Specification
The present invention relates to an apparatus and a method
for decoding an encoded audio signal, an apparatus for
encoding, a method for encoding and an audio signal.
In the art, frequency domain coding schemes such as MP3 or
AAC are known. These frequency-domain encoders are based on
a time-domain/frequency-domain conversion, a subsequent
quantization stage, in which the quantization error is
controlled using information from a psychoacoustic module,
and an encoding stage, in which the quantized spectral
coefficients and corresponding side information are
entropy-encoded using code tables.
On the other hand there are encoders that are very well
suited to speech processing such as the AMR-WB+ as
described in 3GPP TS 26.290. Such speech coding schemes
perform a Linear Predictive filtering of a time-domain
signal. Such a LP filtering is derived from a Linear
Prediction analysis of the input time-domain signal. The
resulting LP filter coefficients are then coded and
transmitted as side information. The process is known as
Linear Prediction Coding (LPC). At the output of the
filter, the prediction residual signal or prediction error
signal which is also known as the excitation signal is
encoded using the analysis-by-synthesis stages of the ACELP
encoder or, alternatively, is encoded using a transform
encoder which uses a Fourier transform with an overlap. The
decision between the ACELP coding and the Transform Coded
excitation coding which is also called TCX coding is done
using a closed loop or an open loop algorithm.

Frequency-domain audio coding schemes such as the high
efficiency-AAC encoding scheme which combines an AAC coding
scheme and a spectral bandwidth replication technique, can
also be combined to a joint stereo or a multi-channel
coding tool which is known under the term "MPEG surround".
On the other hand, speech encoders such as the AMR-WB+ also
have a high frequency enhancement stage and a stereo
functionality.
Said spectral band replication (SBR) comprises a technique
that gained popularity as an add-on to popular perception
audio coded such as MP3 and the advanced audio coding
(AAC). SBR comprise a method of bandwidth extension (BWE)
in which the low band (base band or core band) of the
spectrum is encoded using an existing coding, whereas as
the upper band (or high band) is coarsely parameterized
using fewer parameters. SBR makes use of a correlation
between the low band and the high band in order to predict
the high band signal from extracting lower band features.
SBR is, for example, used in HE-AAC or AAC+SBR. In SBR it
is possible to dynamically change the crossover frequency
(BWE start frequency) as well as the temporal resolution
meaning the number of parameter sets (envelopes) per frame.
AMR-WB+ implements a time domain bandwidth extension in
combination with a switched time/frequency domain core
coder, giving good audio quality especially for speech
signals. A limiting factor to AMR-WB+ audio quality is the
audio bandwidth common to both core codecs and BWE start
frequency that is one quarter of the system's internal
sampling frequency. While the ACELP speech model is capable
to model speech signals quite well over the full bandwidth,
the frequency domain audio coder fails to deliver decent
quality for some general audio signals. Thus, speech coding
schemes show a high quality for speech signals even at low
bit rates, but show a poor quality for music signals at low
bit rates.

Frequency-domain coding schemes such as HE-AAC are
advantageous in that they show a high quality at low bit
rates for music signals. Problematic, however, is the
quality of speech signals at low bit rates.
Therefore, different classes of audio signal demand
different characteristics of bandwidth extension tool.
It is the object of the present invention to provide an
improved encoding/decoding concept.
This object is achieved by an audio decoder in accordance
with claim 1, a method of audio decoding in accordance with
claim 13, an encoder in accordance with claim 8, a method
for encoding in accordance with claim 14, an encoded signal
in accordance with claim 15 or a computer program in
accordance with claim 16.
The present invention is based on the finding that the
crossover frequency or the BWE start frequency is a
parameter influencing the audio quality. While time domain
(speech) codecs usually code the whole frequency range for
a given sampling rate, audio bandwidth is a tuning
parameter to transform-based coders (e.g. coders for
music), as decreasing the total number of spectral lines to
encode will at the same time increase the number of bits
per spectral line available for encoding, meaning a quality
versus audio bandwidth trade-off is made. Hence, in the new
approach, different core coders with variable audio
bandwidths are combined to a switched system with one
common BWE module, wherein the BWE module has to account
for the different audio bandwidths.
A straightforward way would be to find the lowest of all
core coder bandwidths and use this as BWE start frequency,
but this would deteriorate the perceived audio quality.
Also, the coding efficiency would be reduced, because in
time sections where a core coder is active which has a

higher bandwidth than the BWE start frequency, some
frequency regions would be represented twice, by the core
coder as well as the BWE which introduces redundancy. A
better solution is therefore to adapt the BWE start
frequency to the audio bandwidth of the core coder used.
Therefore according to embodiments of the present invention
an audio coding system combines a bandwidth extension tool
with a signal dependent core coder (for example switched
speech-/audio coder), wherein the crossover frequency
comprise a variable parameter. A signal classifier output
that controls the switching between different core coding
modes may also be used to switch the characteristics of the
BWE system such as the temporal resolution and smearing,
spectral resolution and the crossover frequency.
Therefore, one aspect of the present invention is an audio
decoder for an encoded audio signal, the encoded audio
signal comprising a first portion encoded in accordance
with a first encoding algorithm, a second portion encoded
in accordance with a second encoding algorithm, BWE
parameters for the first portion and the second portion and
a coding mode information indicating a first decoding
algorithm or a second decoding algorithm, comprising a
first decoder, a second decoder, a BWE module and a
controller. The first decoder decodes the first portion in
accordance with the first decoding algorithm for a first
time portion of the encoded signal to obtain a first
decoded signal. The second decoder decodes the second
portion in accordance with the second decoding algorithm
for a second time portion of the encoded signal to obtain a
second decoded signal. The BWE module has a controllable
crossover frequency and is configured for performing a
bandwidth extension algorithm using the first decoded
signal and the BWE parameters for the first portion, and
for performing a bandwidth extension algorithm using the
second decoded signal and the bandwidth extension parameter
for the second portion. The controller controls the

