoder.
Natural audio coding and speech coding are two major
classes of codecs for audio signals. Natural audio coding
is commonly used for music or arbitrary signals at medium
bit rates and generally offers wide audio, bandwidths.
Speech coders are basically limited to speech reproduction
and may be used at very low bit rate. Wide band speech
offers a major subjective quality improvement over narrow
band speech. Further, due to the tremendous growth of the
multimedia field, transmission of music and other non-
speech signals as well as storage and, for example,
transmission for radio/TV at high quality over telephone
systems is a desirable feature.
To drastically reduce the bit rate, source coding can be
performed using split-band perceptual audio codecs. These
natural audio codecs exploit perceptual irrelevance ana
statistical redundancy in the signal. In case exploitation
of the' above alone is not sufficient with respect to the
given bit rate constraints the sample rate is reduced. It
is also common to decrease the number of composition
levels, allowing occasional audible quantization
distortion, and to employ degradation of the stereo field
through joint stereo coding or parametric coding of two or
more channels. Excessive use of such methods results in
annoying perceptual degradation. In order to improve th<:-
coding performance, bandwidth extension methods such as
spectral band replication (SBR) is used as an efficient

method to generate high frequency signals in an HFR (high
frequency reconstruction) based codec.
In recording and transmitting acoustic signals a noise
floor such as background noise is always present. In order
to generate an authentic acoustic signal on the decoder
side, the noise floor should either be transmitted or be
generated. In the latter case, the noise floor in the
original audio signal should be determined. In spectral
band replication, this is performed by SBR tools or SBR
related modules, which generate parameters that
characterize (besides other things) the noise floor and
that are transmitted to the decoder to reconstruct the
noise floor.
In WO 00/45379, an adaptive noise floor tool is described,
which provides sufficient noise contents in the synthesized
high band frequency components. However, disturbing
artifacts in the high band frequency components are
generated if, in the base band, short-time energy
fluctuations or so-called transients occur. These artifacts
are perceptually not acceptable and prior art does not
provide an acceptable solution (especially if the bandwidth
is limited).
An objective of the present invention is, therefore, to
provide an apparatus, which allows an efficient coding
without perceivable artifacts, especially for speeon
signals.
This objective is achieved by an apparatus for generating
SBR output data according to claim 1, the encoder according
to claim 7, a method for generating SBR output data
according to claim 10, a decoder according to claim 13, a
method for decoding according to claim 14 or an encoded
audio signal according to claim 16.

The present invention is based on the finding that an
adaptation of a measured noise floor depending on energy
distribution of the audio signal within a time portion can
improve the perceptual quality of a synthesized audio
signal on the decoder side. Although from the theoretical
standpoint an adaptation or manipulation of the measured
noise floor is not needed, the conventional techniques to
generate the noise floor show a number of drawbacks. On the
one hand, the estimation of the noise floor based on a
tonality measure, as it is performed by conventional
methods, is difficult and not always accurate. On the other
hand, the aim of the noise floor is to reproduce the
correct tonality impression on the decoder side. Even if
the subjective tonality impression for the original audi.
signal and the decoded signal is the same, there is still
the possibility of generated artifacts; e.g. for speech
signals.
Subjective tests show that different types of speech
signals should be treated differently. In voiced speech
signals a lowering of the calculated noise floor yields a
perceptually higher quality when compared to the original
calculated noise floor. As result speech sounds less
reverberant in this case. In case the audio signal comprise
sibilants an artificial increase of the noise floor may
cover up drawbacks in the patching method related to
sibilants. For example, short-time energy fluctuations
(transients) produce disturbing artifacts when shifted or
transformed into the higher frequency band and an increase
in the .noise floor may also cover these energy fluctuations
up.
Said transients may be defined as portions within
conventional signals, wherein a strong increase in energy
appears within a short period of time, which may or may not.
be constrained on a specific frequency region. Examples for
transients are hits of castanets and of percussion
instruments, but also certain sounds of the human voice as,

for example, the letters: P, T, K, ... . The detection or
this kind of transient is implemented so far always in the
same way or by the same algorithm (using a transient
threshold), which is independent of the signal, whether it
is classified as speech or classified as music. In
addition, a possible distinction between voiced and
unvoiced speech does not influence the conventional cr
classical transient detection mechanism.
Hence, embodiments provide a decrease of the noise floor
for signals such as voiced speech and an increase of the
noise floor for signals comprising, e.g., sibilants.
To distinguish the different signals, embodiments use
energy distribution data (e.g. a sibilance parameter) that
measure whether the energy is mostly located at higher
frequencies or at lower frequencies, or in other words,
whether the spectral representation of the audio signal
shows an increasing or decreasing tilt towards higher
frequencies. Further embodiments also use the first LPC
coefficient (LPC = linear predictive coding) to generate
the sibilance parameter.
There are two possibilities for changing the noise floor.
The first possibility is to transmit said sibilance
parameter so that the decoder can use the sibilance
parameter in order to adjust the noise floor (e.g. either
to increase or decrease the noise floor in addition to the
calculated noise floor). This sibilance parameter may be
transmitted in addition to the calculated noise floor
parameter by conventional methods or calculated on decoder
side. A second possibility is to change the transmitted
noise floor by using the sibilance parameter (or the energy
distribution data) so that the encoder transmits modified
noise floor data to the decoder and no modifications are
needed on the decoder side - the same decoder may be used.
Therefore, the manipulation of the noise floor can in

