Title of Invention

APPARATUS AND METHOD FOR CONSTRUCTING A MULTI-CHANNEL AUDIO SIGNAL

Abstract The invention relates to an apparatus for constructing a multi-channel output signal (34) using an input signal (28, 30) and parametric side information (26), the input signal comprising a first input channel (Lc, 28) and a second input channel (Rc, 30) derived from an original multi-channel signal (10), the original multi-channel signal (10) having a plurality of channels, the plurality of channels comprising at least two original channels, which are defined as being located at one side of an assumed listener position, wherein a first original channel is a first one of the at least two original channels, and wherein a second original channel is a second one of the at least two original channels, and the parametric side information (26) describing interrelations between original channels of the multi- channel original signal, comprising: means (322) for determining a first base channel by selecting one of the first and the second input channels (28, 30) or a combination of the first and the second input channels (28, 30), and for determining a second base channel by selecting the other of the first and the second input channels (28, 30)or a different combination of the first and the second input channels(28, 30), such that the second base channel is different from the first base channel; and means (324) for synthesizing a first output channel using the parametric side information (26) and the first base channel to obtain a first synthesized output channel (L) which is a reproduced version of the first original channel which is located at the one side of the assumed listener position, and for synthesizing a second output channel (R) using the parametric side information (26) and the second base channel, the second output channel (R) being a reproduced version of the second original channel which is located at the same side of the assumed listener position.
Full Text

Field of the invention
The present invention relates to an apparatus and a method
for processing a multi-channel audio signal and, in par-
ticular, to an apparatus and a method for processing a
multi-channel audio signal in a stereo-compatible manner.
Background of the Invention and Prior Art
In recent times, the multi-channel audio reproduction tech-
nique is becoming more and more important. This may be due
to the fact that audio compression/encoding techniques such
as the well-known mp3 technique have made it possible to
distribute audio records via the Internet or other trans-
mission channels having a limited bandwidth. The mp3 coding
technique has become so famous because of the fact that it
allows distribution of all the records in a stereo format,
i.e., a digital representation of the audio record includ-
ing a first or left stereo channel and a second or right
stereo channel.
Nevertheless, there are basic shortcomings of conventional
two-channel sound systems. Therefore, the surround tech-
nique has been developed. A recommended multi-channel-
surround representation includes, in addition to the two
stereo channels L and R, an additional center channel C and
two surround channels Ls, Rs. This reference sound format
is also referred to as three/two-stereo, which means three

front channels and two surround channels. Generally, five
transmission channels are required. In a playback environ-
ment, at least five speakers at the respective five differ-
ent places are needed to get an optimum sweet spot in a
certain distance from the five well-placed loudspeakers.
Several techniques are known in the art for reducing the
amount of data required for transmission of a multi-channel
audio signal. Such techniques are called joint stereo tech-
niques. To this end, reference is made to Fig. 10, which
shows a joint stereo device 60. This device can be a device
implementing e.g. intensity stereo (IS) or binaural cue
coding (BCC). Such a device generally receives - as an in-
put - at least two channels (CH1, CH2, ... CHn) , and outputs
a single carrier channel and parametric data. The paramet-
ric data are defined such that, in a decoder, an approxima-
tion of an original channel (CH1, CH2, ... CHn) can be calcu-
lated.
Normally, the carrier channel will include subband samples,
spectral coefficients, time domain samples etc, which pro-
vide a comparatively fine representation of the underlying
signal, while the parametric data do not include such sam-
ples of spectral coefficients but include control parame-
ters for controlling a certain reconstruction algorithm
such as weighting by multiplication, time shifting, fre-
quency shifting, ... The parametric data, therefore, include
only a comparatively coarse representation of the signal or
the associated channel. Stated in numbers, the amount of
data required by a carrier channel will be in the range of
60 - 70 kbit/s, while the amount of data required by para-
metric side information for one channel will be in the
range of 1,5 - 2,5 kbit/s. An example for parametric data

