Apparatus and Method for Encoding/Decoding an Audio Signal
Using an Aliasing Switch Scheme
Description
The present invention is related to audio coding and,
particularly, to low bit rate audio coding schemes.
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 quantized/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 band replication technique can
also be combined with 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.
Frequency-domain coding schemes are advantageous in that
they show a high quality at low bitrates for music signals.
Problematic, however, is the quality of speech signals at
low bitrates.
Speech coding schemes show a high quality for speech
signals even at low bitrates, but show a poor quality for
music signals at low bitrates.
Frequency-domain coding schemes often make use of the so-
called MDCT (MDCT= modified discrete Cosine transform). The
MDCT has been initially described in J. Princen, A.
Bradley, "Analysis/Synthesis Filter Bank Design Based on
Time Domain Aliasing Cancellation", IEEE Trans. ASSP, ASSP-
34 (5) :1153-1161, 1986. The MDCT or MDCT filter bank is
widely used in modern and efficient audio coders. This kind
of signal processing provides the following advantages:
Smooth cross-fade between processing blocks: Even if the
signal in each processing block is altered differently
(e.g. due to quantization of spectral coefficients), no
blocking artifacts due to abrupt transitions from block to
block occur because of the windowed overlap/add operation.
Critical sampling: The number of spectral values at the
output of the filterbank is equal to the number of time
domain input values at its input and additional overhead
values have to be transmitted.

The MDCT filterbank provides a high frequency selectivity
and coding gain.
Those great properties are achieved by utilizing the
technique of time domain aliasing cancellation. The time
domain aliasing cancellation is done at the synthesis by
overlap-adding two adjacent windowed signals. If no
quantization is applied between the analysis and the
synthesis stages of the MDCT, a perfect reconstruction of
the original signal is obtained. However, the MDCT is used
for coding schemes, which are specifically adapted for
music signals. Such frequency-domain coding schemes have,
as stated before, reduced quality at low bit rates or
speech signals, while specifically adapted speech coders
have a higher quality at comparable bit rates or even have
significantly lower bit rates for the same quality compared
to frequency-domain coding schemes.
Speech coding techniques such as the so-called AMR-WB+
codec as defined in "Extended Adaptive Multi-Rate
Wideband (AMR-WB+) codec", 3GPP TS 26.290 V6.3.0, 2005-06,
Technical Specification, do not apply the MDCT and,
therefore, can not take any advantage from the excellent
properties of the MDCT which, specifically, rely in a
critically sampled processing on the one hand and a
crossover from one block to the other on the other hand.
Therefore, the crossover from one block to the other
obtained by the MDCT without any penalty with respect to
bit rate and, therefore, the critical sampling property of
MDCT has not yet been obtained in speech coders.
When one would combine speech coders and audio coders
within a single hybrid coding scheme, there is still the
problem of how to obtain a switch from one coding mode to
the other coding mode at a low bit rate and a high quality.
It is an object of the present invention to provide an
improved encoding/decoding concept.

This object is achieved by an apparatus for encoding an
audio signal in accordance with claim 1, an apparatus for
decoding an encoded audio signal in accordance with claim
8, an encoded audio signal in accordance with claim 14, a
method for encoding an audio signal in accordance with
claim 15, a method of decoding an encoded audio signal in
accordance with claim 16 or a computer program in
accordance with claim 17.
An aspect of the present invention is that a hybrid coding
scheme is applied, in which a first coding mode
specifically adapted for certain signals and operating in
one domain is applied, and in which a further coding mode
specifically adapted for other signals and operation in a
different domain are used together. In this coding/decoding
concept, a critically sampled switch from one coding mode
to the other coding mode is made possible in that, on the
encoder side, the same block of audio samples which has
been generated by one windowing operation is processed
differently. Specifically, an aliasing portion of the block
of the audio signal is processed by transforming the sub-
block associated with the aliasing portion of the window
from one domain into the other domain subsequent to
windowing this sub-block, where a different sub-block
obtained by the same windowing operation is transformed
from one domain into the other domain before windowing this
sub-block using an analysis window.
The processed first sub-block and the processed second sub-
block are, subsequently, transformed into a further domain
using the same block transform rule to obtain a converted
first block of the audio signal which can then be further
processed using any of the well-known data compression
algorithms such as quantizing, entropy encoding and so on.
On the decoder-side, this block is again processed
differently based on whether the aliasing portion of the

block is processed or the other further portion of the
block is processed. The aliasing portion is transformed
into a target domain before performing a synthesis
windowing while the further portion is subject to a
synthesis windowing before performing the transforming to
the target domain. Additionally, in order to obtain the
critically sampling property, a time domain aliasing
cancellation is performed, in which the windowed aliasing
portion and a windowed aliasing portion of an encoded other
block of the audio data are combined subsequent to a
transform of the aliasing portion of the encoded audio
signal block into the target domain so that a decoded audio
signal corresponding to the aliasing portion of the first
block is obtained. In view of that, there do exist two sub-
blocks/portions in a window. One portion/sub-block
(aliasing sub-block) has aliasing components, which overlap
a second block coded in a different domain, and a second
sub-block/portion (further sub-block), which may or may not
have aliasing components which overlaps the second block or
a block different from the second block.
Preferably, the aliasing introduced into certain portions
which correspond to each other, but which are encoded in
different domains is advantageously used for obtaining a
critically sampled switch from one coding mode to the other
coding mode by differently processing the aliasing portion
and the further portion within one and the same windowed
block of audio sample.
This is in contrast to prior art processing based on
analysis windows and synthesis windows, since, up to now, a
complete data block obtained by applying an analysis window
has been subjected to the same processing. In accordance
with the present invention, however, the aliasing portion
of the windowed block is processed differently compared to
the further portion of this block.

The further portion can comprise a non-aliasing portion
occurring, when specific start/stop windows are used.
Alternatively, the further portion can comprise an aliasing
portion overlapping with a portion of the result of an
adjacent windowing process. Then, the further (aliasing)
portion overlaps with an aliasing portion of a neighboring
frame processed in the same domain compared to the further
(aliasing) portion of the current frame, and the aliasing
portion overlaps with an aliasing portion of a neighboring
frame processed in a different domain compared to the
aliasing portion of the current frame.
Depending on the implementation, the further portion and
the aliasing portion together form the complete result of
an application of a window function to a block of audio
samples. The further portion can be completely aliasing
free or can be completely aliasing or can include an
aliasing sub-portion and an aliasing free sub-portion.
Furthermore, the order of theses sub-portions and the order
of the aliasing portion and the further portion can be
arbitrarily selected.
In a preferred embodiment of the switched audio coding
scheme, adjacent segments of the input signal could be
processed in two different domains. For example, AAC
computes a MDCT in the signal domain, and the MTPC(Sean A.
Ramprashad, "The Multimode Transform predictive Coding
Paradigm", IEEE Transaction on Speech and Audio Processing,
Vol. 11, No. 2, March 2003) computes a MDCT in the LPC
residual domain. It could be problematic especially when
the overlapped regions have time-domain aliasing components
due to the use of a MDCT. Indeed, the time-domain aliasing
can not be cancelled in the transitions where going from
one coder to another, because they were produced in two
different domains. One solution is to make the transitions
with aliasing-free cross-fade windowed signals. The
switched coder is then no more critically sampled and
produces an overhead of information. Embodiments permit to

maintain the critically sampling advantage by canceling
time-domain aliasing components computed by operating in
two different domains.
In a preferred embodiment of the present invention, two
switches are provided in a sequential order, where a first
switch decides between coding in the spectral domain using
a frequency-domain encoder and coding in the LPC-domain,
i.e., processing the signal at the output of an LPC
analysis stage. The second switch is provided for switching
in the LPC-domain in order to encode the LPC-domain signal
either in the LPC-domain such as using an ACELP coder or
coding the LPC-domain signal in an LPC-spectral domain,
which requires a converter for converting the LPC-domain
signal into an LPC-spectral domain, which is different from
a spectral domain, since the LPC-spectral domain shows the
spectrum of an LPC filtered signal rather than the spectrum
of the time-domain signal.
The first switch decides between two processing branches,
where one branch is mainly motivated by a sink model and/or
a psycho acoustic model, i.e. by auditory masking, and the
other one is mainly motivated by a source model and by
segmental SNR calculations. Exemplarily, one branch has a
frequency domain encoder and the other branch has an LPC-
based encoder such as a speech coder. The source model is
usually the speech processing and therefore LPC is commonly
used.
The second switch again decides between two processing
branches, but in a domain different from the "outer" first
branch domain. Again one "inner" branch is mainly motivated
by a source model or by SNR calculations, and the other
"inner" branch can be motivated by a sink model and/or a
psycho acoustic model, i.e. by masking or at least includes
frequency/spectral domain coding aspects. Exemplarily, one
"inner" branch has a frequency domain encoder/spectral
converter and the other branch has an encoder coding on the