principle be done on the encoder side as well as on the
decoder side.
The spectral band replication as an example for the
bandwidth extension relies on SBR frames defining a time
portion in which the audio signal is separated into
components in the first frequency band and the second
frequency band. The noise floor can be measured and/or
changed for the whole SBR frame. Alternatively, it is also
possible that the SBR frame is divided into noise
envelopes, so that for each of the noise envelopes, an
adjustment for the noise floor can be performed. In other
words, the temporal resolution of the noise floor tools is
determined by the so-called noise-envelopes within the SBR
frames. According to the Standard (ISO/IEC 14496-3), each
SBR frame comprises a maximum of two noise-envelopes, so
that an adjustment of the noise floor can be made on the
basis partial SBR frames. For some applications, this might
be sufficient. It is, however, also possible to increase
the number of noise-envelopes in order to improve the model
for temporal varying tonality.
Hence, embodiments comprise an apparatus for generating BWE
output data for an audio signal, wherein the audio signal
comprises components in a first frequency band and a second
frequency band and the BWE output data is adaptec to
control a synthesis of the components in the second
frequency band. The apparatus comprises a noise floor-
measurer for measuring noise floor data of the second
frequency band for a time portion of the audio signal.
Since the measured noise floor influences the tonality of
the audio signal, the noise floor measurer may comprise a
tonality measurer. Alternatively, the noise floor measurer
can be implemented to measure the noisiness of a signal in
order to obtain the noise floor. The apparatus further
comprises a signal-energy characterizer for deriving energy
distribution data, wherein the energy distribution data
characterize an energy distribution in a spectrum of the

time portion of the audio signal and, finally, the
apparatus comprises a processor for combining the noise
floor data and the energy distribution data to obtain the
BWE output data.
In further embodiments, the signal energy characterizer is
adapted to use the sibilance parameter as the energy
distribution data and the sibilance parameter can, for
example, be the first LPC coefficient. In further
embodiments, the processor is adapted to add the energy
distribution data to the bitstream of encoded audio data
or, alternatively, the processor is adapted to adjust the
noise floor parameter such that the noise floor is either
increased or decreased depending on the energy distribution
data (signal dependent) . In this embodiment, the noise
floor measurer will first measure the noise floor to
generate noise floor data, which will be adjusted or
changed by the processor later on.
In further embodiments, the time portion is an SBR frame
and the signal energy characterizer is adapted to generate
a number of noise floor envelopes per SBR frame. As a
consequence, the noise floor measurer as well as the signal
energy characterizer may be adapted to measure the noise
floor data as well as the derived energy distribution data
for each noise floor envelope. The number of noise floor
envelopes can, for example, be 1, 2, 4, ... per SBR frame.
Further embodiments comprise also a spectral band
replication tool used in a decoder to generate compor.enr.s
in a second frequency band of the audio signal. In this
generation spectral band replication output data and raw
signal spectral representation for the components in the
second frequency band are used. The spectral band
replication tool comprises a noise floor calculation unit,
which is configured to calculate a noise floor in
accordance to the energy distribution data, and a combiner
for combining the raw signal spectral representation with

the calculated noise floor to generate the components in
the second frequency band with the calculated noise floor.
An advantage of embodiments is the combination of an
external decision (speech/audio) with an internal voiced
speech detector or an internal sibilant detector (a signal
energy characterizer) controlling the event of additional
noise being signaled to the decoder or adjusting the
calculated noise floor. For non-speech signals, the usual
noise floor calculation is executed. For speech signals
(derived from the external switching decision) an
additional speech analysis is performed to determine the
actual signal's voicing. The amount of noise to be added in
the decoder or encoder is scaled depending on the degree of
sibilance (to be contrary to voicing) of the signal. The
degree of sibilance can be determined, for example, by
measuring the spectral tilt of short-signal parts.
Brief Description of the Drawings
The present invention will now be described by way of
illustrated examples. Features of the invention will be
more readily appreciated and better understood by reference
to the following detailed description, which should be
considered with reference to the accompanying drawings, in
which:
Fig. 1 shows a block diagram of an apparatus for
generating BWE output data according to
embodiments of the present invention;
Fig. 2a illustrates a negative spectral tilt of a non-
sibilant signal;
Fig. 2b illustrates a positive spectral tilt for a
sibilant-like signal;

Fig. 2c explains the calculation of the spectral tilt m
based on low-order LPC parameters;
Fig. 3 shows a block diagram of an encoder;
Fig. 4 shows block diagrams for processing the coded
audio stream to output PCM samples on a decoder
side;
Fig. 5a,b show a comparison of a conventional noise floor
calculation tool with a modified noise floor
calculation tool according to embodiments; and
Fig. 6 illustrates the partition of an SBR frame in a
predetermined number of time portions.
Fig. 1 shows an apparatus 100 for generating bandwidth
extension (BWE) output data 102 for an audio signal 105.
The audio signal 105 comprises components in a first
frequency band 105a and components of a second frequency
band 105b. The BWE output data 102 are adapted to control a
synthesis of the components in the second frequency band
105b. The apparatus 100 comprises a noise floor measurer
110, a signal energy characterizer 120 and a processor 130.
The noise floor measurer 110 is adapted to measure or
determine noise floor data 115 of the second frequency band
105b for a time portion off the audio signal 105. In detail,
the noise floor may be determined by comparing the measured
noise of the base band with the measured noise of the upper
band, so that the amount of noise needed after patching to
reproduce a natural tonality impression may be determined.
The signal energy characterizer 120 derives energy
distribution data 125 characterizing an energy distribution
in a spectrum of the time portion of the audio signal 105.
Therefore, the noise floor measurer 110 receives, for
example, the first and/or second frequency band 105a,b and