are the well-known scale factors, intensity stereo informa-
tion or binaural cue parameters as will be described below.
Intensity stereo coding is described in AES preprint 3799,
"Intensity Stereo Coding", J. Herre, K. H. Brandenburg, D.
Lederer, February 1994, Amsterdam. Generally, the concept
of intensity stereo is based on a main axis transform to be
applied to the data of both stereophonic audio channels. If
most of the data points are concentrated around the first
principle axis, a coding gain can be achieved by rotating
both signals by a certain angle prior to coding. This is,
however, not always true for real stereophonic production
techniques. Therefore, this technique is modified by ex-
cluding the second orthogonal component from transmission
in the bit stream. Thus, the reconstructed signals for the
left and right channels consist of differently weighted or
scaled versions of the same transmitted signal. Neverthe-
less, the reconstructed signals differ in their amplitude
but are identical regarding their phase information. The
energy-time envelopes of both original audio channels, how-
ever, are preserved by means of the selective scaling op-
eration, which typically operates in a frequency selective
manner. This conforms to the human perception of sound at
high frequencies, where the dominant spatial cues are de-
termined by the energy envelopes.
Additionally, in practically implementations, the transmit-
ted signal, i.e. the carrier channel is generated from the
sum signal of the left channel and the right channel in-
stead of rotating both components. Furthermore, this proc-
essing, i.e., generating intensity stereo parameters for
performing the scaling operation, is performed frequency
selective, i.e., independently for each scale factor band,

i.e., encoder frequency partition. Preferably, both chan-
nels are combined to form a combined or "carrier" channel,
and, in addition to the combined channel, the intensity
stereo information is determined which depend on the energy
of the first channel, the energy of the second channel or
the energy of the combined or channel.
The BCC technique is described in AES convention paper
5574, "Binaural cue coding applied to stereo and multi-
channel audio compression", C. Faller, F. Baumgarte, May
2002, Munich. In BCC encoding, a number of audio input
channels are converted to a spectral representation using a
DFT based transform with overlapping windows. The resulting
uniform spectrum is divided into non-overlapping partitions
each having an index. Each partition has a bandwidth pro-
portional to the equivalent rectangular bandwidth (ERB).
The inter-channel level differences (ICLD) and the inter-
channel time differences (ICTD) are estimated for each par-
tition for each frame k. The ICLD and ICTD are quantized
and coded resulting in a BCC bit stream. The inter-channel
level differences and inter-channel time differences are
given for each channel relative to a reference channel.
Then, the parameters are calculated in accordance with pre-
scribed formulae, which depend on the certain partitions of
the signal to be processed.
At a decoder-side, the decoder receives a mono signal and
the BCC bit stream. The mono signal is transformed into the
frequency domain and input into a spatial synthesis block,
which also receives decoded ICLD and ICTD values. In the
spatial synthesis block, the BCC parameters (ICLD and ICTD)
values are used to perform a weighting operation of the
mono signal in order to synthesize the multi-channel sig-

nals, which, after a frequency/time conversion, represent a
reconstruction of the original multi-channel audio signal.
In case of BCC, the joint stereo module 60 is operative to
output the channel side information such that the paramet-
ric channel data are quantized and encoded ICLD or ICTD pa-
rameters, wherein one of the original channels is used as
the reference channel for coding the channel side informa-
tion.
Normally, the carrier channel is formed of the sum of the
participating original channels.
Naturally, the above techniques only provide a mono repre-
sentation for a decoder, which can only process the carrier
channel, but is not able to process the parametric data for
generating one or more approximations of more than one in-
put channel.
The audio coding technique known as binaural cue coding
(BCC) is also well described in the United States patent
application publications US 2003, 0219130 A1, 2003/0026441
A1 and 2003/0035553 Al. Additional reference is also made
to "Binaural Cue Coding. Part II: Schemes and Applica-
tions", C. Faller and F. Baumgarte, IEEE Trans. On Audio
and Speech Proc, Vol. 11, No. 6, Nov. 2993. The cited
United States patent application publications and the two
cited technical publications on the BCC technique authored
by Faller and Baumgarte are incorporated herein by refer-
ence in their entireties.
In the following, a typical generic BCC scheme for multi-
channel audio coding is elaborated in more detail with ref-