other domain such as the LPC domain, wherein this encoder
is for example an CELP or ACELP quantizer/sealer processing
an input signal without a spectral conversion.
A further preferred embodiment is an audio encoder
comprising a first information sink oriented encoding
branch such as a spectral domain encoding branch, a second
information source or SNR oriented encoding branch such as
an LPC-domain encoding branch, and a switch for switching
between the first encoding branch and the second encoding
branch, wherein the second encoding branch comprises a
converter into a specific domain different from the time
domain such as an LPC analysis stage generating an
excitation signal, and wherein the second encoding branch
furthermore comprises a specific domain such as LPC domain
processing branch and a specific spectral domain such as
LPC spectral domain processing branch, and an additional
switch for switching between the specific domain coding
branch and the specific spectral domain coding branch.
A further embodiment of the invention is an audio decoder
comprising a first domain such as a spectral domain
decoding branch, a second domain such as an LPC domain
decoding branch for decoding a signal such as an excitation
signal in the second domain, and a third domain such as an
LPC-spectral decoder branch for decoding a signal such as
an excitation signal in a third domain such as an LPC
spectral domain, wherein the third domain is obtained by
performing a frequency conversion from the second domain
wherein a first switch for the second domain signal and the
third domain signal is provided, and wherein a second
switch for switching between the first domain decoder and
the decoder for the second domain or the third domain is
provided.
Preferred embodiments of the present invention are
subsequently described with respect to the attached
drawings, in which:

Fig. 1A is a schematic representation of a preferred
apparatus or method for encoding an audio signal;
Fig. 1B is a schematic representation of the transition
from MDCT-TCX to AAC;
Fig. 1C is a schematic representation of a transition
from AAC to MDCT-TCX;
Fig. 1D is an illustration of a preferred embodiment of
the inventive concept as a flow chart;
Fig. 2 is a schematic representation for illustrating
four different domains and their relations, which
occur in embodiments of the invention;
Fig. 3A is a scheme illustrating an inventive
apparatus/method for decoding an audio signal;
Fig. 3B is a further illustration of decoding schemes in
accordance with embodiments of the present
invention;
Fig. 4A illustrates details of aliasing-transforms such
as the MDCT applicable in both encoding modes;
Fig. 4B illustrates window functions comparable to the
window function in Fig. 4A, but with an aliasing
portion and a non-aliasing portion;
Fig. 5 is a schematic representation of an encoder and a
decoder in one coding mode such as the AAC-MDCT
coding mode;
Fig. 6 is a representation of an encoder and a decoder
applying MDCT in a different domain such as the

LPC domain in the context of TCX encoding in AMR-
WB+;
Fig. 7 is a specific sequence of windows for transitions
between AAC and AMR-WB+;
Fig. 8A is a representation of a preferred embodiment for
an encoder and a decoder in the context of
switching from the TCX mode to the AAC mode;
Fig. 8B is a preferred embodiment for illustrating an
encoder and a decoder for a transition from AAC
to TCX;
Fig. 9A is a block diagram of a preferred hybrid switched
coding scheme, in which the present invention is
applied;
Fig. 9B is a flow chart illustrating the process
performed in the controller of Fig. 9A;
Fig. 10A is a preferred embodiment of a decoder in a
hybrid switched coding scheme;
Fig. 10B is a flow chart for illustrating the procedure
performed in the transition controller of Fig.
10A;
Fig. 11A illustrates a preferred embodiment of an encoder
in which the present invention is preferably
applied; and
Fig. 11B illustrates a preferred decoder, in which the
present invention is preferably applied.
Fig. 11A illustrates an embodiment of the invention having
two cascaded switches. A mono signal, a stereo signal or a
multi-channel signal is input into a switch 200. The switch

200 is controlled by a decision stage 300. The decision
stage receives, as an input, a signal input into block 200.
Alternatively, the decision stage 300 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 300 actuates the switch 200 in order to
feed a signal either in a frequency encoding portion 400
illustrated at an upper branch of Fig. 11A or an LPC-domain
encoding portion 500 illustrated at a lower branch in Fig.
11A. A key element of the frequency domain encoding branch
is a spectral conversion block 411 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 sub-band signals in this
filterbank may be real valued signals or complex valued
signals. The output of the spectral conversion block 411 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 400 is a processing in
a perception based model or information sink model. Thus,
this branch models the human auditory system receiving
sound. Contrary thereto, the processing in branch 500 is to
generate a signal in the excitation, residual or LPC
domain. Generally, the processing in branch 500 is a
processing in 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 500 may be different.
In the lower encoding branch 500, a key element is an LPC
device 510, which outputs an 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 such as an
excitation signal or a weighted (TCX) signal 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 can be signal-adaptive
so that the decision stage performs a music/speech
discrimination and controls the switch 200 in such a way
that music signals are input into the upper branch 400, and
speech signals are input into the lower branch 500. In one
embodiment, the decision stage 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.
Such a decoder is illustrated in Fig. 11B. The signal
output by 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. Analogously, the output of the
LPC domain encoding branch 500 of Fig. 11A received on the
decoder side and processed by elements 536 and 537 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 600. The switch
600 is controlled via a switch control signal which was,
for example, generated by the decision stage 300, or which
was externally provided such as by a creator of the
original mono signal, stereo signal or multi-channel
signal. The output of the switch 600 is a complete mono
signal, stereo signal or multi-channel signal.
The input signal into the switch 200 and the decision stage
300 can be a mono signal, a stereo signal, a multi-channel
signal or generally an audio signal. Depending on the
decision which can be derived from the switch 200 input
signal or from any external source such as a producer of
the original audio signal underlying the signal input into
stage 200, the switch switches between the frequency
encoding branch 400 and the LPC encoding branch 500. The
frequency encoding branch 400 comprises a spectral
conversion stage 411 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, the switch output signal is
processed via an LPC analysis stage 510 generating LPC side
info and an LPC-domain signal. The excitation encoder
comprises an additional switch 521 for switching the
further processing of the LPC-domain signal between a
quantization/coding operation 526 in the LPC-domain or a
quantization/coding stage 527, which is processing values
in the LPC-spectral domain. To this end, a spectral

converter 527 is provided. 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 for the LPC domain
signal, an inverse quantizer/coder for the LPC spectral
domain signal and an inverse spectral converter for the
output of the inverse quantizer/coder. Both encoded and
again decoded signals in the processing branches of the
second encoding branch are input into a switch control
device. In the switch control device, 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 signal to noise ratio of this branch
is lower than the signal to noise ratio of the other
branch. Alternatively, the target function could use, as an
input, the signal to noise ratio 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 by the
inverse quantizer/coder and the inverse spectral converter.
However, when the main goal is to have the best quality for
a certain bit rate, then the switch control 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 signal to noise ratio, i.e., having the smaller
quantization/coding distortions.

The decoding scheme in accordance with the present
invention is, as stated before, illustrated in Fig. 11B.
For each of the three possible output signal kinds, a
specific decoding/re-quantizing stage 431, 536 or 537
exists. While stage 431 outputs a frequency-spectrum, which
may also be called "time-spectrum" (frequency spectrum of
the time domain signal) , and which is converted into the
time-domain using the frequency/time converter 440, stage
536 outputs an LPC-domain signal, and item 537 receives an
frequency-spectrum of the LPC-domain signal, which may also
be called an "LPC-spectrum". In order to make sure that the
input signals into switch 532 are both in the LPC-domain, a
frequency/time converter 537 is provided in the LPC domain.
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. 11A.
Fig. 11A therefore, illustrates a preferred encoding scheme
in accordance with the invention. A common preprocessing
scheme connected to the switch 200 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 alternatively
to the block 101 or in addition to the block 101 a