crossover frequency for the BWE module in accordance with
the coding mode information.
According to another aspect of the present invention, an
apparatus for encoding an audio signal comprises a first
and a second encoder, a decision stage and a BWE module.
The first encoder is configured to encode in accordance
with a first encoding algorithm, the first encoding
algorithm having a first frequency bandwidth. The second
encoder is configured to encode in accordance with a second
encoding algorithm, the second encoding algorithm having a
second frequency bandwidth being smaller than the first
frequency bandwidth. The decision stage indicates the first
encoding algorithm for a first portion of the audio signal
and the second encoding algorithm for a second portion of
the audio signal, the second portion being different from
the first portion. The bandwidth extension module
calculates BWE parameters for the audio signal, wherein the
BWE module is configured to be controlled by the decision
stage to calculate the BWE parameters for a band not
including the first frequency bandwidth in the first
portion of the audio signal and for a band not including
the second frequency bandwidth in the second portion of the
audio signal.
In contrast to embodiments, SBR in prior art is applied to
a non-switch audio codec only which results in the
following disadvantages. Both temporal resolution as well
as crossover frequency could be applied dynamically, but
state of art implementations such as 3GPP source apply
usually only a change of temporary resolution for
transients as, for example, castanets. Furthermore, a finer
overall temporal resolution might be chosen at higher rates
as a bit rate dependent tuning parameter. No explicit
classification is carried out determining the temporal
resolution or a decision threshold controlling the temporal
resolution, best matching the signal type as, for example,
stationary, tonal music versus speech. Embodiments of the

present invention overcome these disadvantages. Embodiments
allow especially an adapted crossover frequency combined
with a flexible choice for the used core coder so that the
coded signal provides a significantly higher perceptual
quality compared to prior art encoder/decoder.
Brief description of the drawings
Preferred embodiments of the present invention are
subsequently described with respect to the attached
drawings, in which:
Fig. 1 shows a block diagram of an apparatus for
decoding in accordance with a first aspect of the
present invention;
Fig. 2 shows a block diagram of an apparatus for
encoding in accordance with the first aspect of
the present invention;
Fig. 3 shows a block diagram of an encoding scheme in
more details;
Fig. 4 shows a block diagram of a decoding scheme in
more details;
Fig. 5 shows a block diagram of an encoding scheme in
accordance with a second aspect;
Fig. 6 is a schematic diagram of a decoding scheme in
accordance with the second aspect;
Fig. 7 illustrates an encoder-side LPC stage providing
short-term prediction information and the
prediction error signal;

Fig. 8 illustrates a further embodiment of an LPC device
for generating a weighted signal;
Figs. 9a-9b show an encoder comprising an audio/speech-
switch resulting in different temporal resolution
for an audio signal; and
Fig. 10 illustrates a representation for an encoded audio
signal.
Detailed description of the invention
Fig. 1 shows a decoder apparatus 100 for decoding an
encoded audio signal 102. The encoded audio signal 102
comprising a first portion 104a encoded in accordance with
the first encoding algorithm, a second portion 104b encoded
in accordance with a second encoding algorithm, BWE
parameter 106 for the first time portion 104a and the
second time portion 104b and a coding mode information 108
indicating a first decoding algorithm or a second decoding
algorithm for the respective time portions. The apparatus
for decoding 100 comprises a first decoder 110a, a second
decoder 110b, a BWE module 130 and a controller 140. The
first decoder 110a is adapted to decode the first portion
104a in accordance with the first decoding algorithm for a
first time portion of the encoded signal 102 to obtain a
first decoded signal 114a. The second decoder 110b is
configured to decode the second portion 104b in accordance
with the second decoding algorithm for a second time
portion of the encoded signal to obtain a second decoded
signal 114b. The BWE module 130 has a controllable
crossover frequency fx that adjusts the behavior of the BWE
module 130. The BWE module 130 is configured to perform a
bandwidth extension algorithm to generate components of the
audio signal in the upper frequency band based on the first
decoded signal 114a and the BWE parameters 106 for the
first portion, and to generate components of the audio

signal in the upper frequency band based on the second
decoded signal 114b and the bandwidth extension parameter
106 for the second portion. The controller 140 is
configured to control the crossover frequency fx of the BWE
module 130 in accordance with the coding mode information
108.
The BWE module 130 may comprise also a combiner combining
the audio signal components of lower and the upper
frequency band and outputs the resulting audio signal 105.
The coding mode information 108 indicates, for example
which time portion of the encoded audio signal 102 is
encoded by which encoding algorithm. This information may
at the same time identify the decoder to be used for the
different time portions. In addition, the coding mode
information 108 may control a switch to switch between
different decoders for different time portions.
Hence, the crossover frequency fx is an adjustable
parameter which is adjusted in accordance with the used
decoder which may, for example, comprise a speech coder as
the first decoder 110a and an audio decoder as the second
decoder 110b. As said above, the crossover frequency fx for
a speech decoder (as for example based on LPC) may be
higher than the crossover frequency used for an audio
decoder (e.g. for music). Thus, in further embodiments the
controller 220 is configured to increase the crossover
frequency fx or to decrease the crossover frequency fx
within one of the time portion (e.g. the second time
portion) so that the crossover frequency may be changed
without changing the decoding algorithm. This means that a
change in the crossover frequency may not be related to a
change in the used decoder: the crossover frequency may be
changed without changing the used decoder or vice versa the
decoder may be changed without changing the crossover
frequency.

The BWE module 130 may also comprise a switch which is
controlled by the controller 140 and/or by the BWE
parameter 106 so that the first decoded signal 114a is
processed by the BWE module 130 during the first time
portion and the second decoded signal 114b is processed by
the BWE module 130 during the second time portion. This
switch may be activated by a change in the crossover
frequency fx or by an explicit bit within the encoded audio
signal 102 indicating the used encoding algorithm during
the respective time portion.
In further embodiments the switch is configured to switch
between the first and second time portion from the first
decoder to the second decoder so that the bandwidth
extension algorithm is either applied to the first decoded
signal or to the second decoded signal. Alternatively, the
bandwidth extension algorithm is applied to the first
and/or to second decoded signal and the switch is placed
after this so that one of the bandwidth extended signals is
dropped.
Fig. 2 shows a block diagram for an apparatus 200 for
encoding an audio signal 105. The apparatus for encoding
200 comprises a first encoder 210a, a second encoder 210b,
a decision stage 220 and a bandwidth extension module (BWE
module) 230. The first encoder 210a is operative to encode
in accordance with a first encoding algorithm having a
first frequency bandwidth. The second encoder 210b is
operative to encode in accordance with a second encoding
algorithm having a second frequency bandwidth being smaller
than the first frequency bandwidth. The first encoder may,
for example, be a speech coder such as an LPC-based coder,
whereas the second encoder 210b may comprise an audio
(music) encoder. The decision stage 220 is configured to
indicate the first encoding algorithm for a first portion
204a of the audio signal 105 and to indicate the second
encoding algorithm for a second portion 204b of the audio
signal 105, wherein the second time portion being different