erence to Figures 11 to 13. Figure 11 shows such a generic
binaural cue coding scheme for coding/transmission of
multi-channel audio signals. The multi-channel audio input
signal at an input 110 of a BCC encoder 112 is downmixed in
a downmix block 114. In the present example, the original
multi-channel signal at the input 110 is a 5-channel sur-
round signal having a front left channel, a front right
channel, a left surround channel, a right surround channel
and a center channel. In a preferred embodiment of the pre-
sent invention, the downmix block 114 produces a sum signal
by a simple addition of these five channels into a mono
signal. Other downmixing schemes are known in the art such
that, using a multi-channel input signal, a downmix signal
having a single channel can be obtained. This single chan-
nel is output at a sum signal line 115. A side information
obtained by a BCC analysis block 116 is output at a side
information line 117. In the BCC analysis block, inter-
channel level differences (ICLD), and inter-channel time
differences (ICTD) are calculated as has been outlined
above. Recently, the BCC analysis block 116 has been en-
hanced to also calculate inter-channel correlation values
(ICC values) . The sum signal and the side information is
transmitted, preferably in a quantized and encoded form, to
a BCC decoder 120. The BCC decoder decomposes the transmit-
ted sum signal into a number of subbands and applies scal-
ing, delays and other processing to generate the subbands
of the output multi-channel audio signals. This processing
is performed such that ICLD, ICTD and ICC parameters (cues)
of a reconstructed multi-channel signal at an output 121
are similar to the respective cues for the original multi-
channel signal at the input 110 into the BCC encoder 112.
To this end, the BCC decoder 120 includes a BCC synthesis
block 122 and a side information processing block 123.

In the following, the internal construction of the BCC syn-
thesis block 122 is explained with reference to Fig. 12.
The sum signal on line 115 is input into a time/frequency
conversion unit or filter bank FB 125. At the output of
block 125, there exists a number N of sub band signals or,
in an extreme case, a block of a spectral coefficients,
when the audio filter bank 125 performs a 1:1 transform,
i.e., a transform which produces N spectral coefficients
from N time domain samples.
The BCC synthesis block 122 further comprises a delay stage
126, a level modification stage 127, a correlation process-
ing stage 128 and an inverse filter bank stage IFB 129. At
the output of stage 129, the reconstructed multi-channel
audio signal having for example five channels in case of a
5-channel surround system, can be output to a set of loud-
speakers 124 as illustrated in Fig. 11.
As shown in Fig. 12, the input signal s (n) is converted
into the frequency domain or filter bank domain by means of
element 125. The signal output by element 125 is multiplied
such that several versions of the same signal are obtained
as illustrated by multiplication node 130. The number of
versions of the original signal is equal to the number of
output channels in the output signal, to be reconstructed
When, in general, each version of the original signal at
node 130 is subjected to a certain delay d1, d2, —, di, ...,
dN. The delay parameters are computed by the side informa-
tion processing block 123 in Fig. 11 and are derived from
the inter-channel time differences as determined by the BCC
analysis block 116.

The same is true for the multiplication parameters al, a2,
..., ai, ..., aN, which are also calculated by the side infor-
mation processing block 123 based on the inter-channel
level differences as calculated by the BCC analysis block
116.
The ICC parameters calculated by the BCC analysis block 116
are used for controlling the functionality of block 128
such that certain correlations between the delayed and
level-manipulated signals are obtained at the outputs of
block 128. It is to be noted here that the ordering of the
stages 126, 127, 128 may be different from the case shown
in Fig. 12.
It is to be noted here that, in a frame-wise processing of
an audio signal, the BCC analysis is performed frame-wise,
i.e. time-varying, and also frequency-wise. This means
that, for each spectral band, the BCC parameters are ob-
tained. This means that, in case the audio filter bank 125
decomposes the input signal into for example 32 band pass
signals, the BCC analysis block obtains a set of BCC pa-
rameters for each of the 32 bands. Naturally the BCC syn-
thesis block 122 from Fig. 11, which is shown in detail in
Fig. 12, performs a reconstruction which is also based on
the 32 bands in the example.
In the following, reference is made to Fig. 13 showing a
setup to determine certain BCC parameters. Normally, ICLD,
ICTD and ICC parameters can be defined between pairs of
channels. However, it is preferred to determine ICLD and
ICTD parameters between a reference channel and each other
channel. This is illustrated in Fig. 13A.