bandwidth extension stage 102. In the Fig. 11A embodiment,
the output of block 101 is input into the bandwidth
extension block 102 which, in the encoder of Fig. 11A,
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 102, bandwidth extension parameters 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 300 receives the signal
input into block 101 or input into block 102 in order to
decide between, for example, a music mode or a speech mode.
In the music mode, the upper encoding branch 400 is
selected, while, in the speech mode, the lower encoding
branch 500 is selected. Preferably, the decision stage
additionally controls the joint stereo block 101 and/or the
bandwidth extension block 102 to adapt the functionality of
these blocks to the specific signal. Thus, when the
decision stage determines that a certain time portion of
the input signal is of the first mode such as the music
mode, then specific features of block 101 and/or block 102
can be controlled by the decision stage 300. Alternatively,
when the decision stage 300 determines that the signal is
in a speech mode or, generally, in a second LPC-domain
mode, then specific features of blocks 101 and 102 can be
controlled in accordance with the decision stage output.
Preferably, the spectral conversion of the coding branch
400 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 incurs
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.
In the first coding branch 400, 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. 11A.
In the second coding branch, there is the LPC block 510
followed by a switch 521, again followed by an ACELP block
526 or an 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. The
TCX block 527 receives a weighted signal.
In TCX, the transform is applied to the weighted signal
computed by filtering the input signal through an LPC-based
weighting filter. The weighting filter used in preferred
embodiments of the invention is given by (1-A(z/γ))/(1-µZ-1) .
Thus, the weighted signal is an LPC domain signal and its
transform is an LPC-spectral domain. The signal processed

by ACELP block 526 is the excitation signal and is
different from the signal processed by the block 527, but
both signals are in the LPC domain. The excitation signal
is obtained by filtering the input signal through the
analysis filter (1-A(z/γ)) .
At the decoder side illustrated in Fig. 11B, after the
inverse spectral transform in block 537, the inverse of the
weighting filter is applied, that is (1-µz-1)/(1-A(z/γ)) .
Optionally, the signal can be filtered additionally through
(l-A(z)) to go to the LPC excitation domain. Thus, a signal
from the TCX-1 block 537 can be converted from the weighted
domain to the excitation domain by a filtering through
and then be used in the block 536. This
typical filtering is done in AMR-WB+ at the end of the
inverse TCX (537) for feeding the adaptive codebook of
ACELP in case this last coding is selected for the next
frame.
Although item 510 in Fig. 11A 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.
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. 11a or
lib, the signal is preferably pre-emphasized through a
filter l-0.68z-1 before encoding. At the ACELP/TCX decoder
in Fig. 11B the synthesized signal is deemphasized with the
filter l/(l-0.68z-1) . 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.

In a preferred embodiment, the first switch 200 (see Fig.
11A) is controlled through an open-loop decision and the
second switch is controlled through a closed-loop decision.
Exemplarily, there can be the situation that in the first
processing branch, the first LPC domain represents the LPC
excitation, and in the second processing branch, the second
LPC domain represents the LPC weighted signal. That is, the
first LPC domain signal is obtained by filtering through
(l-A(z)) to convert to the LPC residual domain, while the
second LPC domain signal is obtained by filtering through
the filter (1-A(z/γ))/(1-µz-1) to convert to the LPC weighted
domain. In a preferred mode, µ is equal to 0,68.
Fig. 11B illustrates a decoding scheme corresponding to the
encoding scheme of Fig. 11A. The bitstream generated by
bitstream multiplexer 800 of Fig. 11a is input into a
bitstream demultiplexer 900. Depending on an information
derived for example from the bitstream via a mode detection
block 601, a decoder-side switch 600 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 decision
601, reconstructs the high band based on the low band
output by switch 600.
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 by this block has more channels than the
input into this block.

The switch 200 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
may also be arranged subsequent to for example the
frequency-domain encoder 421 and the LPC domain encoder
510, 521, 526, 527, which means that both branches 400, 500
process the same signal in parallel. In order to not double
the bitrate, however, only the signal output by one of
those encoding branches 400 or 500 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 implementation having two switches, i.e., the first
switch 200 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 operating in the LPC-
domain. Exemplarily, the frequency domain/LPC-domain switch
200 may switch blocks of a length of 1024 samples, and the
second switch 521 can switch blocks having 256 or 512
samples each.
Generally, the audio encoding algorithm used in the first
encoding branch 400 reflects and models the situation in

an audio sink. The sink of an audio information is
normally the human ear. The human ear can be modeled as a
frequency analyzer. Therefore, the first encoding branch
outputs encoded spectral information. Preferably, the
first encoding branch furthermore includes a
psychoacoustic model for additionally applying a
psychoacoustic masking threshold. This psychoacoustic
masking threshold is used when quantizing audio spectral
values where, preferably, the quantization is performed
such that a quantization noise is introduced by quantizing
the spectral audio values, which are hidden below the
psychoacoustic masking threshold.
The second encoding branch represents an information
source model, which reflects the generation of audio
sound. Therefore, information source models may include a
speech model which is reflected by an LPC analysis stage,
i.e., by transforming a time domain signal into an LPC
domain and by subsequently processing the LPC residual
signal, i.e., the excitation signal. Alternative sound
source models, however, are sound source models for
representing a certain instrument or any other sound
generators such as a specific sound source existing in
real world. A selection between different sound source
models can be performed when several sound source models
are available, for example based on an SNR calculation,
i.e., based on a calculation, which of the source models
is the best one suitable for encoding a certain time
portion and/or frequency portion of an audio signal.
Preferably, however, the switch between encoding branches
is performed in the time domain, i.e., that a certain time
portion is encoded using one model and a certain different
time portion of the intermediate signal is encoded using
the other encoding branch.
Information source models are represented by certain
parameters. Regarding the speech model, the parameters are
LPC parameters and coded excitation parameters, when a

modern speech coder such as AMR-WB+ is considered. The AMR-
WB+ comprises an ACELP encoder and a TCX encoder. In this
case, the coded excitation parameters can be global gain,
noise floor, and variable length codes.
The audio input signal in Fig. 11A is present in a first
domain which can, for example, be the time domain but
which can also be any other domain such as a frequency
domain, an LPC domain, an LPC spectral domain or any other
domain. Generally, the conversion from one domain to the
other domain is performed by a conversion algorithm such
as any of the well-known time/frequency conversion
algorithms or frequency/time conversion algorithms.
An alternative transform from the time domain, for example
in the LPC domain is the result of LPC filtering a time
domain signal which results in an LPC residual signal or
excitation signal. Any other filtering operations producing
a filtered signal which has an impact on a substantial
number of signal samples before the transform can be used
as a transform algorithm as the case may be. Therefore,
weighting an audio signal using an LPC based weighting
filter is a further transform, which generates a signal in
the LPC domain. In a time/frequency transform, the
modification of a single spectral value will have an impact
on all time domain values before the transform.
Analogously, a modification of any time domain sample will
have an impact on each frequency domain sample. Similarly,
a modification of a sample of the excitation signal in an
LPC domain situation will have, due to the length of the
LPC filter, an impact on a substantial number of samples
before the LPC filtering. Similarly, a modification of a
sample before an LPC transformation will have an impact on
many samples obtained by this LPC transformation due to the
inherent memory effect of the LPC filter.
Fig. 1A illustrates a preferred embodiment for an apparatus
for encoding an audio signal 10. The audio signal is

preferably introduced into a coding apparatus having a
first encoding branch such as 400 in Fig. 11A for encoding
the audio signal in a third domain which can, for example,
be the straightforward frequency domain. The encoder
furthermore can comprise a second encoding branch for
encoding the audio signal based on a forth domain which can
be, for example, the LPC frequency domain as obtained by
the TCX block 527 in Fig. 11A.
Preferably, the inventive apparatus comprises a windower 11
for windowing the first block of the audio signal in the
first domain using a first analysis window having an
analysis window shape, the analysis window having an
aliasing portion such as Lk or Rk as discussed in the
context of Fig. 8A and Fig. 8B or other figures, and having
a non-aliasing portion such as Mk illustrated in Fig. 5 or
other figures.
The apparatus furthermore comprises a processor 12 for
processing a first sub-block of the audio signal associated
with the aliasing portion of the analysis window by
transforming the sub-block from the first domain such as
the signal domain or straightforward time domain into a
second domain such as the LPC domain subsequent to
windowing the first sub-block to obtain a processed first
sub-block, and for processing a second sub-block of the
audio signal associated with the further portion of the
analysis window by transforming the second sub-block from
the first domain such as the straightforward time domain
into the second domain such as the LPC domain before
windowing the second sub-block to obtain a processed second
sub-block. The inventive apparatus furthermore comprises a
transformer 13 for converting the processed first sub-block
and the processed second sub-block from the second domain
into the fourth domain such as the LPC frequency domain
using the same block transform rule to obtain a converted
first block. This converted first block can, then, be