from the first time portion. The first portion 204a may
correspond to a first time portion and the second portion
204b may correspond to a second time portion which is
different from the first time portion.
The BWE module 230 is configured to calculate BWE
parameters 106 for the audio signal 105 and is configured
to be controlled by the decision stage 220 to calculate the
BWE parameter 106 for a first band not including the first
frequency bandwidth in the first time portion 204a of the
audio signal 105. The BWE module 230 is further configured
to calculate the BWE parameter 106 for a second band not
including the second bandwidth in the second time portion
204b of the audio signal 105. The first (second) band
comprises hence frequency components of the audio signal
105 which are outside the first (second) frequency
bandwidth and are limited towards the lower end of the
spectrum by the crossover frequency fx. The first or the
second bandwidth can therefore be defined by a variable
crossover frequency which is controlled by the decision
stage 220.
In addition, the BWE module 230 may comprise a switch
controlled by the decision stage 220. The decision stage
220 may determine a preferred coding algorithm for a given
time portion and controls the switch so that during the
given time portion the preferred coder is used. The
modified coding mode information 108' comprises the
corresponding switch signal. Moreover, the BWE module 230
may also comprise a filter to obtain components of the
audio signal 105 in the lower/upper frequency band which
are separated by the crossover frequency fx which may
comprise a value of about 4 kHz or 5 kHz. Finally the BWE
module 130 may also comprise an analyzing tool to determine
the BWE parameter 106. The modified coding mode information
108' may be equivalent (or equal) to the coding mode
information 108. The coding mode information 108 indicates,
for example, the used coding algorithm for the respective

time portions in the bitstream of the encoded audio signal
105.
According to further embodiments, the decision stage 220
comprises a signal classifier tool which analyzes the
original input signal 105 and generates the control
information 108 which triggers the selection of the
different coding modes. The analysis of the input signal
105 is implementation dependent with the aim to choose the
optimal core coding mode for a given input signal frame.
The output of the signal classifier can (optionally) also
be used to influence the behavior of other tools, for
example, MPEG surround, enhanced SBR, time-warped
filterbank and others. The input to the signal classifier
tool comprises, for example, the original unmodified input
signal 105, but also optionally additional implementation
dependent parameters. The output of the signal classifier
tool comprises the control signal 108 to control the
selection of the core codec (for example non-LP filtered
frequency domain or LP filtered time or frequency domain
coding or further coding algorithms).
According to embodiments, the crossover frequency fx is
adjusted signal dependent which is combined with the
switching decision to use a different coding algorithm.
Therefore, a simple switch signal may simply be a change (a
jump) in the crossover frequency fx. In addition, the
coding mode information 108 may also comprise the change of
the crossover frequency fx indicating at the same time a
preferred coding scheme (e.g. speech/audio/music).
According to further embodiments the decision stage 220 is
operative to analyze the audio signal 105 or a first output
of he first encoder 210a or a second output of the second
encoder 210b or a signal obtained by decoding an output
signal of the encoder 210a or the second encoder 210b with
respect to a target function. The decision stage 220 may
optionally be operative to perform a speech/music

discrimination in such a way that a decision to speech is
favored with respect to a decision to music so that a
decision to speech is taken, e.g., even when a portion less
than 50% of a frame for the first switch is speech and a
portion more than 50% of the frame for the first switch is
music. Therefore, the decision stage 220 may comprise an
analysis tool that analyses the audio signal to decide
whether the audio signal is mainly a speech signal or
mainly a music signal so that based on the result the
decision stage can decide which is the best codec to be
used for the analysed time portion of the audio signal.
Figs. 1 and 2 do not show many of these details for the
encoder/decoder. Possible detailed examples for the
encoder/decoder are shown in the following figures. In
addition to the first and second decoder 110a,b of Fig. 1
further decoders may be present which may or may not use
e.g. further encoding algorithms. In the same way, also the
encoder 200 of Fig. 2 may comprise additional encoders
which may use additional encoding algorithms. In the
following the example with two encoders/decoders will be
explained in more detail.
Fig. 3 illustrates in more details an encoder having two
cascaded switches. A mono signal, a stereo signal or a
multi-channel signal is input into a decision stage 220 and
into a switch 232 which is part of the BWE module 230 of
Fig. 2. The switch 232 is controlled by the decision stage
220.. Alternatively, the decision stage 220 may also
receive a side information which is included in the mono
signal, the stereo signal or the multi-channel signal or is
at least associated to such a signal, where information is
existing, which was, for example, generated when originally
producing the mono signal, the stereo signal or the multi-
channel signal.
The decision stage 220 actuates the switch 232 in order to
feed a signal either in a frequency encoding portion 210b

illustrated now at an upper branch of Fig. 3 or an LPC-
domain encoding portion 210a illustrated at a lower branch
in Fig. 3. A key element of the frequency domain encoding
branch is a spectral conversion block 410 which is
operative to convert a common preprocessing stage output
signal (as discussed later on) into a spectral domain. The
spectral conversion block may include an MDCT algorithm, a
QMF, an FFT algorithm, a Wavelet analysis or a filterbank
such as a critically sampled filterbank having a certain
number of filterbank channels, where the subband signals in
this filterbank may be real valued signals or complex
valued signals. The output of the spectral conversion block
410 is encoded using a spectral audio encoder 421 which may
include processing blocks as known from the AAC coding
scheme.
Generally, the processing in branch 210b is a processing
based on a perception based model or information sink
model. Thus, this branch models the human auditory system
receiving sound. Contrary thereto, the processing in branch
210a is to generate a signal in the excitation, residual or
LPC domain. Generally, the processing in branch 210a is a
processing based on a speech model or an information
generation model. For speech signals, this model is a model
of the human speech/sound generation system generating
sound. If, however, a sound from a different source
requiring a different sound generation model is to be
encoded, then the processing in branch 210a may be
different. In addition to the shown coding branches,
further embodiments comprise additional branches or core
coders. For example, different coders may optionally be
present for the different sources, so that sound from each
source may be coded by employing a preferred coder.
In the lower encoding branch 210a, a key element is an LPC
device 510 which outputs LPC information which is used for
controlling the characteristics of an LPC filter. This LPC
information is transmitted to a decoder. The LPC stage 510