the signal energy characterizer 120 receives, for example,
the first and/or the second frequency band 105a, b. The
processor 130 receives the noise floor data 115 and the
energy distribution data 125 and combines them to obtain
the BWE output data 102. Spectral band replication
comprises one example for the bandwidth extension, wherein
the BWE output data 102 become SBR output data. The
following embodiments will mainly describe the example of
SBR, but the inventive apparatus/method is not restricted
to this example.
The energy distribution data 125 indicates a relation
between the energy contained within the second frequency
band compared to the energy contained in the first
frequency band. In the simplest case the energy
distribution data is given by a bit indicating whether more
energy is stored within the base band compared to the SBR
band (upper band) or vice versa. The SBR band (upper band)
may, for example, be defined as frequency components above
a threshold, which may be given, for example, by 4 kHz and
the base band (lower band) may be the components of the
signal, which are below this threshold frequency (for
example, below 4 kHz or another frequency). Examples for
these threshold frequencies would be 5 kHz or 6 kHz.
Figs. 2a and 2b show two energy distributions in the
spectrum within a time portion of the audio signal 105. The
energy distributions displayed by a level P as a function
of the frequency F as analog signal, which may also be an
envelope of a signal given by a plurality of samples or
lines (transformed into the frequency domain). The shown
graphs are also much simplified to visualize the spectral
tilt concept. The lower and upper frequency band may be
defined as frequencies below or above a threshold frequency
F0 (cross over frequency, e.g. 500 Hz, 1 kHz or 2 kHz).
Fig. 2a shows an energy distribution exhibiting a falling
spectral tilt (decreasing with higher frequencies). In

other words, in this case, there is more energy stored in
the low frequency components than in the high frequency
components. Hence, the level P decreases for higher
frequencies implying a negative spectral tilt (decreasing
function). Hence, a level P comprises a negative spectral
tilt if the signal level P indicates that there is less
energy in the upper band (F > F0) than in the lower band (F
< F0) . This type of signal occurs, for example, for an
audio signal comprising a low or no amount of sibilance.
Fig. 2b shows the case, wherein the level P increases with
the frequencies F implying a positive spectral tilt (an
increasing function of the level P depending on the
frequencies). Hence, the level P comprises a positive
spectral tilt if the signal level P indicates that there is
more energy in the upper band (F > Fo) compared to the
lower band (F < F0) . Such an energy distribution is
generated if the audio signal 105 comprises, for example,
said sibilants.
Fig. 2a illustrates a power spectrum of a .signal having a
negative spectral tilt. A negative spectral tilt means a
falling slope of the spectrum. Contrary thereto, Fig. 2b
illustrates a power spectrum of a signal having a positive
spectral tilt. Said in other words, this spectral tilt has
a rising slope. Naturally, each spectrum such as the
spectrum illustrated in Fig. 2a or the spectrum illustrated
in Fig. 2b will have variations in a local scale which have
slopes different from the spectral tilt.
The spectral tilt may be obtained, when, for example, a
straight line is fitted to the power spectrum such as by
minimizing the squared differences between this straight
line and the actual spectrum. Fitting a straight line to
the spectrum can be one of the ways for calculating the
spectral tilt of a short-time spectrum. However, it is
preferred to calculate the spectral tilt using LPC
coefficients.

The publication "Efficient calculation of spectral tilt.
from various LPC parameters" by V. Goncharoff, E. Von Colin
and R. Morris, Naval Command, Control and Ocean
Surveillance Center (NCCOSC), RDT and E Division, San
Diego, CA 92152-52001, May 23, 1996 discloses several ways
to calculate the spectral tilt.
In one implementation, the spectral tilt is defined as the
slope of a least-squares linear fit to the log power
spectrum. However, linear fits to the non-log power
spectrum or to the amplitude spectrum or any other kind of
spectrum can also be applied. This is specifically true in
the context of the present invention, where, in the
preferred embodiment, one is mainly interested in the siqn
of the spectral tilt, i.e., whether the slope of the linear
fit result is positive or negative. The actual value of the
spectral tilt, however, is of no big importance in a high
efficiency embodiment of the present invention, but the
actual value can be important in more elaborate
embodiments.
When linear predictive coding (LPC) of speech is used to
model its short-time spectrum, it is computationally more
efficient to calculate spectral tilt directly from the LPC
model parameters instead of from the log power spectrum.
Fig. 2c illustrates an equation for the cepstrai
coefficients Ck corresponding to the nth order all-pole log
power spectrum. In this equation, k is an integer index, pn
is the nth pole in the all-pole representation of the z-
domain transfer function H(z) of the LPC filter. The nexr
equation in Fig. 2c is the spectral tilt in terms of the
cepstrai coefficients. Specifically, m is the spectral
tilt, k and n are integers and N is the highest order pole
of the all-pole model for H(z). The next equation in Fig.
2c defines the log power spectrum S(ω) of the Nth order LPC
filter. G is the gain constant and a* are the linear
predictor coefficients, and ω is equal to 2xπxf, where f is