further processed in a further processing stage 14 to
perform a data compression.
Preferably, the further processing also receives, as an
input, a second block of the audio signal in the first
domain overlapping the first block, wherein the second
block of the audio signal in the first domain such as the
time domain is processed in the third domain, i.e., the
straightforward frequency domain using a second analysis
window. This second analysis window has an aliasing portion
which corresponds to an aliasing portion of the first
analysis window. The aliasing portion of the first analysis
window and the aliasing portion of the second analysis
window preferably relate to the same audio samples of the
original audio signal before windowing, and these portions
are subjected to a time domain aliasing cancellation, i.e.,
an overlap-add procedure on the decoder side.
Fig. 1B illustrates the situation occurring, when
transition from a block encoded in the fourth domain, for
example the LPC frequency domain to a third domain such as
the frequency domain takes place. In an embodiment, the
fourth domain is the MDCT-TCX domain, and the third domain
is the AAC domain. A window applied to the audio signal
encoded in the MDCT-TCX domain has an aliasing portion 20
and a non-aliasing portion 21. The same block, which is
named "first block" in Fig. 1B may or may not have a
further aliasing portion 22. The same is true for the non-
aliasing portion. It may or may not be present.
The second block of the audio signal coded in the other
domain such as the AAC domain comprises a corresponding
aliasing portion 23, and this second block may include
further portions such as a non-aliasing portion or an
aliasing portion as the case may be, which is indicated at
24 in Fig. 1B. Therefore, Fig. 1B illustrates an
overlapping processing of the audio signal so that the
audio samples in the aliasing portion 20 of the first block

before windowing are identical to the audio samples in the
corresponding aliasing portion 23 of the second block
before windowing. Hence, the audio samples in the first
block are obtained by applying an analysis window to the
audio signal which is a stream of audio samples, and the
second block is obtained by applying a second analysis
window to a number of audio samples which include the
samples in the corresponding aliasing portion 23 and the
samples in the further portion 24 of the second block.
Therefore, the audio samples in the aliasing portion 20 are
the first block of the audio signal associated with the
aliasing portion 20, and the audio samples in the further
portion 21 of the audio signal correspond to the second
sub-block of the audio signal associated with the further
portion 21.
Fig. 1C illustrates a similar situation as in Fig. 1B, but
as a transition from AAC, i.e., the third domain into the
MDCT-TCX domain, i.e., the fourth domain.
The difference between Fig. 1B and Fig. 1C is, in general,
that the aliasing portion 20 in Fig. 1B includes audio
samples occurring in time subsequent to audio samples in
the further portion 21, while, in Fig. 1C, the audio
samples in the aliasing portion 20 occur, in time, before
the audio samples in the further portion 21.
Fig. 1D illustrates a detailed representation of the steps
performed with the audio samples in the first sub-block and
the second sub-block of one and same windowed block of
audio samples. Generally, an window has an increasing
portion and a decreasing portion, and depending on the
window shape, there can be a relatively constant middle
portion or not.
In a first step 30, a block forming operation is performed,
in which a certain number of audio samples from a stream of
audio samples is taken. Specifically, the block forming

operation 30 will define, which audio samples belong to the
first block and which audio samples belong to the second
block of Fig. 1B and Fig. 1C.
The audio samples in the aliasing portion 20 are windowed
in a step 31a. Importantly, however, the audio samples in
the non-aliasing portion, i.e., in the second sub-block are
transformed into the second domain, i.e., the LPC domain in
the preferred embodiment in step 32. Then, subsequent to
transforming the audio samples in the second sub-block, the
windowing operation 31b is performed. The audio samples
claimed by the windowing operation 31b form the samples
which are input into a block transform operation to the
fourth domain illustrated in Fig. 1D as item 35.
The windowing operation in block 31a, 31b may or may not
include a folding operation as discussed in connection with
Fig. 8A, 8B, 9A, 10A. Preferably, the windowing operation
31a, 31b additionally comprises a folding operation.
However, the aliasing portion is transformed into the
second domain such as the LPC domain in block 33. Thus, the
block of samples to be transformed into the fourth domain
which is indicated at 34 is completed, and block 34
constitutes one block of data input into one block
transform operation, such as a time/frequency operation.
Since the second domain is, in the preferred embodiment the
LPC domain, the output of the block transform operation as
in step 35 will be in the fourth domain, i.e., the LPC
frequency domain. This block generated by block transform
35 will be the converted first block 36, which is then
first processed in step 37, in order to apply any kind of
data compression which comprises, for example, the data
compression operations applied to TCX data in the AMR-WB+
coder. Naturally, all other data compression operations can
be performed as well in block 37. Therefore, block 37
corresponds to item 14 in Fig. 1A, and block 35 in Fig. 1D
corresponds to item 13 in Fig. 1A, and the windowing

operations correspond to 31b and 31a in Fig. 1D correspond
to item 11 in Fig. 1A, and scheduling of the order between
transforming and windowing which is different for the
further portion and the aliasing portion is performed by
the processor 12 in Fig. 1A.
Fig. 1D illustrates the case, in which the further portion
consists of the non-aliasing sub-portion 21 and an aliasing
sub-portion 22 of Fig. 1B or 1C. Alternatively, the further
portion can only include an aliasing portion without a non-
aliasing portion. In this case, 21 in Fig. 1B and 1C would
not be there and 22 would extend from the border of the
block to the border of the aliasing portion 20. In any
case, the further portion/further sub-block is processed in
the same way (irrespective of being fully aliasing-free or
fully aliasing or having an aliasing sub-portion and a non-
aliasing sub-portion), but differently from the aliasing
sub-block.
Fig. 2 illustrates an overview over different domains which
occur in preferred embodiments of the present invention.
Normally, the audio signal will be in the first domain 40
which can, for example, be the time domain. However, the
invention actually applies to all situations, which occur
when an audio signal is to be encoded in two different
domains, and when the switch from one domain to the other
domain has to be performed in a bit-rate optimum way, i.e.,
using critically sampling.
The second domain will be, in a preferred embodiment, an
LPC domain 41. A transform from the first domain to the
second domain will be done via an LPC filter/transform as
indicated in Fig. 2.
The third domain is, in a preferred embodiment, the
straightforward frequency domain 42, which is obtained by
any of the well-known time/frequency transforms such as a

DCT (discrete cosine transform), a DST (discrete sine
transform), a Fourier transform or a fast Fourier transform
or any other time/frequency transform.
Correspondingly, a conversion from the second domain into a
fourth domain 43, such as an LPC frequency domain or,
generally stated, the frequency domain with respect to the
second domain 41 can also be obtained by any of the well-
known time/frequency transform algorithms, such as DCT,
DST, FT, FFT.
Then Fig. 2 is compared to Fig. 11A or 11B, the output of
block 421 will have a signal in the third domain.
Furthermore, the output of block 526 will have a signal in
the second domain, and the output of block 527 will
comprise a signal in the fourth domain. The other signal
input into switch 200 or, generally, input into the
decision stage 300 or the surround/joint stereo stage 101
will be in the first domain such as the time domain.
Fig. 3A illustrates a preferred embodiment of an inventive
apparatus for decoding an encoded audio signal having an
encoded first block 50 of audio data, where the encoded
block has an aliasing portion and a further portion. The
inventive decoder furthermore comprises a processor 51 for
processing the aliasing portion by transforming the
aliasing portion into a target domain for performing a
synthesis windowing to obtain a windowed aliasing portion
52, and for performing a synthesis windowing of the further
portion before performing a transform of the windowed
further portion into the target domain.
Therefore, on the decoder side, portions of a block
belonging to the same window are processed differently. A
similar processing has been applied on the encoder side to
allow a critically sampled switch over between different
domains.

The inventive decoder furthermore comprises a time domain
aliasing canceller 53 for combining the windowed aliasing
portion of the first block, i.e., input 52, and a windowed
aliasing portion of an encoded second block of audio data
subsequent to a transform of the aliasing portion of the
encoded second block into the target domain, in order to
obtain a decoded audio signal 55, which corresponds to the
aliasing portion of the first block. The windowed aliasing
portion of the encoded second block is input via 54 into
the time domain aliasing canceller 53.
Preferably, a time domain aliasing canceller 53 is
implemented as an overlap/add device, which, for example
applies a 50% overlap. This means that the result of a
synthesis window of one block is overlapped with the result
of a synthesis window processing of an adjacent encoded
block of audio data, where this overlap preferably
comprises 50% of the block. This means that the second
portion of synthesis windowed audio data of an earlier
block is added in a sample-wise manner to the first portion
of a later second block of encoded audio data, so that, in
the end, the decoded audio samples are the sum of
corresponding windowed samples of two adjacent blocks. In
other embodiments, the overlapping range can be more or
less than 50%. This combining feature of the time domain
aliasing canceller provides a continuous cross-fade from
one block to the next, which completely removes any
blocking artifacts occurring in any block-based transform
coding scheme. Due to the fact that aliasing portions of
different domains can be combined by the present invention,
a critically sampled switching operation from a block of
one domain to a block of the other domain is obtained.
Compared to a switch encoder without any cross-fading, in
which a hard switch from one block to the other block is
performed, the audio quality is improved by the inventive
procedure, since the hard switch would inevitably result in