output signal is an LPC-domain signal which consists of an
excitation signal and/or a weighted signal.
The LPC device generally outputs an LPC domain signal which
can be any signal in the LPC domain or any other signal
which has been generated by applying LPC filter
coefficients to an audio signal. Furthermore, an LPC device
can also determine these coefficients and can also
quantize/encode these coefficients.
The decision in the decision stage 220 can be signal-
adaptive so that the decision stage performs a music/speech
discrimination and controls the switch 232 in such a way
that music signals are input into the upper branch 210b,
and speech signals are input into the lower branch 210a. In
one embodiment, the decision stage 220 is feeding its
decision information into an output bit stream so that a
decoder can use this decision information in order to
perform the correct decoding operations. This decision
information may, for example, comprise the coding mode
information 108 which may also comprise information about
the crossover frequency fx or a change of the crossover
frequency fx.
Such a decoder is illustrated in Fig. 4. The signal output
of the spectral audio encoder 421 is, after transmission,
input into a spectral audio decoder 431. The output of the
spectral audio decoder 431 is input into a time-domain
converter 440 (the time-domain converter may in general be
a converter from a first to a second domain). Analogously,
the output of the LPC domain encoding branch 210a of Fig. 3
received on the decoder side and processed by elements 531,
533, 534, and 532 for obtaining an LPC excitation signal.
The LPC excitation signal is input into an LPC synthesis
stage 540 which receives, as a further input, the LPC
information generated by the corresponding LPC analysis
stage 510. The output of the time-domain converter 440
and/or the output of the LPC synthesis stage 540 are input

into a switch 132 which may be part of the BWE module 130
in Fig. 1. The switch 132 is controlled via a switch
control signal (such as the coding mode information 108
and/or the BWE parameter 106) which was, for example,
generated by the decision stage 220, or which was
externally provided such as by a creator of the original
mono signal, stereo signal or multi-channel signal.
In Fig. 3, the input signal into the switch 232 and the
decision stage 220 can be a mono signal, a stereo signal, a
multi-channel signal or generally any audio signal.
Depending on the decision which can be derived from the
switch 232 input signal or from any external source such as
a producer of the original audio signal underlying the
signal input into stage 232, the switch switches between
the frequency encoding branch 210b and the LPC encoding
branch 210a. The frequency encoding branch 210b comprises a
spectral conversion stage 410 and a subsequently connected
quantizing/coding stage 421. The quantizing/coding stage
can include any of the functionalities as known from modern
frequency-domain encoders such as the AAC encoder.
Furthermore, the quantization operation in the
quantizing/coding stage 421 can be controlled via a
psychoacoustic module which generates psychoacoustic
information such as a psychoacoustic masking threshold over
the frequency, where this information is input into the
stage 421.
In the LPC encoding branch 210a, the switch output signal
is processed via an LPC analysis stage 510 generating LPC
side info and an LPC-domain signal. The excitation encoder
may comprise an additional switch for switching the further
processing of the LPC-domain signal between a
quantization/coding operation 522 in the LPC-domain or a
quantization/coding stage 524 which is processing values in
the LPC-spectral domain. To this end, a spectral converter
523 is provided at the input of the quantizing/coding stage
524. The switch 521 is controlled in an open loop fashion

or a closed loop fashion depending on specific settings as,
for example, described in the AMR-WB+ technical
specification.
For the closed loop control mode, the encoder additionally
includes an inverse quantizer/coder 531 for the LPC domain
signal, an inverse quantizer/coder 533 for the LPC spectral
domain signal and an inverse spectral converter 534 for the
output of item 533. Both encoded and again decoded signals
in the processing branches of the second encoding branch
are input into the switch control device 525. In the switch
control device 525, these two output signals are compared
to each other and/or to a target function or a target
function is calculated which may be based on a comparison
of the distortion in both signals so that the signal having
the lower distortion is used for deciding, which position
the switch 521 should take. Alternatively, in case both
branches provide non-constant bit rates, the branch
providing the lower bit rate might be selected even when
the distortion or the perceptional distortion of this
branch is lower than the distortion or perceptional
distortion of the other branch (an example for the
distortion may be the signal to noise ratio).
Alternatively, the target function could use, as an input,
the distortion of each signal and a bit rate of each signal
and/or additional criteria in order to find the best
decision for a specific goal. If, for example, the goal is
such that the bit rate should be as low as possible, then
the target function would heavily rely on the bit rate of
the two signals output of the elements 531, 534. However,
when the main goal is to have the best quality for a
certain bit rate, then the switch control 525 might, for
example, discard each signal which is above the allowed bit
rate and when both signals are below the allowed bit rate,
the switch control would select the signal having the
better estimated subjective quality, i.e., having the
smaller quantization/coding distortions or a better signal
to noise ratio.

The decoding scheme in accordance with an embodiment is, as
stated before, illustrated in Fig. 4. For each of the three
possible output signal kinds, a specific decoding/re-
quantizing stage 431, 531 or 533 exists. While stage 431
outputs a frequency-spectrum which is converted into the
time-domain using the frequency/time converter 440, stage
531 outputs an LPC-domain signal, and item 533 outputs an
LPC-spectrum. In order to make sure that the input signals
into switch 532 are both in the LPC-domain, the LPC-
spectrum/LPC-converter 534 is provided. The output data of
the switch 532 is transformed back into the time-domain
using an LPC synthesis stage 540 which is controlled via
encoder-side generated and transmitted LPC information.
Then, subsequent to block 540, both branches have time-
domain information which is switched in accordance with a
switch control signal in order to finally obtain an audio
signal such as a mono signal, a stereo signal or a multi-
channel signal which depends on the signal input into the
encoding scheme of Fig. 3.
Figs. 5 and 6 show further embodiments for the
encoder/decoder, wherein the BWE stages as part of the BWE
modules 130, 230 represent a common processing unit.
Fig. 5 illustrates an encoding scheme, wherein the common
preprocessing scheme connected to the switch 232 input may
comprise a surround/joint stereo block 101 which generates,
as an output, joint stereo parameters and a mono output
signal which is generated by downmixing the input signal
which is a signal having two or more channels. Generally,
the signal at the output of block 101 can also be a signal
having more channels, but due to the downmixing
functionality of block 101, the number of channels at the
output of block 101 will be smaller than the number of
channels input into block 101.

The common preprocessing scheme may comprise in addition to
the block 101 a bandwidth extension stage 230. In the Fig.
5 embodiment, the output of block 101 is input into the
bandwidth extension block 230 which outputs a band-limited
signal such as the low band signal or the low pass signal
at its output. Preferably, this signal is downsampled (e.g.
by a factor of two) as well. Furthermore, for the high band
of the signal input into block 230, bandwidth extension
parameters 106 such as spectral envelope parameters,
inverse filtering parameters, noise floor parameters etc.
as known from HE-AAC profile of MPEG-4 are generated and
forwarded to a bitstream multiplexer 800.
Preferably, the decision stage 220 receives the signal
input into block 101 or input into block 230 in order to
decide between, for example, a music mode or a speech mode.
In the music mode, the upper encoding branch 210b (second
encoder in Fig. 2) is selected, while, in the speech mode,
the lower encoding branch 210a is selected. Preferably, the
decision stage additionally controls the joint stereo block
101 and/or the bandwidth extension block 230 to adapt the
functionality of these blocks to the specific signal. Thus,
when the decision stage 220 determines that a certain time
portion of the input signal corresponds to the first mode
such as the music mode, then specific features of block 101
and/or block 230 can be controlled by the decision stage
220. Alternatively, when the decision stage 220 determines
that the signal corresponds to a speech mode or, generally,
in a second LPC-domain mode, then specific features of
blocks 101 and 230 can be controlled in accordance with the
decision stage output. The decision stage 220 yields also
the control information 108 and/or the crossover frequency
fx which may also be transmitted to the BWE block 230 and,
in addition, to a bitstream multiplexer 800 so that it will
be transmitted to the decoder side.
Preferably, the spectral conversion of the coding branch
210b is done using an MDCT operation which, even more