the frequency. The lowest e'quation in Fig. 2c directly
results in the cepstral coefficients as a function of the
LPC coefficients αk. The cepstral coefficients ck are then
used to calculate the spectral tilt. Generally, this method
will be more computationally efficient than factoring the
LPC polynomial to obtain the pole values, and solving for
spectral tilt using the pole equations. Thus, after having
calculated the LPC coefficients αk, one can calculate the
cepstral coefficients Ck using the equation at the bottom
of Fig. 2c and, then, one can calculate the poles pn from
the cepstral coefficients using the first equation in Fig.
2c. Then, based on the poles, one can calculate the
spectral tilt m as defined in the second equation of Fig.
2c.
It has been found that the first order LPC coefficient or
is sufficient for having a good estimate for the sign of
the spectral tilt. α1 is, therefore, a good estimate for
c1. Thus, c1 is a good estimate for p1. When p1 is inserted
into the equation for the spectral tilt m, it becomes clear
that, due to the minus sign in the second equation in Fig.
2c, the sign of the spectral tilt m is inverse to the sign
of the first LPC coefficient α1 in the LPC coefficient
definition in Fig. 2c.
Preferably, the signal energy characterizer 120 is
configured to generate, as the energy distribution data, ar.
indication on a sign of the spectral tilt of the audio
signal in a current time portion of the audio signal.
Preferably, the signal energy characterizer 120 is
configured to generate, as the energy distribution data,
data derived from an LPC analysis of a time portion of trv?
audio signal for estimating one or more low order LPC
coefficients and derive the energy distribution data from
the one or more low order LPC coefficients.

Preferably, the signal energy characterizer 120 is
configured only calculate the first LPC coefficient and to
not calculate additional LPC coefficients and to derive the
energy distribution data from a sign of the first LPC
coefficient.
Preferably, the signal energy characterizer 120 is
configured for determining the spectral tilt as a negative
spectral tilt, in which a spectral energy decreases from
lower frequencies to higher frequencies, when the first LPC
coefficient has a positive sign, and to detect the spectral
tilt as a positive spectral tilt, in which the spectral
energy increases from lower frequencies to higher
frequencies, when the first LPC coefficient has a negative
sign.
In other embodiments, the spectral zilt detector or signal
energy characterizer 120 is configured to not only
calculate the first order LPC coefficients but to calculate
several low order LPC coefficients such as LPC coefficients
until the order of 3 or 4 or even higher. In such a:i
embodiment, the spectral tilt is calculated to such an high
accuracy that one can not only indicate the sign as a
sibilance parameter, but also a value depending on the
tilt, which has more than two values as in the sign
embodiment.
As said above sibilance comprises a large amount of energy
in the upper frequency region, whereas for parts with no or
only little sibilance (for example, vowels) the energy is
mostly distributed within the base band (the low frequency
band) . This observation can be used in order to determine
whether or to which extend a speech signal part comprise a
sibilant or not.
Hence, the noise floor measurer 110 (detector) can use the
spectral tilt for the decision about the amount of
sibilance or to give the degree of sibilance within a

signal. The spectral tilt can basically be obtained from a
simple LPC analysis of the energy distribution. It may, for
example, be sufficient to calculate the first LPC
coefficient in order to determine the spectral tilt
parameter (sibilance parameter), because from the first LPC
coefficient the behavior of the spectrum (whether an
increasing or decreasing function) can be inferred. This
analysis may be performed within the signal energy
characterizer 120. In case the audio encoder uses LPC for
decoding the audio signal, there may be no need to transmit
the sibilance parameter, since the first LPC coefficient
may be used as energy distribution data on the decoder
side.
In embodiments the processor 130 may be configured to
change the noise floor data 115 in accordance to the energy
distribution data 125 (spectral tilt) to obtain modified
noise floor data, and the processor 130 may be configured
to add the modified noise floor data to a bitstream
comprising the BWE output data 102. The change of the ncise
floor data 115 may be such that the modified noise floor is
increased for an audio signal 105 comprising more sibilance
(Fig. 2b) compared to an audio signal 105 comprising less
sibilance (Fig. 2a).
The apparatus 100 for generating bandwidth extension (BWE)
output data 102 can be part of an encoder 300. Fig. 3 shows
an embodiment for the encoder 300, which comprises BWE
related modules 310 (which may, e.g., comprise SBR related
modules), an analysis QMF bank 320, a low pass filter (LP-
filter) 330, an AAC core encoder 340 and a bit stream
payload formatter 350. In addition, the encoder 300
comprises the envelope data calculator 210. The encoder 300
comprises an input for PCM samples (audio signal 105; PCM =
pulse code modulation), which is connected to the analysis
QMF bank 320, and to the BWE-related modules 310 and to the
LP-filter 330. The analysis QMF bank 320 may comprise a
high pass filter to separate the second frequency band 105b

and is connected to the envelope data calculator 210,
which, in turn, is connected to the bit stream payload
formatter 350. The LP-filter 330 may comprise a low pass
filter to separate the first frequency band 105a and is
connected to the AAC core encoder 340, which, in turn, is
connected to the bit stream payload formatter 350. Finally,
the BWE-related module 310 is connected to the envelope
data calculator 210 and to the AAC core encoder 340.
Therefore, the encoder 300 down-samples the audio signal
105 to generate components in the core frequency band 105a
(in the LP-filter 330) , which are input into the AAC core
encoder 340, which encodes the audio signal in the core
frequency band and forwards the encoded signal 355 to the
bit stream payload formatter 350 in which the encoded audio
signal 355 of the core frequency band is added to the coded
audio stream 345 (a bit stream) . On the other hand, the
audio signal 105 is analyzed by the analysis QMF bank 320
and the high pass filter of the analysis QMF bank extracts
frequency components of the high frequency band 105b and
inputs this signal into the envelope data calculator 210 to
generate BWE data 375. For example, a 64 sub-band QMF BANK
320 performs the sub-band filtering of the input signal.
The output from the filterbank (i.e. the sub-band samples)
are complex-valued and, thus, over-sampled by a factor of
two compared to a regular QMF bank.
The BWE-related module 310 may, for example, comprise -he
apparatus 100 for generating the BWE output data 102 and
controls the envelope data calculator 210 by providing,
e.g., the BWE output data 102 (sibilance parameter) to the
envelope data calculator 210. Using the audio components
105b generated by the Analysis QMF bank 320, the envelope
data calculator 210 calculates the BWE data 375 and
forwards the BWE data 375 to the bit stream payload
formatter 350, which combines the BWE data 375 with the
components 355 encoded by the core encoder 34C in the coded
audio stream 345. In addition, the envelope data calculator