blocking artifacts such as audible cracks or any other
unwanted noise at the block border.
Compared to the non-critically sampled cross-fade, which
indeed, would remove such an unwanted sharp noise at the
block border, however, the present invention does not
result in any data rate increase due to the switch. When,
in the prior art, the same audio samples would be encoded
in the first block via the first coding branch and would be
encoded in the second block via the second coding branch, a
sample amount has been encoded in both coding branches
would consume bit rate, when it would be processed without
an aliasing introduction. In accordance with the present
invention, however, an aliasing is introduced at the block
borders. This aliasing-introduction which is obtained by a
sample reduction, however, results in a possibility to
apply a cross-fading operation by the time domain aliasing
canceller 53 without the penalty of an increased bit rate
or a non-critically sampled switch-over.
In the most preferred embodiment, a truly critically
sampled switchover is performed. However, there can also
be, in certain situations, less efficient embodiments, in
which only a certain amount of aliasing is introduced and a
certain amount of bit rate overhead is allowed. Due to the
fact that aliasing portions are used and combined, however,
all these less efficient embodiments are, nevertheless,
always better than a completely aliasing free transition
with cross-fade or are with respect to quality, better than
a hard switch from one encoding branch to the other
encoding branch.
In this context, it is to be noted that the non-aliasing
portion in TCX still produces critically sampled coded
samples. Adding a non-aliasing portion in TCX does not
compromise the critical sampling, but compromises the
quality of the transition (lower handover) and the quality
of the spectral representation (lower energy compaction).

In view of this, it is preferred to have the non-aliasing
portion in TCX as small as possible or even close to zero
so that the further portion is fully aliasing and does not
have an aliasing-free sub-portion.
Subsequently, Fig. 3B will be discussed in order to
illustrate a preferred embodiment of the procedure in Fig.
3A.
In a step 56, the decoder processing of the encoded first
block which is, for example, in the fourth domain, is
performed. This decoder processing may be an entropy-
decoding such as Huffman decoding or an arithmetic decoding
corresponding to the further processing operations in block
14 of Fig. 1A on the encoder side. In step 57, a
frequency/time conversion of the complete first block is
performed as indicated at step 57. In accordance with Fig.
2, this procedure in step 57 results in a complete first
block in the second domain. Now, in accordance with the
present invention, the portions of the first block are
processed differently. Specifically, the aliasing portion,
i.e., the first sub-block of the output of step 57 will be
transformed to the target domain before a windowing
operation using a synthesis window is performed. This is
indicated by the order of the transforming step 58a and the
windowing step 59a. The second sub-block, i.e., the
aliasing-free sub-block is windowed using a synthesis
window as indicated at 59b, as it is, i.e., without the
transforming operation in item 58a in Fig. 3B. The
windowing operation in block 59a or 59b may or may not
comprise a folding (unfolding) operation. Preferably,
however, the windowing operation comprises a folding
(unfolding operation).
Depending on whether the second sub-block corresponding to
the further portion is indeed an aliasing sub-block or a
non-aliasing sub-block, the transforming operation into the
target domain as indicated at 59b is performed without any

TDAC operation/combining operation in the case of the
second sub-block being a non-aliasing sub-block. When,
however, the second sub-block is an aliasing sub-block, a
TDAC operation, i.e., a combining operation 60b is
performed with a corresponding portion of another block,
before the transforming operation into the target domain in
step 59b is obtained to calculate the decoded audio signal
for the second block.
In the other branch, i.e., for the aliasing portion
corresponding to the first sub-block, the result of the
windowing operation in step 59a is input into a combining
stage 60a. This combining stage 60a also receives, as an
input, the aliasing portion of the second block, i.e., the
block which has been encoded in the other domain, such as
the AAC domain in the example of Fig. 2. Then, the output
of block 60a constitutes the decoded audio signal for the
first sub-block.
When, Fig. 3A and Fig. 3B are compared, it becomes clear
that the combining operation 60a corresponds to the
processing performed in the block 53 of Fig. 3A.
Furthermore, the transforming operation and the windowing
operation performed by the processor 51 corresponds to
items 58a, 58b with respect to the transforming operation
and 59a and 59b with respect to the windowing operation,
where the processor 51 in Fig. 3A furthermore insures that
the correct order for the aliasing portion and the other
portion, i.e., the second sub-block, is maintained.
In the preferred embodiment, the modified discrete cosine
transform (MDCT) is applied in order to obtain the
critically sampling switchover from an encoding operation
in one domain to an encoding operation in a different other
domain. However, all other transforms can be applied as
well. Since, however, the MDCT is the preferred embodiment,
the MDCT will be discussed in more detail with respect to
Fig. 4A and Fig. 4B.

Fig.4A illustrates a window 70, which has an increasing
portion to the left and a decreasing portion to the right,
where one can divide this window into four portions: a, b,
c, and d. Window 70 has, as can be seen from the figure
only aliasing portions in the 50% overlap/add situation
illustrated. Specifically, the first portion having samples
from zero to N corresponds to the second portions of a
preceding window 69, and the second half extending between
sample N and sample 2N of window 70 is overlapped with the
first portion of window 71, which is in the illustrated
embodiment window i+1, while window 70 is window i.
The MDCT operation can be seen as the cascading of the
folding operation and a subsequent transform operation and,
specifically, a subsequent DCT operation, where the DCT of
type-IV (DCT-IV) is applied. Specifically, the folding
operation is obtained by calculating the first portion N/2
of the folding block as -cR-d, and calculating the second
portion of N/2 samples of the folding output as a-bR, where
R is the reverse operator. Thus, the folding operation
results in N output values while 2N input values are
received.
A corresponding unfolding operation on the decoder-side is
illustrated, in equation form, in Fig. 4A as well.
Generally, an MDCT operation on (a,b,c,d) results in
exactly the same output values as the DCT-IV of (-cR-d, a-
bR) as indicated in Fig. 4A.
Correspondingly, and using the unfolding operation, an
IMDCT operation results in the output of the unfolding
operation applied to the output of a DCT-IV inverse
transform.
Therefore, time aliasing is introduced by performing a
folding operation on the decoder-side. Then, the result of

the folding operation is transformed into the frequency
domain using a DCT-IV block transform requiring N input
values.
On the decoder-side, N input values are transformed back
into the time domain using a DCT-IV-1 operation, and the
output of this inverse transform operation is thus changed
into an unfolding operation to obtain 2N output values
which, however, are aliased output values.
In order to remove the aliasing which has been introduced
by the folding operation and which is still there
subsequent to the unfolding operation, the overlap/add
operation by the time domain aliasing canceller 53 of Fig.
3A is required.
Therefore, when the result of the unfolding operation is
added with the previous IMDCT result in the overlapping
half, the reversed terms cancel in the equation in the
bottom of Fig. 4A and one obtains simply, for example, b
and d, thus recovering the original data.
In order to obtain a TDAC for the windowed MDCT, a
requirement exists, which is known as "Princen-Bradley"
condition, which means that the window coefficients raised
to 2 for the corresponding samples which are combined in
the time domain aliasing canceller as to result in unity
(1) for each sample.
While Fig. 4A illustrates the window sequence as, for
example, applied in the AAC-MDCT for long windows or short
windows, Fig. 4D illustrates a different window function
which has, in addition to aliasing portions, a non-aliasing
portion as well.
Fig. 4D illustrates an analysis window function 72 having a
zero portion a1 and d2, having an aliasing portion 72a,
72b, and having a non-aliasing portion 72c.