preferably, is the time-warped MDCT operation, where the
strength or, generally, the warping strength can be
controlled between zero and a high warping strength. In a
zero warping strength, the MDCT operation in block 411 is a
straight-forward MDCT operation known in the art. The time
warping strength together with time warping side
information can be transmitted/input into the bitstream
multiplexer 800 as side information.
In the LPC encoding branch, the LPC-domain encoder may
include an ACELP core 526 calculating a pitch gain, a pitch
lag and/or codebook information such as a codebook index
and gain. The TCX mode as known from 3GPP TS 26.290
includes a processing of a perceptually weighted signal in
the transform domain. A Fourier transformed weighted signal
is quantized using a split multi-rate lattice quantization
(algebraic VQ) with noise factor quantization. A transform
is calculated in 1024, 512, or 256 sample windows. The
excitation signal is recovered by inverse filtering the
quantized weighted signal through an inverse weighting
filter. The TCX mode may also be used in modified form in
which the MDCT is used with an enlarged overlap, scalar
quantization, and an arithmetic coder for encoding spectral
lines.
In the "music" coding branch 210b, a spectral converter
preferably comprises a specifically adapted MDCT operation
having certain window functions followed by a
quantization/entropy encoding stage which may consist of a
single vector quantization stage, but preferably is a
combined scalar quantizer/entropy coder similar to the
quantizer/coder in the frequency domain coding branch,
i.e., in item 421 of Fig. 5.
In the "speech" coding branch 210a, there is the LPC block
510 followed by a switch 521, again followed by an ACELP
block 526 or a TCX block 527. ACELP is described in 3GPP TS
26.190 and TCX is described in 3GPP TS 26.290. Generally,

the ACELP block 526 receives an LPC excitation signal as
calculated by a procedure as described in Fig. 7. The TCX
block 527 receives a weighted signal as generated by Fig.
8.
At the decoder side illustrated in Fig. 6, after the
inverse spectral transform in block 537, the inverse of the
weighting filter is applied that is (1 - uz-1) / (1 - A(z / y) ) .
Then, the signal is filtered through (l-A(z)) to go to the
LPC excitation domain. Thus, the conversion to LPC domain
block 534 and the TCX-1 block 537 include inverse transform
and then filtering through (1-μz-1) (1-A(z) ) to convert
(1-A(z/Y))
from the weighted domain to the excitation domain.
Although item 510 in Figs. 3, 5 illustrates a single block,
block 510 can output different signals as long as these
signals are in the LPC domain. The actual mode of block 510
such as the excitation signal mode or the weighted signal
mode can depend on the actual switch state. Alternatively,
the block 510 can have two parallel processing devices,
where one device is implemented similar to Fig. 7 and the
other device is implemented as Fig. 8. Hence, the LPC
domain at the output of 510 can represent either the LPC
excitation signal or the LPC weighted signal or any other
LPC domain signal.
In the second encoding branch (ACELP/TCX) of Fig. 5, the
signal is preferably pre-emphasized through a filter 1-μz-1
before encoding. At the ACELP/TCX decoder in Fig. 6 the
synthesized signal is deemphasized with the filter
1/(1-μz-1). In a preferred embodiment, the parameter μ has
the value 0.68 . The preemphasis can be part of the LPC
block 510 where the signal is preemphasized before LPC
analysis and quantization. Similarly, deemphasis can be
part of the LPC synthesis block LPC-1 540.

Fig. 6 illustrates a decoding scheme corresponding to the
encoding scheme of Fig. 5. The bitstream generated by
bitstream multiplexer 800 (or output interface) of Fig. 5
is input into a bitstream demultiplexer 900 (or input
interface). Depending on an information derived for example
from the bitstream via a mode detection block 601 (e.g.
part of the controller 140 in Fig. 1) , a decoder-side
switch 132 is controlled to either forward signals from the
upper branch or signals from the lower branch to the
bandwidth extension block 701. The bandwidth extension
block 701 receives, from the bitstream demultiplexer 900,
side information and, based on this side information and
the output of the mode detection 601, reconstructs the high
band based on the low band output by switch 132. The
control signal 108 controls the used crossover frequency
fx.
The full band signal generated by block 701 is input into
the joint stereo/surround processing stage 702 which
reconstructs two stereo channels or several multi-channels.
Generally, block 702 will output more channels than were
input into this block. Depending on the application, the
input into block 702 may even include two channels such as
in a stereo mode and may even include more channels as long
as the output of this block has more channels than the
input into this block.
The switch 232 in Fig. 5 has been shown to switch between
both branches so that only one branch receives a signal to
process and the other branch does not receive a signal to
process. In an alternative embodiment, however, the switch
232 may also be arranged subsequent to for example the
audio encoder 421 and the excitation encoder 522, 523, 524,
which means that both branches 210a, 210b process the same
signal in parallel. In order to not double the bitrate,
however, only the signal output of one of those encoding
branches 210a or 210b is selected to be written into the
output bitstream. The decision stage will then operate so

that the signal written into the bitstream minimizes a
certain cost function, where the cost function can be the
generated bitrate or the generated perceptual distortion or
a combined rate/distortion cost function. Therefore, either
in this mode or in the mode illustrated in the Figures, the
decision stage can also operate in a closed loop mode in
order to make sure that, finally, only the encoding branch
output is written into the bitstream which has for a given
perceptual distortion the lowest bitrate or, for a given
bitrate, has the lowest perceptual distortion. In the
closed loop mode, the feedback input may be derived from
outputs of the three quantizer/sealer blocks 421, 522 and
424 in Fig. 3.
Also in the embodiment of Fig. 6, the switch 132 may in
alternative embodiments be arranged after the BWE module
701 so that the bandwidth extension is performed in
parallel for both branches and the switch selects one of
the two bandwidth extended signals.
In the implementation having two switches, i.e., the first
switch 232 and the second switch 521, it is preferred that
the time resolution for the first switch is lower than the
time resolution for the second switch. Stated differently,
the blocks of the input signal into the first switch which
can be switched via a switch operation are larger than the
blocks switched by the second switch 521 operating in the
LPC-domain. Exemplarily, the frequency domain/LPC-domain
switch 232 may switch blocks of a length of 1024 samples,
and the second switch 521 can switch blocks having 256
samples each.
Fig. 7 illustrates a more detailed implementation of the
LPC analysis block 510. The audio signal is input into a
filter determination block 83 which determines the filter
information A(z). This information is output as the short-
term prediction information required for a decoder. The
short-term prediction information is required by the