210 may for example use the sibilance parameter 125 to
adjust the noise floors within the noise envelopes.
Alternatively, the apparatus 100 for generating the BWE
output data 102 may also be part of the envelope data
calculator 210 and the processor may also be part of the
Bitstream payload formatter 350. Therefore, the different
components of the apparatus 100 may be part of different
encoder components of Fig. 3.
Fig. 4 shows an embodiment for a decoder 400, wherein the
coded audio stream 345 is input into a bit stream payload
deformatter 357, which separates the coded audio signal 355
from the BWE data 375. The coded audio signal 355 is input
into, for example, an AAC core decoder 360, which generates
the decoded audio signal 105a in the first frequency band.
The audio signal 105a (components in the first frequency
band) is input into an analysis 32 band QMF-bank 370,
generating, for example, 32 frequency subbands 105;,? from
the audio signal 105a in the first frequency band. The
frequency subband audio signal 10532 is input into the
patch generator 410 to generate a raw signal spectral
representation 425 (patch), which is input into an BWE tool
430a. The BWE tool 430a may, for example, comprise a noise
floor calculation unit to generate a noise floor. In
addition, the BWE tool 430a may reconstruct missing
harmonics or perform an inverse filtering step. The BWE
tool 430a may implement known spectral band replication
methods to be used on the QMF spectral data output of the
patch generator 410. The patching algorithm used in the
frequency domain could, for example, employ the simple
mirroring or copying of the spectral data within the
frequency domain.
On the other hand, the BWE data 375 (e.g. comprising the
BWE output data 102) is input into a bit stream parser 380,
which analyzes the BWE data 375 to obtain different sub-
information 385 and input them into, for example, an

Huffman decoding and dequantization unit 390 which, for
example, extracts the control information 412 and the
spectral band replication parameters 102. The control
information 412 controls the patch generator 430 (e.g. to
use a specific patching algorithm) and the BWE parameter
102 comprise, for example, also the energy distribution
data 125 (e.g. the sibilance parameter). The control
information 412 is input into the BWE tool 4 30a and the
spectral band replication parameters 102 are input into the
BWE tool 430a as well as into an envelope adjuster 430b.
The envelope adjuster 430b is operative to adjust the
envelope for the generated patch. As a result, the envelope
adjuster 430b generates the adjusted raw signal 105b for
the second frequency band and inputs it into a synthesis
QMF-bank 440, which combines the components of the second
frequency band 105b with the audio signal in the frequency
domain 10532. The synthesis QMF-bank 440 may, for example,
comprise 64 frequency bands and generates by combining both
signals (the components in the second frequency band 105b
and the frequency domain audio signal 10532) the synthesis
audio signal 105 (for example, an output of PCM samples,
PCM = pulse code modulation).
The synthesis QMF bank 440 may comprise a combiner, which
combines the frequency domain signal 10532 with the second
frequency band 105b before it will be transformed into the
time domain and before it will be output as the audio
signal 105. Optionally, the combiner may output the audio
signal 105 in the frequency domain.
The BWE tools 430a may comprise a conventional noise floor
tool, which adds additional noise to the patched spectrum
(the raw signal spectral representation 425), so that the
spectral components 105a that have been transmitted by a
core coder 340 and are used to synthesize the components of
the second frequency band 105b exhibit the tonality of the
second frequency band 105b of the original signal.
Especially in voiced speech paths, however, the additional

noise added by the conventional noise floor tool can harrr.
the perceived quality of the reproduced signal.
According to embodiments the noise floor tool may bo
modified so that the noise floor tool takes into account
the energy distribution data 125 (part of the BWE data 102)
to change the noise floor in accordance to the detected
degree of sibilance (see Fig. 2) . Alternatively, as
described above the decoder may not be modified and instead
the encoder can change the noise floor data in accordance
to the detected degree of sibilance.
Fig. 5 shows a comparison of a conventional noise floor
calculation tool with a modified noise floor calculation
tool according to embodiments of the present invention.
This modified noise floor calculation tool may be part of
the BWE tool 430.
Fig. 5a shows the conventional noise floor calculation
comprising a calculator 433, which uses the spectral band
replication parameters 102 and the raw signal spectral
representation 425 in order to calculate raw spectral lines
and noise spectral lines. The BWE data 375 may comprise
envelope data and noise floor data, which are transmitted
from the encoder as part of the coded audio stream 34 5. The
raw signal spectral representation 425 is, for example,
obtained from a patch generator, which generates components
of the audio signal in the upper frequency band
(synthesized components in the second frequency band 105b).
The raw spectral lines and noise spectral lines will
further be processed, which may involve an inverse
filtering, envelope adjusting, adding missing harmonics and
so on. Finally, a combiner 434 combines the raw spectral
lines with the calculated noise spectral lines to the
components in the second frequency band 105b.
Fig. 5b shows a noise floor calculation tool accordinq to
embodiments of the present invention. In addition to the