The aliasing portion 72b extending over C2, d1 has a
corresponding aliasing portion of a subsequent window 73,
which is indicated at 73b. Correspondingly, window 73
additionally comprises a non-aliasing portion 73a. Fig. 4B,
when compared to Fig. 4A makes clear that, due to the fact
that there are zero portions a1, da, for window 72 or C1 for
window 73, both windows receive a non-aliasing portion, and
the window function in the aliasing portion is steeper than
in Fig. 4A. In view of that, the aliasing portion 72a
corresponds to Lk, the non-aliasing portion 72c corresponds
to portion Mk, and the aliasing portion 72b corresponds to
Rk in Fig. 4B.
When the folding operation is applied to a block of samples
windowed by window 72, a situation is obtained as
illustrated in Fig. 4B. The left portion extending over the
first N/4 samples has aliasing. The second portion
extending over N/2 samples is aliasing-free, since the
folding operation is applied on window portions having zero
values, and the last N/4 samples are, again, aliasing-
affected. Due to the folding operation, the number of
output values of the folding operation is equal to N, while
the input was 2N, although, in fact, N/2 values in this
embodiment were set to zero due to the windowing operation
using window 72.
Now, the DCT IV is applied to the result of the folding
operation, but, importantly, the aliasing portion 72 which
is at the transition from one coding mode to the other
coding mode is differently processed than the non-aliasing
portion, although both portions belong to the same block of
audio samples and, importantly, are input into the same
block transform operation performed by the transformer 30
in Fig. 1A.
Fig. 4B furthermore illustrates a window sequence of
windows 72, 73, 74, where the window 73 is a transition

window from a situation where there does exist non-aliasing
portions to a situation, where only exist aliasing
portions. This is obtained by asymmetrically shaping the
window function. The right portion of window 73 is similar
to the right portion of the windows in the window sequence
of Fig. 4A, while the left portion has a non-aliasing
portion and the corresponding zero portion (at C1).
Therefore, Fig. 4B illustrates a transition from MDCT-TCX
to AAC, when AAC is to be performed using fully-overlapping
windows or, alternatively, a transition from AAC to MDCT-
TCX is illustrated, when window 74 windows a TCX data block
in a fully-overlapping manner, which is the regular
operation for MDCT-TCX on the one hand and MDCT-AAC on the
other hand when there is no reason for switching from one
mode to the other mode.
Therefore, window 73 can be termed to be a "start window"
or a "stop window", which has, in addition, the preferred
characteristic that the length of this window is identical
to the length of at least one neighboring window so that
the general block raster or frame raster is maintained,
when a block is set to have the same number as window
coefficients, i.e., 2n samples in the Fig. 4D or Fig. 4A
example.
Subsequently, the AAC-MDCT procedure on the encoder-side
and on the decoder-side is discussed with respect to Fig.
5.
In a windowing operation 80, a window function is
illustrated at 81 is applied. The window function has two
aliasing portions Lk and Rk, and a non-aliasing portion Mk.
Therefore, the window function 81 is similar to the window
function 72 in Fig. 4B. Applying this window function to a
corresponding plurality of audio samples results in the
windowed block of audio samples having an aliasing sub-
block corresponding to Rk/Lk and a non-aliasing sub-block
corresponding to Mk.

The folding operation illustrated by 82 is performed as
indicated in Fig. 4B and results in N outputs, which means
that the portions Lkr Rk are reduced to have a smaller
number of samples.
Then, a DCT IV 83 is performed as discussed in connection
with the MDCT equation in Fig. 4A. The MDCT output is
further processed by any available data compressor such as
a quantizer 84 or any other device performing any of the
well-known AAC tools.
On the decoder side, an inverse processing 85 is performed.
Then, a transform from the third domain into the first
domain is performed via the DCT-1 IV 86. Then, an unfolding
operation 87 is performed as discussed in connection with
Fig. 4A. Then, in a block 88, a synthesis windowing
operation is performed, and items 89a and 89b together
perform a time domain aliasing cancellation. Item 89b is a
delay device applying a delay of Mk+Rk samples in order to
obtain the overlap as discussed in connection with Fig. 4A,
and adder 89a performs a combination of the current portion
of the audio samples such as the first portion Lk of a
current window output and the last portion Rk-x of the
previous window. This results, as indicated at 90, in
aliasing-free portions Lk and Mk. It is to be noted that Mk
was aliasing-free from the beginning, but the processing by
the devices 89a, 89b has cancelled the aliasing in the
aliasing portion Lk.
In the preferred embodiment, the AAC-MDCT can also be
applied with windows only having aliasing portions as
indicated in Fig. 4A, but, for a switch between one coding
mode to the other coding mode, it is preferred that an AAC
window having an aliasing portion and having a non-aliasing
portion is applied.

An embodiment of the present invention is used in a
switched audio coding which switches between AAC and AMR-
WB+[4].
AAC uses a MDCT as described in Fig. 5. AAC is very well
suited for music signal. The switched coding uses AAC when
the input signal is detected in a previous processing as
music or labeled as music by the user.
The input signal frame k is windowed by a three parts
window of sizes Lk, Mk and Rk. The MDCT introduces time-
domain aliasing components before transforming the signal
in frequency domain where the quantization is performed.
After adding the overlapped previous windowed signal of
size Rk-i = Lk, the Lk+Mk first samples of original signal
frame could be recovered if any quantization error was
introduced. The time-domain aliasing is cancelled.
Subsequently, the TCX-MDCT procedure with respect to the
present invention is discussed in connection with Fig. 6.
In contrast to the encoder in Fig. 5, a transform into the
second domain is performed by item 92. Item 92 is an LPC
transformer either generating an LPC residual signal or a
weighted signal which can be calculated by weighting an LPC
residual signal using a weighting filter as known from TCX
processing. Naturally, the TCX signal can also be
calculated with a single filter by filtering the time
domain signal in order to obtain the TCX signal, which is a
signal in the LPC domain or, generally state, in the second
domain. Therefore, the first domain/second domain converter
92 provides, at its output site, the signal input into the
windowing device 80. Apart from the transformer 92, the
procedure in the encoder in Fig. 6 is similar to the
procedure in the encoder of Fig. 5. Naturally, one can
apply different data compression algorithms in blocks 84 in
Fig. 5 and Fig. 6, which are readily apparent, when the AAC
coding tools are compared to the TCX coding tools.

On the decoder side, the same steps as discussed in
connection with Fig. 5 are performed, but these steps are
not performed on an encoded signal in the straightforward
frequency domain (third domain) , but are performed on a
coded signal which is generated in the fourth domain, i.e.,
the LPC frequency domain.
Therefore, the overlap add procedure by devices 8 9a, 8 9b in
Fig. 6 is performed in the second domain rather than in the
first domain as illustrated in Fig. 5.
AMR-WB+ is based on a speech coding ACELP and a transform-
based coding TCX. For each super-frame of 1024 samples,
AMR-WB+ select with closed-loop decision between 17
different combination of TCX and ACELP, the best one
according to closed-decision using the SegSNR objective
evaluation. The AMR-WB+ is well-suited for speech and
speech over music signals. The original DFT of the TCX was
replaced by a MDCT in order to enjoy its great properties.
The TCX of AMR-WB+ is then equivalent to the MPTC coding
excepting for the quantization which was kept as it is. The
modified AMR-WB+ is used by the switched audio coder when
the input signal is detected or labeled as speech or speech
over music.
The TCX-MDCT performs a MDCT not directly on the signal
domain but after filtering the signal by a analysis filter
W(z) based on an LPC coefficient. The filter is called
weighting analysis filter and permits the TCX in the same
time to whiten the signal and to shape the quantization
noise by a formant-based curve which is in line with
psycho-acoustic theories.
The processing illustrated in Fig. 5 is performed for a
straightforward AAC-MDCT mode without any switching to TCX
mode or any other mode using the fully overlapping windows
in Fig.4A. When, however, a transition is detected, a

specific window is applied, which is an AAC start window
for a transition to the other coding mode or an AAC stop
window for the transition from the other coding mode into
the AAC mode as illustrated in Fig. 7. An AAC stop window
93 has an aliasing portion illustrated at 93b and a non-
aliasing portion illustrated at 93a, i.e., indicated in the
figure as the horizontal part of the window 93.
Correspondingly, the AAC stop window 94 is illustrated as
having an aliasing portion 94b and a non-aliasing portion
94a. In the AMR-WB+ portion, a window is applied similar to
window 72 of Fig. 4B, where this window has an aliasing
portion 72a and a non-aliasing portion 72c. Although only a
single AMR-WB+ window which can be seen as a start/stop
window as illustrated in Fig. 7, there can be a plurality
of windows which, preferably, have a 50% overlapping and
can, therefore, be similar to the windows in Fig. 4A. Usu-
ally TCX in AMR-WB+ does not use any 50% overlap. Only a
small overlap is adopted for being able to switch promptly
to/from ACELP which uses inherently rectangular window,
i.e. 0% of overlap.
However, when the transition takes place, an AMR-WB+ start
window is applied illustrated at the left center position
in Fig. 7, and when it is decided that the transition from
AMR-WB+ to AAC is to be performed, an AMR-WB+ stop window
is applied. The start window has an aliasing portion to the
left and the stop window has an aliasing portion to the
right, where these aliasing portions are indicated as 72a,
and where these aliasing portions correspond to the
aliasing portions of the neighboring AAC start/stop windows
indicated at 93b or 94b.
The specific processing occurs in the two overlapped
regions of 128 samples of Fig. 7. For canceling the time-
domain aliasing of AAC, the first and the last frames of
the AMR-WB+ segment are forced to be TCX and not ACELP.
this is done by biasing the SegSNR score in the closed-loop
decision. Furthermore the first 128 samples of the TCX-MDCT