actual prediction filter 85. In a subtracter 86, a current
sample of the audio signal is input and a predicted value
for the current sample is subtracted so that for this
sample, the prediction error signal is generated at line
84.
While Fig. 7 illustrates a preferred way to calculate the
excitation signal, Fig. 8 illustrates a preferred way to
calculate the weighted signal. In contrast to Fig. 7, the
filter 85 is different, when γ is different from 1. A
value smaller than 1 is preferred for y. Furthermore, the
block 87 is present, and μ is preferable a number smaller
than 1. Generally, the elements in Fig. 7 and 8 can be
implemented as in 3GPP TS 26.190 or 3GPP TS 26.290.
Subsequently, an analysis-by-synthesis CELP encoder is
discussed in order to illustrate the modifications applied
to this algorithm. This CELP encoder is discussed in
detail in "Speech Coding: A Tutorial Review", Andreas
Spanias, Proceedings of the IEEE, Vol. 82, No. 10, October
1994, pages 1541-1582.
For specific cases, when a frame is a mixture of unvoiced
and voiced speech or when speech over music occurs, a TCX
coding can be more appropriate to code the excitation in
the LPC domain. The TCX coding processes directly the
excitation in the frequency domain without doing any
assumption of excitation production. The TCX is then more
generic than CELP coding and is not restricted to a voiced
or a non-voiced source model of the excitation. TCX is
still a source-filter model coding using a linear
predictive filter for modelling the formants of the
speech-like signals.
In the AMR-WB+-like coding, a selection between different
TCX modes and ACELP takes place as known from the AMR-WB+
description. The TCX modes are different in that the
length of the block-wise Fast Fourier Transform is

different for different modes and the best mode can be
selected by an analysis by synthesis approach or by a
direct "feedforward" mode.
As discussed in connection with Fig. 5 and 6, the common
pre-processing stage 100 preferably includes a joint
multi-channel (surround/joint stereo device) 101 and,
additionally, a bandwidth extension stage 230.
Correspondingly, the decoder includes a bandwidth
extension stage 701 and a subsequently connected joint
multichannel stage 702. Preferably, the joint multichannel
stage 101 is, with respect to the encoder, connected
before the band width extension stage 230, and, on the
decoder side, the band width extension stage 701 is
connected before the joint multichannel stage 702 with
respect to the signal processing direction. Alternatively,
however, the common pre-processing stage can include a
joint multichannel stage without the subsequently
connected bandwidth extension stage or a bandwidth
extension stage without a connected joint multichannel
stage.
Figs. 9a to 9b show a simplified view on the encoder of
Fig. 5, where the encoder comprises the switch-decision
unit 220 and the stereo coding unit 101. In addition, the
encoder also comprises the bandwidth extension tools 230
as, for example, an envelope data calculator and SBR-
related modules. The switch-decision unit 220 provides a
switch decision signal 108' that switches between the audio
coder 210b and the speech coder 210a. The speech coder 210a
may further be divided into a voiced and unvoiced coder.
Each of these coders may encode the audio signal in the
core frequency band using different numbers of sample
values (e.g. 1024 for a higher resolution or 256 for a
lower resolution). The switch decision signal 108' is also
supplied to the bandwidth extension (BWE) tool 230. The BWE
tool 230 will then use the switch decision 108' in order,
for example, to adjust the number of the spectral envelopes

104 and to turn on/off an optional transient detector and
adjust the crossover frequency fx. The audio signal 105 is
input into the switch-decision unit 220 and is input into
the stereo coding 101 so that the stereo coding 101 may
produce the sample values which are input into the
bandwidth extension unit 230. Depending on the decision
108' generated by the switch-unit decision unit 220, the
bandwidth extension tool 230 will generate spectral band
replication data which are, in turn, forwarded either to an
audio coder 210b or a speech coder 210a.
The switch decision signal 108' is signal dependent and can
be obtained from the switch-decision unit 220 by analyzing
the audio signal, e.g., by using a transient detector or
other detectors which may or may not comprise a variable
threshold. Alternatively, the switch decision signal 108'
may be adjusted manually (e.g. by a user) or be obtained
from a data stream (included in the audio signal).
The output of the audio coder 210b and the speech coder
210a may again be input into the bitstream formatter 800
(see Fig. 5) .
Fig. 9b shows an example for the switch decision signal
108' which detects an audio signal for a time period before
a first time ta and after a second time tb. Between the
first time ta and the second time tb, the switch-decision
unit 220 detects a speech signal resulting in different
discrete values for the switch decision signal 108'.
The decision to use a higher crossover frequency fx is
controlled by the switching decision unit 220. This means
that the described method is also usable within a system in
which the SBR module is combined with only a single core
coder and a variable crossover frequency fx.
Although some of the Figs. 1 through 9 are illustrated as
block diagrams of an apparatus, these figures

simultaneously are an illustration of a method, where the
block functionalities correspond to the method steps.
Fig. 10 illustrates a representation for an encoded audio
signal 102 comprising the first portion 104a, the second
portion 104b, a third portion 104c and a fourth portion
104d. In this representation the encoded audio signal 102
is a bitstream transmitted over a transmission channel
which comprises furthermore the coding mode information
108. Each portion 104 of the encoded audio signal 102 may-
represent a different time portion, although different
portions 104 may be in the frequency as well as time domain
so that the encoded audio signal 102 may not represent a
time line.
In this embodiment the encoded audio signal 102 comprises
in addition a first coding mode information 108a
identifying the used coding algorithm for the first portion
104a; a second coding mode information 108b identifying the
used coding algorithm for the second portion 104b; a third
coding mode information 108d identifying the used coding
algorithm for the fourth portion 104d. The first coding
mode information 108a may also identify the used first
crossover frequency fx1 within the first portion 104a, and
the second coding mode information 108b may also identify
the used second crossover frequency fx2 within the second
portion 104b. For example, within the first portion 104a
the "speech" coding mode may be used and within the second
portion 104b the "music" coding mode may be used so that
the first crossover frequency fxl may be higher than the
second crossover frequency fx2.
In this exemplary embodiment the encoded audio signal 102
comprises no coding mode information for the third portion
104c which indicates that there is no change in the used
encoder and/or crossover frequency fx between the first and
third portion 104a, c. Therefore, the coding mode
information 108 may appear as header only for those