conventional noise floor calculation tool as shown in Fig.
5a, embodiments comprise a noise floor modifying unit 43i
which is configured, for example, to modify the transmitted
noise floor data based on the energy distribution data 125
before they are processed in the noise floor calculation
tool 433. The energy distribution data 125 may also be
transmitted from the encoder as part of or in addition to
the BWE data 375. The modification of the transmitted noise
floor data comprises, for example, an increase for a
positive spectral tilt (see Fig. 2a) or decrease for a
negative spectral tilt (see Fig. 2b) of the level of the
noise floor, for example, an increase by 3 dB or a decrease
by 3 dB or any other discrete value (e.g. +/- 1 dB or +/- 2
dB) . The discrete value can be an integer dB value or a
non-integer dB value. There may also be a functional
dependence (e.g. a linear relation) between the
decrease/increase and the spectral tilt.
Based on this modified noise floor data the noise floor,
calculation tool 433 calculates again raw spectral lines
and modified noise spectral lines based on the raw siqnal
spectral representation 425, which may again be obtained
from a patch generator. The spectral band replication tool
430 of Fig. 5b comprise also a combiner 434 for combining
the raw spectral lines with the calculated noise floor
(with the modification from the modifying unit 431) to
generate the components in the second frequency band 105b.
The energy distribution data 125 may indicate in the
simplest case a modification in the transmitted level of
the noise floor data. As said above also the first LPC
coefficient may be used as energy distribution data 125.
Therefore, if the audio signal 105 was encoded using LPC,
further embodiments use the first LPC coefficient, which is
already transmitted by the coded audio stream 345, as the
energy distribution data 125. In this case there is no need
to transmit in addition the energy distribution data 125.

Alternatively a modification of the noise floor may also be
carried out after the calculation within the calculator 433
so that the noise floor modifying unit 431 may be arranged
after the processor 433. In further embodiments the energy
distribution data 125 may be directly input in the
calculator 433 modifying directly the calculation of the
noise floor as calculation parameter. Hence, the noise
floor modifying unit 431 and the calculator/processor 433
may be combined to a noise floor modifier tool 4 33, 4 31.
In another embodiment the BWE tool 4 30 comprising the noise
floor calculation tool comprises a switch, wherein the
switch is configured to switch between a high level for the
noise floor (positive spectral tilt) and a low level for
the noise floor (negative spectral tilt) . The high level
may, for example, correspond to the case wherein the
transmitted level for the noise is doubled (or multiplied
by a factor), whereas the low level corresponds to the case
wherein the transmitted level is decreased by factor. The
switch may be controlled by a bit in the bit stream of the
coded audio signal 345 indicating a positive or negative
spectral tilt of the audio signal. Alternatively the switch
may also be activated by an analysis of the decoded audio
signal 105a (components in the first frequency band) or of
the frequency subband audio signal 10532, for example with
respect to the spectral tilt (whether the spectral tilt is
positive or negative). Alternatively, the switch may also
be controlled by the first LPC coefficient, since this
coefficient indicates the spectral tile (see above).
Although some of the Figs. 1, 3 through 5 are illustrated
as block diagrams of apparatuses, these figures
simultaneously are an illustration of a method, where the
block functionalities correspond to the method steps.
As said above, an SBR time unit (SBR frame) or a time
portion can be divided into various data blocks, so-called
envelopes. This partition may be uniform over the SBR frame

and allows adjusting flexibly the synthesis of the audio
signal within the SBR frame.
Fig. 6 illustrates such partition for the SBR frame in a
number n of envelopes. The SBR frame covers a time period
or time portion T between the initial time to and a final
time tn. The time portion T is, for example, divided into
eight time portions, a first time portion T1, a second time
portion T2, ..., an eighth time portion T8. In this
example, the maximum number of envelopes coincides with the
number of time portions and is given by n = 8. The 8 time
portions Tl, ... , T8 are separated by 7 borders, that means
a border 1 separates the first and second time portion Tl,
T2, a border 2 is located between the second portion T2 and
a third portion T3, and so on until, a border 7 separates
the seventh portion T7 and the eighth portion T8.
In further embodiments, the SBR frame is divided into four
noise envelopes (n = 4) or is divided inro two noise
envelopes (n = 2). In the embodiment as shown in Fig. £,
all envelopes comprise the same temporal length, which may
be different in other embodiments so that the noise
envelopes cover differing time lengths. In detail, the case
with two noise envelopes (n = 2) comprise a first envelope
extending from the time t0 over the first four time
portions (Tl, T2, T3 and T4) and the second noise envelope
covering the fifth to the eighth time portion (T5, T6, T .'
and T8). Due to the Standard ISO/IEC 14496-3, the maximal
number of envelopes is restricted to two. But embodiments
may use. any number of envelopes (e.g. two, four or eight
envelopes).
In further embodiments the envelope data calculator 210 is
configured to change the number of envelopes depending on a
change of the measured noise floor data 115. For example,
if the measured noise floor data 115 indicates a varying
noise floor (e.g. above a threshold) the number 01
envelopes may be increased whereas in case the noise floor