are processed specifically as illustrated in Figure 8A,
where Lk=128.
The last 128 samples of AMR-WB+ are processed as
illustrated in the Figure 8B, where Rk=128.
Fig. 8A illustrates the processing for the aliasing portion
Rk to the right of the non-aliasing portion for a
transition from TCX to AAC, and Fig. 8B illustrates the
specific processing of the aliasing portion Lk to the left
of a non-aliasing portion for a transition from AAC to TCX.
The processing is similar with respect to Fig. 6, but the
weighting operation, i.e., the transform from the first
domain to the second domain is positioned differently.
Specifically, in Fig. 6, the transform is performed before
windowing, while, in Fig. 8B, the transform 92 is performed
subsequent to the windowing 80 (and the folding 82), i.e.,
the time domain aliasing introducing operation indicated by
"TDA".
On the decoder side, again, quite similar processing steps
as in Fig. 6 are performed, but, again, the position of the
inverse weighting for the aliasing portion is before
windowing 88 (and before unfolding 87) and subsequent to
the transform from the first domain to the second domain
indicated by 86 in Fig. 8A.
Therefore, in accordance with a preferred embodiment of the
present invention, the aliasing portion of a transition
window for TCX is processed as indicated in Fig. 1A or Fig. 1B, and a non-aliasing portion for the same window is
processed in accordance with Fig. 6.
The processing for any AAC-MDCT window remains the same
apart from the fact that a start window or a stop window is
selected at the transition. In other embodiments, however,
the TCX processing can remain the same and the aliasing

portion of the AAC-MDCT window is processed differently-
compared to the non-aliasing portion.
Furthermore, both aliasing portions of both windows, i.e.,
an AAC window or a TCX window can be processed differently
from their non-aliasing portions as the case may be. In the
preferred embodiment, however, it is preferred that the AAC
processing is done as it is, since it is already in the
signal domain subsequent to the overlap-add procedure as is
clear from Fig. 5, and that the TCX transition window is
processed as illustrated in the context of Fig. 6 for a
non-aliasing portion and as illustrated in Fig. 8A or 8B
for the aliasing portion.
Subsequently, Fig. 9A will be discussed, in which the
processor 12 of Fig. 1A has been indicated as a controller
98.
Devices in Fig. 9A having corresponding reference numerals
which correspond to items of Fig. 11A have a similar
functionality and are not discussed again.
Specifically, the controller 98 illustrated in Fig. 9A
operates as indicated in Fig. 9B. In step 98a, a transition
is detected, where this transition is indicated by the
decision stage 300. Then, the controller 98 is active to
bias the switch 521 so that the switch 521 selects
alternative (2b) in any case.
Then, step 98b is performed by the controller 98.
Specifically, the controller is operative to take the data
in the aliasing portion and to not feed the data into the
LPC 510 directly, but to feed the data before LPC filter
510 directly, without weighting by an LPC filter, into the
TDA block 527a. Then, this data is taken by the controller
98 and weighted and, then, fed into DCT block 527b, i.e.,
after having been weighted by the weighting filter at the
controller 98 output. The weighting filter at the

controller 98 uses the LPC coefficients calculated in the
LPC block 510 after a signal analysis. The LPC block is
able to feed either ACELP or TCX and moreover perform a LPC
analysis for obtaining the LPC coefficients.The DCT portion
527b of the MDCT device consists of the TDA device 527a and
the DCT device 527b. The weighting filter at the output of
the controller 98 has the same characteristic as the filter
in the LPC block 510 and a potentially present additional
weighting filter such as the perceptual filter in AMR-WB+
TCX processing. Hence, in step 98b, TDA-, LPC-, and DCT
processing are performed in this order.
The data in the further portion is fed into the LPC block
510 and, subsequently, in the MDCT block 527a, 527b as
indicated by the normal signal path in Fig. 9A. In this
case, the TCX weighting filter is not explicitly
illustrated in Fig. 9A because it belongs to the LPC block
510.
As stated before, the data in the aliasing portion is, as
indicated in Fig. 8A windowed in block 527a, and the
windowed data generated within block 527 is LPC filtered at
the controller output and the result of the LPC filtering
is then applied to the transform portion 527b of the MDCT
block 527. The TCX weighting filter for weighting the LPC
residual signal generated by LPC device 510 is not
illustrated in Fig. 9A. Additionally, device 527a includes
the windowing stage 80 and, the folding stage 82 and device
527b includes the DCT IV stage 83 as discussed in
connection with Fig. 8A. The DCT IV stage 83/527b then
receives the aliasing portion after processing and the
further portion after the corresponding processing and
performs the common MDCT operation, and a subsequent data
compression in block 528 is performed as indicated by step
98d in Fig. 9B. Therefore, in case of an encoder hardwired
or software-controlled as discussed in connection with Fig.
9A, the controller 98 performs the data scheduling as

indicated in Fig. 9D between the different blocks 510 and
527a, 527b.
On the decoder side, a transition controller 99 is provided
in addition to the blocks indicated in Fig. 11B, which have
already been discussed.
The functionality of the transition controller 99 is
discussed in connection with Fig. 10B.
As soon as the transition controller 99 has detected a
transition as outlined in step 99a in Fig. 10B, the whole
frame is fed into the MDCT-1 stage 537b subsequent to a
data decompression in data decompressor 537a. This
procedure is indicated in step 99b of Fig. 10B. Then, as
indicated in step 99c, the aliasing portion is fed directly
into the LPC-1 stage before performing a TDAC processing.
However, the aliasing portion is not subjected to a
complete "MDCT" processing, but only, as illustrated in
Fig. 8B, subjected to the inverse transform from the fourth
domain to the second domain.
Feeding the aliasing portion subsequent to the DCT"1 IV
stage 86/stage 537b of Fig. 8B into the additional LPC-1
stage 537d in Fig. 10A makes sure that a transform from the
second domain to the first domain is performed, and,
subsequently, the unfolding operation 87 and the windowing
operation 88 of Fig. 8B are performed in block 537c.
Therefore, the transition controller 99 receives data from
block 537b subsequent to the DCT-1 operation of stage 8 6,
and then feeds this data to the LPC-1 block 537d. The
output of this procedure is then fed into block 537d to
perform unfolding 87 and windowing 88. Then, the result of
windowing the aliasing portion is forwarded to TDAC block
440b in order to perform an overlap-add operation with the
corresponding aliasing portion of an AAC-MDCT block. In
view of that, the order of processing for the aliasing
block is: data decompression in 537a, DCT-1 in 537b,

inverse LPC and inverse TCX perceptual weighting (together
meaning inverse weighting) in 537d, TDA -1 processing in
537c and, then, overlap and add in 440b.
Nevertheless, the remaining portion of the frame is fed
into the windowing stage before TDAC and inverse
filtering/weighting in 540 as discussed in connection with
Fig. 6 and as illustrated by the normal signal flow
illustrated in Fig. 10A, when the arrows connected to block
99 are ignored.
In view of that, step 99c results the decoded audio signal
for the aliasing portion subsequent to the TDAC 440b, and
step 99d results in the decoded audio signal for the
remaining/further portion subsequent to the TDAC 537c in
the LPC domain and the inverse weighting in block 540.
Depending on certain implementation requirements, embodi-
ments of the invention can be implemented in hardware or in
software. The implementation can be performed using a digi-
tal storage medium, for example a floppy disk, a DVD, a CD,
a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory, hav-
ing electronically readable control signals stored thereon,
which cooperate (or are capable of cooperating) with a pro-
grammable 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 com-
puter system, such that one of the methods described herein
is performed.
Generally, embodiments of the present invention can be im-
plemented 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 per-
forming 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, there-
fore, 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 de-
scribed 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 de-
scribed 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 ex-
ample a computer, or a programmable logic device, confi-
gured to or adapted to perform one of the methods described
herein. A1
A further embodiment comprises a computer having installed
thereon the computer program for performing one of the me-
thods described herein.
In some embodiments, a programmable logic device (for exam-
ple 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 ar-

ray may cooperate with a microprocessor in order to perform
one of the methods described herein.
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 lim-
ited 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. Apparatus for encoding an audio signal, comprising:
a windower (11) for windowing a first block of the
audio signal using an analysis window, the analysis
window having an aliasing portion (Lk,Rk), and a
further portion (Mk) ;
a processor (12) for processing a first sub-block (20)
of the audio signal associated with the aliasing
portion by transforming the first sub-block into a
domain different from the domain, in which the audio
signal is, subsequent to windowing the first sub-block
to obtain a processed first sub-block, and for
processing a second sub-block (21) of the audio signal
associated with the further portion by transforming
the second sub-block into the different domain before
windowing the second sub-block to obtain a processed
second sub-block; and
a transformer (13) for converting the processed first
sub-block and the processed second sub-block from the
different domain into a further domain using the same
block transform rule to obtain a converted first
block,
wherein the apparatus is configured for further
processing (14) the converted first block using a data
compression algorithm.
2. Apparatus in accordance with claim 1, which is
configured for processing a second block of the audio
signal overlapping with the first block using a second
analysis window (73) having an aliasing portion (73b)
corresponding to the aliasing portion (72b) of the
first analysis window.