portions 104 which use a different core coder and/or
crossover frequency compared to the preceding portion. In
further embodiments instead of signaling the values of the
crossover frequencies for the different portions 104, the
code mode information 108 may comprise a single bit
indicating the core coder (first or second encoder 210a,b)
used for the respective portion 104.
Therefore, the signaling of the switch behavior between the
different SBR-tools can be done by submitting, for example,
as specific bit within the bitstream, so that this specific
bit may turn on or off a specific behavior in the decoder.
Alternatively, in systems with two core coders according to
embodiments the signaling of the switch may also be
initiated by analyzing the core codec. In this case the
submission of the adaptation of the SBR tools is done
implicitly, that means it is determined by the
corresponding core coder activity.
More details about the standard description of the
bitstream elements for the SBR payload can be found in
ISO/IEC 14496-3, sub-clause 4.5.2.8. A modification of this
standard bitstream comprises an extension of the index to
the master frequency table (to identify the used crossover
frequency). The used index is coded, for example, with four
bits allowing the crossover band to be variable over a
range of 0 to 15 bands.
Embodiments of the present invention can hence be
summarized as follows. Different signals with different
time/frequency characteristics have different demands on
the characteristic on the bandwidth extension. Transient
signals (e.g. within a speech signal) need a fine temporal
resolution of the BWE and the crossover frequency fx (the
upper frequency border of the core coder) should be as high
as possible (e.g. 4 kHz or 5 kHz or 6 kHz). Especially in
voiced speech, a distorted temporal structure can decrease
perceived quality. Tonal signals need a stable reproduction

of spectral components and a matching harmonic pattern of
the reproduced high frequency portions. The stable
reproduction of tonal parts limits the core coder bandwidth
but it does not need a BWE with fine temporal but finer
spectral resolution. In a switched speech-/audio core coder
design, it is possible to use the core coder decision also
to adapt both the temporal and spectral characteristics of
the BWE as well as adapting the BWE start frequency
(crossover frequency) to the signal characteristics.
Therefore, embodiments provide a bandwidth extension where
the core coder decision acts as adaptation criterion to
bandwidth extension characteristics.
The signaling of the changed BWE start (crossover)
frequency can be realized explicitly by sending additional
information (as, for example, the coding mode information
108) in the bitstream or implicitly by deriving the
crossover frequency fx directly from the core coder used
(in case the core coder is, e. g., signaled within the
bitstream) . For example, a lower BWE frequency fx for the
transform coder (for example audio/music coder) and a
higher for a time domain (speech) coder. In this case, the
crossover frequency may be in the range between 0 Hz up to
the Nyquist frequency.
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 of 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 performed 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, for
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 apparatus for decoding (100) an encoded audio
signal (102), the encoded audio signal (102)
comprising a first portion (104a) encoded in
accordance with a first encoding algorithm, a second
portion (104b) encoded in accordance with a second
encoding algorithm, BWE parameters (106) for the first
portion (104a) and the second portion (104b) and a
coding mode information (108) indicating a first
decoding algorithm or a second decoding algorithm,
comprising:
a first decoder (110a) for decoding the first portion
(104a) in accordance with the first decoding algorithm
for a first time portion of the encoded signal (102)
to obtain a first decoded signal (114a), wherein the
first decoder (110a) comprises an LPC-based coder;
a second decoder (110b) for decoding the second
portion (104b) in accordance with the second decoding
algorithm for a second time portion of the encoded
signal (102) to obtain a second decoded signal (114b),
wherein the second decoder (110b) comprises a
transform-based coder;
a BWE module (130) having a controllable crossover
frequency (fx), the BWE module (130) being configured
for performing a bandwidth extension algorithm using
the first decoded signal (114a) and the BWE parameters
(106) for the first portion (104a), and for performing
a bandwidth extension algorithm using the second
decoded signal (114b) and the bandwidth extension
parameter (106) for the second portion (104b),
wherein the BWE module (130) is configured to use a
first crossover frequency (fx1) for the bandwidth

extension for the first decoded signal (114a) and to
use a second crossover frequency (fx2) for the
bandwidth extension for the second decoded signal
(114b), wherein the first crossover frequency (fx1) is
higher than the second crossover frequency (fx2); and
a controller (140) for controlling the crossover
frequency (fx) for the BWE module (130) in accordance
with the coding mode information (108).
2. The apparatus for decoding (100) of claim 1, further
comprising an input interface (900) for inputting the
encoded audio signal (102) as a bitstream.
3. The apparatus for decoding (100) of claim 1 or of
claim 2, wherein the BWE module (130) comprises a
switch (132) which is configured to switch between the
first and second time portion from the first decoder
(110a) to the second decoder (110b) so that the
bandwidth extension algorithm is either applied to the
first decoded signal (114a) or to the second decoded
signal (114b).
4. The apparatus for decoding (100) of claim 3, wherein
the controller (140) is configured to control the
switch (132) dependent on the indicated decoding
algorithm within the coding mode information (108).
5. The apparatus for decoding (100) of one of the
preceding claims, wherein the controller (140) is
configured to increase the crossover frequency (fx)
within the first time portion or to decrease the
crossover frequency (fx) within the second time
portion.
6. An apparatus for encoding (200) an audio signal (105)
comprising:

a first encoder (210a) which is configured to encode
in accordance with a first encoding algorithm, the
first encoding algorithm having a first frequency
bandwidth, wherein the first encoder (210a) comprises
an LPC-based coder;
a second encoder (210b) which is configured to encode
in accordance with a second encoding algorithm, the
second encoding algorithm having a second frequency
bandwidth being smaller than the first frequency
bandwidth, wherein the second encoder (210b) comprises
a transform-based coder;
a decision stage (220) for indicating the first
encoding algorithm for a first portion (204a) of the
audio signal (105) and for indicating the second
encoding algorithm for a second portion (204b) of the
audio signal (105), the second portion (204b) being
different from the first portion (204a); and
a bandwidth extension module (230) for calculating BWE
parameters (106) for the audio signal (105), wherein
the BWE module (230) is configured to be controlled by
the decision stage (220) to calculate the BWE
parameters (106) for a band not including the first
frequency bandwidth in the first portion (204a) of the
audio signal (105) and for a band not including the
second frequency bandwidth in the second portion
(204b) of the audio signal (105),
wherein the first or the second frequency bandwidth is
defined by a variable crossover frequency (fx) and
wherein the decision stage (220) is configured to
output the variable crossover frequency (fx),
wherein the BWE module (230) is configured to use a
first crossover frequency (fx1) for calculating the
BWE parameters for a signal encoded using the first