data 115 indicates a constant noise floor the number of
envelopes may be decreased.
In other embodiments, the signal energy characterizer 120
can be based on linguistic information in order to detect
sibilants in speech. When, for example, a speech signal has
associated meta information such a the international
phonetic spelling, then an analysis of this meta
information will provide a sibilant detection of a speech
portion as well. In this context, the meta data portion of
the audio signal is analyzed.
Although some aspects have been described in the context of
an apparatus, it is clear that these aspects also represent
a description of the corresponding method, where a block or
device corresponds to a method step or a feature of a
method step. Analogously, aspects described in the context
of a method step also represent a description 01 a
corresponding block or item or feature of a corresponding
apparatus.
The inventive encoded audio signal can be stored on a
digital storage medium or can be transmitted on a
transmission medium such as a wireless transmission medium
or a wired transmission medium such as the Internet.
Depending on certain implementation requirements,
embodiments of the invention can be implemented in hardware
or in software. The implementation can be performec using a
digital storage medium, for example a floppy disk, a DVD, a
CD, a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory,
having electronically readable control signals stored
thereon, which cooperate (or are capable of cooperating)
with a programmable computer system such that the
respective method is performed.
Some embodiments according to the invention comprise a data
carrier having electronically readable control signals,

which are capable of cooperating with a programmable
computer system, such that one of the methods described
herein is performed.
Generally, embodiments of the present invention can be
implemented as a computer program product with a program
code, the program code being operative for performing one
of the methods when the computer program product runs on a
computer. The program code may for example be stored on a
machine readable carrier.
Other embodiments comprise the computer program for
performing one of the methods described herein, stored on a
machine readable carrier.
In other words, an embodiment of the inventive method is,
therefore, a computer program having a program code for
performing one of the methods described herein, when the
computer program runs on a computer.
A further embodiment of the inventive methods is,
therefore, a data carrier (or a digital storage medium, or
a computer-readable medium) comprising, recorded thereon,
the computer program for performing one of the methods
described herein.
A further embodiment of the inventive method is, therefore,
a data stream or a sequence of signals representing the
computer program for performing one of the methods
described herein. The data stream or the sequence of
signals may for example be configured to be transferred via
a data communication connection, for example via the
Internet.
A further embodiment comprises a processing means, fox-
example a computer, or a programmable logic device,
configured to or adapted to perform one of the methods-
described herein.

A further embodiment comprises a computer having installed
thereon the computer program for performing one of the
methods described herein.
In some embodiments, a programmable logic device (for
example a field programmable gate array) may be used to
perform some or all of the functionalities of the methods
described herein. In some embodiments, a field programmable
gate array may cooperate with a microprocessor in order to
perform one of the methods described herein. Generally, the
methods are preferably performed by any hardware apparatus.
The above described embodiments are merely illustrative for
the principles of the present invention. It is understood
that modifications and variations of the arrangements and
the details described herein will be apparent to others
skilled in the art. It is the intent, therefore, to be
limited only by the scope of the impending patent claims
and not by the specific details presented by way of
description and explanation of the embodiments herein.

WE CLAIM
1. An encoder (300) for encoding an audio signal (105),
the audio signal (105) comprising components in a
first frequency band (105a) and components in a second
frequency band (105b), the encoder comprising:
a core coder (340) for encoding the components in the
first frequency band (105a) to obtain an encoded audio
signal (355);
an envelope data calculator (210) for calculating
bandwidth extension (BWE) data (375) based on the
components in the second frequency band (105b), the
envelope data calculator comprising an apparatus (100)
for generating bandwidth extension output data (102)
for the audio signal (105), the bandwidth extension
output data (102) being adapted to control a synthesis
of the components in the second frequency band (105b),
the apparatus (100) comprising:
a noise floor measurer (110) for measuring noise
floor data (115) of the second frequency band
(105b) for a time portion (T) of the audio signal
(105);
a signal energy characterizer (120) for deriving
energy distribution data (125), the energy
distribution data (125) characterizing an energy
distribution in a spectrum of the time portion
(T) of the audio signal (105); and
a processor (130) for combining the noise floor
data (115) and the energy distribution data (125)
to obtain the bandwidth extension output data
(102),

wherein the bandwidth extension data (375) comprise
the bandwidth extension output data (102) and envelope
data; and
a bitstream payload formatter (350) adapted for
outputting a coded audio stream (345) by combining the
bandwidth extension data (375) with the encoded audio
signal (355), wherein the processor (130) is part of
the bitstream payload formatter (350).
2. The encoder of claim 1, wherein the signal energy
characterizer (120) is configured to use, as energy-
distribution data (125), a sibilance parameter or a
spectral tilt parameter, the sibilanoe pararmeter or
spectral tilt parameter identifying an increasing or
decreasing level of the audio signal (105) with
frequency (F).
3. The encoder of claim 2, wherein the signal energy
characterizer (120) is configured to use the firs-
linear predictive coding coefficient as the sibilance
parameter.
4. The encoder of one of the preceding claims, wherein
the processor (130) is configured to add the noise
floor data (115) and the spectral energy distribution
data (125) to a bitstream as the BWE output data
(102) .
5. The encoder of one of the claims 1 to 3, wherein the
processor (130) is configured to change the noise
floor data (115) in accordance to the energy
distribution data (125) to obtain modified noise floor
data, and wherein the processor (130) is configured to
add the modified noise floor data to a bitstream as
the BWE output data (102).