3. Apparatus in accordance with claim 1 or claim 2, in
which the domain, in which the audio signal is
positioned, is a time domain, in which the different
domain is an LPC domain, in which a third domain, in
which a second block of the audio signal overlapping
with the first block of the audio signal is encoded,
is a frequency domain, and in which the further
domain, in which the transformer (13) is configured
for transforming, is an LPC frequency domain, and
wherein the processor (12) comprises an LPC filter for
transforming from the first domain to the second
domain, or wherein the transformer (13) comprises a
Fourier-based conversion algorithm for transforming
input data into a frequency domain of the input data
such as a DCT, a DST, an FFT, or a DFT.
4. Apparatus in accordance with one of the preceding
claims, in which the windower (11) comprises a folding
function (82) for folding input values to obtain
output values, the number of output values being
smaller than the number of input values, wherein the
folding function is such that time aliasing is
introduced into the output values.
5. Apparatus in accordance with any one of the preceding
claims, in which the windower (11) is operative to
perform the windowing to obtain the input values for a
subsequently performed folding function (82).
6. Apparatus in accordance with one of the preceding
claims, in which the apparatus comprises a first
encoding branch (400) for encoding the audio signal in
a frequency domain, and a second encoding branch (500)
for encoding the audio signal based on a different
frequency domain,

wherein the second encoding branch has a first sub-
branch (527,528) for encoding the audio signal in the
other frequency domain, and a second sub-branch (526)
for encoding the audio signal in the other domain, the
apparatus further comprising a decision stage (300)
for deciding, whether a block of audio data is
represented in an output bit stream by data generated
using the first encoding branch or the first sub-
branch or the second sub-branch of the second encoding
branch, and
wherein the controller (98) is configured for
controlling the decision stage (300) to decide in
favor of the first sub-branch, when the transition
from the first encoding branch to the second encoding
branch or from the second encoding branch to the first
encoding branch is to be performed.
7. Apparatus in accordance with any one of the preceding
claims, in which the further portion comprises a non-
aliasing portion (Mk) and an additional aliasing
portion or an aliasing portion overlapping with a
corresponding aliasing portion of a neighboring block
of the audio signal.
8. Apparatus for decoding an encoded audio signal having
an encoded first block of audio data, the encoded
block having an aliasing portion and a further
portion, comprising:
a processor (51) for processing the aliasing portion
(Lk,Rk) by transforming (86) the aliasing portion into
a target domain before performing a synthesis
windowing (88) to obtain a windowed aliasing portion,
and for performing a synthesis windowing (88) of the
further portion before performing a transform (98)
into the target domain; and

a time domain aliasing canceller (53) for combining
the windowed aliasing portion and the windowed
aliasing portion of an encoded second block of audio
data subsequent to a transform (91) of the aliasing
portion of the encoded first block of audio data into
the target domain to obtain a decoded audio signal
corresponding to the aliasing portion of the first
block.
9. Apparatus in accordance with claim 8,
in which the processor (51) comprises a transformer
(86) for converting the aliasing portion from a fourth
domain into a second domain, and wherein the processor
furthermore comprises a transformer (91) for
converting the aliasing portion represented in the
second domain into the first domain, wherein the
transformer (8 6) is operative to perform a block-based
frequency time conversion algorithm.
10. Apparatus in accordance with claim 8 or 9, in which
the processor (12) is operative to perform an
unfolding operation (87) for obtaining output data
having a number of values larger than a number of
values input into the unfolding operation (87).
11. Apparatus in accordance with any one of claims 8,9, or
10, in which the processor (12) is operative to use a
synthesis windowing function (88) being related to an
analysis window function used when generating the
encoded audio signal.
12. Apparatus in accordance with any one of claims 8-11,
in which the encoded audio signal comprises a coding
mode indicator indicating a coding mode for the
encoded first block and the encoded second block,

wherein the apparatus further comprises a transition
controller (99) for controlling the processor (12),
when the coding mode indicator indicates a coding mode
change from a first coding mode to a different second
coding mode or vice versa, and for controlling the
processor (12) to perform the same operating for a
complete encoding block, when a coding mode change
between two encoding blocks is not signaled.
13. Apparatus in accordance with any one of claims 8-12,
in which a first coding mode and a second coding mode
comprise an entropy decoding stage, a dequantizing
stage, a frequency-time converting stage comprising an
unfolding operation, and a synthesis windowing stage,
in which the time domain aliasing canceller (53)
comprises an adder (89a) for adding corresponding
aliasing portions of encoded blocks obtained by the
synthesis windowing stage (88), the corresponding
aliasing portions being obtained by an overlapping
processing (89b) of the audio signal, and
in which, in the first coding mode, the time domain
aliasing canceller (53) is configured for adding
portions of blocks obtained by the synthesis windowing
to obtain, as an output of the addition (89a), the
decoded signal in the target domain, and
in which, in the second coding mode, the output of the
addition (89a) is processed by the processor (12) to
perform a transform (91) of the output of the addition
to the target domain.
14. Encoded audio signal comprising an encoded first block
of an audio signal and an overlapping encoded second
block of the audio signal, the encoded first block of
the audio signal comprising an aliasing portion and a

further portion, the aliasing portion having been
transformed from a first domain to a second domain
subsequent to windowing (80) the aliasing portion, and
the further portion having been transformed from the
first domain into the second domain before windowing
(80) the second sub-block, wherein the second sub-
block has been transformed into a fourth domain using
the same block transform rule, and
wherein the encoded second block has been generated by
windowing (80) an overlapping block of audio samples
and by transforming a windowed block into a third
domain, wherein the encoded second block has an
aliasing portion corresponding to the aliasing portion
of the encoded first block of audio samples.
15. Method of encoding an audio signal, comprising:
windowing (11) a first block of the audio signal using
an analysis window, the analysis window having an
aliasing portion (Lk,Rk), and a further portion (Mk) ;
processing (12) a first sub-block (20) of the audio
signal associated with the aliasing portion by
transforming the first sub-block into a domain
different from the domain, in which the audio signal
is, subsequent to windowing the first sub-block to
obtain a processed first sub-block;
processing a second sub-block (21) of the audio signal
associated with the further portion by transforming
the second sub-block into the different domain before
windowing the second sub-block to obtain a processed
second sub-block;
converting (13) the processed first sub-block and the
processed second sub-block from the different domain

into a further domain using the same block transform
rule to obtain a converted first block; and
further processing (14) the converted first block
using a data compression algorithm.
16. Method of decoding an encoded audio signal having an
encoded first block of audio data, the encoded block
having an aliasing portion and a further portion,
comprising:
processing (51) the aliasing portion (Lk,Rk) by
transforming (86) the aliasing portion into a target
domain before performing a synthesis windowing (88) to
obtain a windowed aliasing portion;
a synthesis windowing (88) of the further portion
before performing a transform (98) into the target
domain; and
combining (53) the windowed aliasing portion and the
windowed aliasing portion of an encoded second block
of audio data to obtain a time-domain aliasing
cancellation, subsequent to a transform (91) of the
aliasing portion of the encoded first block of audio
data into the target domain to obtain a decoded audio
signal corresponding to the aliasing portion of the
first block.
17. Computer program having a program code for performing,
when running on a computer, the method for encoding in
accordance with claim 15 or the method of decoding in
accordance with claim 16.

An apparatus for encoding an audio signal comprises the
windower (11) for windowing a first block of the audio
signal using an analysis window having an aliasing portion
and a further portion. The apparatus furthermore comprises
a processor (12) for processing the first sub-block of the
audio signal associated with the aliasing portion by
transforming the sub-block from a domain into a different
domain subsequent to windowing the first sub-block to
obtain the processed first sub-block, and for processing a
second sub-block of the audio signal associated with the
further portion by transforming the second sub-block from
the domain into the different domain before windowing the
second sub-block to obtain a processed second sub-block.
The apparatus furthermore comprises a transformer (13) for
converting the processed first sub-block and the processed
second sub-block from the different domain into a further
different domain using the same block transform rule to
obtain a converted first block which may then be compressed
using any of the well-known data compression algorithms.
Thus, a critically sampled switch between two coding modes
can be obtained, since aliasing portions occurring in two
different domains are matched to each other.