encoder (210a) and to use a second crossover frequency
(fx2) for a signal encoded using the second encoder
(210b), wherein the first crossover frequency (fx1) is
higher than the second crossover frequency (fx2).
7. The apparatus for encoding (200) of claim 6, further
comprising an output interface (800) for outputting
the encoded audio signal (102), the encoded audio
signal (102) comprising a first portion (104a) encoded
in accordance with a first encoding algorithm, a
second portion (104b) encoded in accordance with a
second encoding algorithm, BWE parameters (106) for
the first portion (104a) and the second portion (104b)
and coding mode information (108) indicating the first
decoding algorithm or the second decoding algorithm.
8. The apparatus for encoding (200) of claim 6 or claim
7, wherein the first or the second frequency bandwidth
is defined by a variable crossover frequency (fx) and
wherein the decision stage (220) is configured to
output the variable crossover frequency (fx).
9. The apparatus for encoding (200) of one of the claims
6 to 8, wherein the BWE module (230) comprises a
switch (232) controlled by the decision stage (220),
wherein the switch (232) is configured to switch
between the first and second time encoder (210a, 210b)
so that the audio signal (105) is for different time
portions either encoded by the first or by the second
encoder (210a, 210b).
10. The apparatus for encoding (200) of one of the claims
6 to 9, wherein the decision stage (220) is operative
to analyze the audio signal (105) or a first output of
the first encoder (210a) or a second output of the
second encoder (210b) or a signal obtained by decoding
an output signal of the first encoder (210a) or the

second encoder (210b) with respect to a target
function.
11. A method for decoding an encoded audio signal (102),
the encoded audio signal (102) comprising a first
portion (104a) encoded in accordance with a first
encoding algorithm, a second portion (104b) encoded in
accordance with a second encoding algorithm, BWE
parameters (106) for the first portion (104a) and the
second portion (104b) and a coding mode information
(108) indicating a first decoding algorithm or a
second decoding algorithm, the method comprising:
decoding the first portion (104a) in accordance with
the first decoding algorithm for a first time portion
of the encoded signal (102) to obtain a first decoded
signal (114a), wherein the step of decoding the first
portion comprises using an LPC-based coder;
decoding the second portion (104b) in accordance with
the second decoding algorithm for a second time
portion of the encoded signal (102) to obtain a second
decoded signal (114b), wherein the step of decoding
the second portion (104b) comprises using a transform-
based coder;
performing a bandwidth extension algorithm by a BWE
module (130) having a controllable crossover frequency
(fx) , using the first decoded signal (114a) and the
BWE parameters (106) for the first portion (104a), and
performing, by the BWE module (130) having the
controllable crossover frequency (fx), a bandwidth
extension algorithm using the second decoded signal
(114b) and the bandwidth extension parameter (106) for
the second portion (104b),
wherein a first crossover frequency (fx1) is used for
the bandwidth extension for the first decoded signal

(114a) and a second crossover frequency (fx2) is used
for the bandwidth extension for the second decoded
signal (114b), wherein the first crossover frequency
(fx1) is higher than the second crossover frequency
(fx2); and
controlling the crossover frequency (fx) for the BWE
module (130) in accordance with the coding mode
information (108).
12. A method for encoding an audio signal (105)
comprising:
encoding in accordance with a first encoding
algorithm, the first encoding algorithm having a first
frequency bandwidth, wherein the step of encoding in
accordance with a first encoding algorithm comprises
using an LPC-based coder;
encoding in accordance with a second encoding
algorithm, the second encoding algorithm having a
second frequency bandwidth being smaller than the
first frequency bandwidth, wherein the step of
encoding in accordance with a second encoding
algorithm comprises using a transform-based coder;
indicating the first encoding algorithm for a first
portion (204a) of the audio signal (105) and the
second encoding algorithm for a second portion (204b)
of the audio signal (105), the second portion (204b)
being different from the first portion (204a); and
calculating BWE parameters (106) for the audio signal
(105) such that the BWE parameters (106) are
calculated for a band not including the first
frequency bandwidth in the first portion (204a) of the
audio signal (105) and for a band not including the

second frequency bandwidth in the second portion
(204b) of the audio signal (105),
wherein the first or the second frequency bandwidth is
defined by a variable crossover frequency (fx),
wherein the BWE module (230) is configured to use a
first crossover frequency (fx1) for calculating the
BWE parameters for a signal encoded using the LPC-
based coder and to use a second crossover frequency
(fx2) for a signal encoded using the transform-based
coder (210b), wherein the first crossover frequency
(fxl) is higher than the second crossover frequency
(fx2).
13. An encoded audio signal (102) comprising:
a first portion (104a) encoded in accordance with a
first encoding algorithm, the first encoding algorithm
comprising an LPC-based coder;
a second portion (104b) encoded in accordance with a
second different encoding algorithm, the second
encoding algorithm comprising a transform-based coder;
bandwidth extension parameters (106) for the first
portion (104a) and the second portion (104b); and
a coding mode information (108) indicating a first
crossover frequency (fx1) used for the first portion
(104a) or a second crossover frequency (fx2) used for
the second portion (104b), wherein the first crossover
frequency (fx1) is higher than the second crossover
frequency (fx2).
14. Computer program for performing, when running on a
computer, the method of claim 13 or claim 14.

An apparatus for decoding (100) an encoded audio signal
(102), the encoded audio signal (102) comprising a first
portion (104a) encoded in accordance with a first encoding
algorithm, a second portion (104b) encoded in accordance
with a second encoding algorithm, BWE parameters (106) for
the first portion (104a) and the second portion (104b) and
a coding mode information (108) indicating a first decoding
algorithm or a second decoding algorithm, comprises a first
decoder (110a), a second decoder (110b), a BWE module (130)
and a controller (140). The first decoder (110a) decodes
the first portion (104a) in accordance with the first
decoding algorithm for a first time portion of the encoded
signal (102) to obtain a first decoded signal (114a). The
second decoder (110b) decodes the second portion (104b) in
accordance with the second decoding algorithm for a second
time portion of the encoded signal (102) to obtain a second
decoded signal (114b). The BWE module (130) has a
controllable crossover frequency (fx) and is configured for
performing a bandwidth extension algorithm using the first
decoded signal (114a) and the BWE parameters (106) for the
first portion (104b), and for performing a bandwidth
extension algorithm using the second decoded signal (114b)
and the bandwidth extension parameter (106) for the second
portion (104b). The controller (140) controls the crossover
frequency (fx) for the BWE module (130) in accordance with
the coding mode information (108).