6. The encoder of claim 5, wherein the change of the
noise floor data (115) is such that the modified noise
floor is increased for an audio signal (105)
comprising more sibilance compared to an audio signal
(105) comprising less sibilance.
7. The encoder (300) of claim 1, wherein the time portion
(T) covers an SBR frame, the SBR frame comprising a
plurality of noise envelopes, and wherein the noise
envelope data calculator (210) is configured to
calculate different BWE data (375) for different noise
envelopes of the plurality of noise envelopes.
8. The encoder (300) of claim 1 or claim 7, wherein the
envelope data calculator (210) is configured to change
a number of envelopes depending on a change of the
measured noise floor data (115) .
9. A method of encoding an audio signal (105), the audio
signal (105) comprising components in a first
frequency band (105a) and components in a second
frequency band (105b), the method comprising:
encoding (340) the components in the first frequency
band (105a) to obtain an encoded audio signal (355);
calculating bandwidth extension (BWE) data (375) by an
envelope data calculator (210) based on the components
in the second frequency band (105b), the step of
calculating comprising a step of generating bandwidth
extension output data (102) for the audic signal
(105), the bandwidth extension output data (102) beinn
adapted to control a synthesis of the components in
the second frequency band (105b) , the step of
generating bandwidth extension output data comprising:

measuring noise floor data (115) of the second
frequency band (105b) for a time portion (T) of
the audio signal (105);
deriving energy distribution data (125), the
energy distribution data (125) characterizing
an energy distribution in a spectrum of the
time portion (T) of the audio signal (105); and
combining the noise floor data (115) and the
energy distribution data (125) to obtain the
bandwidth extension output data (102)); and
wherein the bandwidth extension data (375) comprise
the bandwidth extension output data (102) and envelope
data, and
bitstream payload formatting and outputting a coded
audio stream (345) by combining the bandwidth
extension data (375) with the encoded audio signal
(355), wherein the step of combining is part of the
step of bitstream payload formatting.
10. A bandwidth extension tool (430) for generating
components in a second frequency band (105b) of an
audio signal (105) based on bandwidth extension output
data (102) and based on a raw signal spectral
representation (425) for the components in the second
frequency band (105b), wherein the bandwidth extension
output data (102) comprise energy distribution data
(125), the energy distribution data (125)
characterizing an energy distribution in a spectrum of
a time portion (T) of the audio signal (105), the
bandwidth extension tool (430) comprising:
a noise floor modifier tool (433, 431), which is
configured to modify a transmitted noise floor in
accordance to the energy distribution data (125); and

a combiner (434) for combining the raw signal spectral
representation (425) with the modified noise floor to
generate the components in the second frequency bar.d
(105b) with the modified noise floor.
11. The bandwidth extension tool (430) of claim 10,
wherein the audio signal (105) comprises components in
a first frequency band (105a) and the bandwidth
extension parameter (102) comprise transmitted noise
floor data indicating a noise level for the noise
floor, and
wherein the noise floor modifier tool (433, 431) is
adapted
to increase the noise level in case the energy
distribution data (125) indicates an audio signal
(105) comprising more energy in the components of
the second frequency band (105b) than in first
frequency band (105a), or
to decrease the noise level in case the energy
distribution data (125) indicates an audio signal
(105) comprising more energy in the components of
the first frequency band (105a) than in the
second frequency band (105b).
12. A decoder (4C0) for decoding a coded audio stream
(345) to obtain an audio signal (105) comprising:
a bitstream deformatter (357) separating an encoded
signal (355) and the BWE output data (102);
a bandwidth extension tool (43C) of claim 10 or clairr.
11;

a core decoder (360) for decoding components in a
first frequency band (105a) from the encoded audio
signal (355); and
a synthesis unit (440) for synthesizing the audio
signal (10b) by combining the components of the first
and second frequency band (105a, 105b).
13. A method for decoding a coded audio stream (345) to
obtain an audio signal (105), the audio signal (105)
comprising components in a first frequency band (105a)
and bandwidth extension output data (102), wherein the
bandwidth extension output data (102) comprise energy
distribution data (125) and noise floor data, the
energy distribution data (125) characterizing an
energy distribution in a spectrum of a time portion
(T) of the audio signal (105) , the method comprising:
separating from the coded audio stream (345) an
encoded audio signal (355) and the BWE output data
(102);
decoding components in a first frequency band (105a)
from the encoded audio signal (355);
generating a raw signal spectral representation (425)
for components in a second frequency band (105b) from
the components in the first frequency band (105a);
modifying a noise floor in accordance to the energy
distribution data (125) and in accordance to the
transmitted noise floor data;
combining the raw signal spectral representation (425
with the modified noise floor to generate the
components in the second frequency band (105b) with
the calculated noise floor; and

synthesizing the audio signal (105) by combining the
components of the first and second frequency ban:;
(105a, 105b).
14. Computer program for performing, when running on a
computer, the method of claim 9 or claim 13.
15. An encoded audio stream (345), comprising:
an encoded audio signal (355) for components in a first
frequency band (105a) of an audio signal (1C5);
noise floor data adapted to control a synthesis of a
noise floor for components in a second frequency band
(105b) of the audio signal (105);
energy distribution data (125) adapted to control a
modification of the noise floor; and
envelope data (375) for the components in the second
frequency band (105b).

An apparatus (100) for generating bandwidth extension
output data (102) for an audio signal (105) comprises a
noise floor measurer (110), a signal energy characterizer
(120) and a processor (130). The audio signal (105)
comprises components in a first frequency band (105a) and
components in a second frequency band (105b), the bandwidth
extension output data (102) are adapted to control a
synthesis of the components in the second frequency band
(105b). The noise floor measurer (110) measures noise floor
data (115) of the second frequency band (105b) for a time
portion (T) of the audio signal (105) . The signal energy
characterizer (120) derives energy distribution data (125) ,
the energy distribution data (125) characterizing an energy
distribution in a spectrum of the time portion (T) of the
audio signal (105). The processor (130) combines the noise
floor data (115) and the energy distribution data (125) to
obtain the bandwidth extension output data (102).