ICC parameters can be defined in different ways. Most gen-
erally, one could estimate ICC parameters in the encoder
between all possible channel pairs as indicated in Fig.
13B. In this case, a decoder would synthesize ICC such that
it is approximately the same as in the original multi-
channel signal between all possible channel pairs. It was,
however, proposed to estimate only ICC parameters between
the strongest two channels at each time. This scheme is il-
lustrated in Fig. 13C, where an example is shown, in which
at one time instance, an ICC parameter is estimated between
channels 1 and 2, and, at another time instance, an ICC pa-
rameter is calculated between channels 1 and 5. The decoder
then synthesizes the inter-channel correlation between the
strongest channels in the decoder and applies some heuris-
tic rule for computing and synthesizing the inter-channel
coherence for the remaining channel pairs.
Regarding the calculation of, for example, the multiplica-
tion parameters ai, aN based on transmitted ICLD parame-
ters, reference is made to AES convention paper 5574 cited
above. The ICLD parameters represent an energy distribution
in an original multi-channel signal. Without loss of gener-
ality, it is shown in Fig. 13A that there are four ICLD pa-
rameters showing the energy difference between all other
channels and the front left channel. In the side informa-
tion processing block 123, the multiplication parameters
a1, ..., aN are derived from the ICLD parameters such that the
total energy of all reconstructed output channels is the
same as (or proportional to) the energy of the transmitted
sum signal. A simple way for determining these parameters
is a 2-stage process, in which, in a first stage, the mul-
tiplication factor for the left front channel is set to
unity, while multiplication factors for the other channels

in Fig. 13A are set to the transmitted ICLD values. Then,
in a second stage, the energy of all five channels is cal-
culated and compared to the energy of the transmitted sum
signal. Then, all channels are downscaled using a down-
scaling factor which is equal for all channels, wherein the
downscaling factor is selected such that the total energy
of all reconstructed output channels is, after downscaling,
equal to the total energy of the transmitted sum signal.
Naturally, there are other methods for calculating the mul-
tiplication factors, which do not rely on the 2-stage proc-
ess but which only need a 1-stage process.
Regarding the delay parameters, it is to be noted that the
delay parameters ICTD, which are transmitted from a BCC en-
coder can be used directly, when the delay parameter d1 for
the left front channel is set to zero. No rescaling has to
be done here, since a delay does not alter the energy of
the signal.
Regarding the inter-channel coherence measure ICC transmit-
ted from the BCC encoder to the BCC decoder, it is to be
noted here that a coherence manipulation can be done by
modifying the multiplication factors a1, ..., an such as by
multiplying the weighting factors of all subbands with ran-
dom numbers with values between 201ogl0(-6) and 201ogl0(6).
The pseudo-random sequence is preferably chosen such that
the variance is approximately constant for all critical
bands, and the average is zero within each critical band.
The same sequence is applied to the spectral coefficients
for each different frame. Thus, the auditory image width is
controlled by modifying the variance of the pseudo-random
sequence. A larger variance creates a larger image width.

The variance modification can be performed in individual
bands that are critical-band wide. This enables the simul-
taneous existence of multiple objects in an auditory scene,
each object having a different image width. A suitable am-
plitude distribution for the pseudo-random sequence is a
uniform distribution on a logarithmic scale as it is out-
lined in the US patent application publication 2003/0219130
A1. Nevertheless, all BCC synthesis processing is related
to a single input channel transmitted as the sum signal
from the BCC encoder to the BCC decoder as shown in Fig.
11.
To transmit the five channels in a compatible way, i.e., in
a bitstream format, which is also understandable for a nor-
mal stereo decoder, the so-called matrixing technique has
been used as described in "MUSICAM surround: a universal
multi-channel coding system compatible with ISO 11172-3",
G. Theile and G. Stoll, AES preprint 3403, October 1992,
San Francisco. The five input channels L, R, C, Ls, and Rs
are fed into a matrixing device performing a matrixing op-
eration to calculate the basic or compatible stereo chan-
nels Lo, Ro, from the five input channels. In particular,
these basic stereo channels Lo/Ro are calculated as set out
below:

x and y are constants. The other three channels C, Ls, Rs
are transmitted as they are in an extension layer, in addi-
tion to a basic stereo layer, which includes an encoded
version of the basic stereo signals Lo/Ro. With respect to

the bitstream, this Lo/Ro basic stereo layer includes a
header, information such as scale factors and subband sam-
ples. The multi-channel extension layer, i.e., the central
channel and the two surround channels are included in the
multi-channel extension field, which is also called ancil-
lary data field.
At a decoder-side, an inverse matrixing operation is per-
formed in order to form reconstructions of the left and
right channels in the five-channel representation using the
basic stereo channels Lo, Ro and the three additional chan-
nels. Additionally, the three additional channels are de-
coded from the ancillary information in order to obtain a
decoded five-channel or surround representation of the
original multi-channel audio signal.
Another approach for multi-channel encoding is described in
the publication "Improved MPEG-2 audio multi-channel encod-
ing", B. Grill, J. Herre, K. H. Brandenburg, E. Eberlein,
J. Roller, J. Mueller, AES preprint 3865, February 1994,
Amsterdam, in which, in order to obtain backward compati-
bility, backward compatible modes are considered. To this
end, a compatibility matrix is used to obtain two so-called
downmix channels Lc, Rc from the original five input chan-
nels. Furthermore, it is possible to dynamically select
the three auxiliary channels transmitted as ancillary data.
In order to exploit stereo irrelevancy, a joint stereo
technique is applied to groups of channels, e. g. the three
front channels, i.e., for the left channel, the right chan-
nel and the center channel. To this end, these three chan-
nels are combined to obtain a combined channel. This com-
bined channel is quantized and packed into the bitstream.

Then, this combined channel together with the corresponding
joint stereo information is input into a joint stereo de-
coding module to obtain joint stereo decoded channels,
i.e., a joint stereo decoded left channel, a joint stereo
decoded right channel and a joint stereo decoded center
channel. These joint stereo decoded channels are, together
with the left surround channel and the right surround chan-
nel input into a compatibility matrix block to form the
first and the second downmix channels Lc, Rc. Then, quan-
tized versions of both downmix channels and a quantized
version of the combined channel are packed into the bit-
stream together with joint stereo coding parameters.
Using intensity stereo coding, therefore, a group of inde-
pendent original channel signals is transmitted within a
single portion of "carrier" data. The decoder then recon-
structs the involved signals as identical data, which are
rescaled according to their original energy-time envelopes.
Consequently, a linear combination of the transmitted chan-
nels will lead to results, which are quite different from
the original downmix. This applies to any kind of joint
stereo coding based on the intensity stereo concept. For a
coding system providing compatible downmix channels, there
is a direct consequence: The reconstruction by dematrixing,
as described in the previous publication, suffers from ar-
tifacts caused by the imperfect reconstruction. Using a so-
called joint stereo predistortion scheme, in which a joint
stereo coding of the left, the right and the center chan-
nels is performed before matrixing in the encoder, allevi-
ates this problem. In this way, the dematrixing scheme for
reconstruction introduces fewer artifacts, since, on the
encoder-side, the joint stereo decoded signals have been
used for generating the downmix channels. Thus, the imper-

feet reconstruction process is shifted into the compatible
downmix channels Lc and Rc, where it is much more likely to
be masked by the audio signal itself.
Although such a system has resulted in fewer artifacts be-
cause of dematrixing on the decoder-side, it nevertheless
has some drawbacks. A drawback is that the stereo-
compatible downmix channels Lc and Rc are derived not from
the original channels but from intensity stereo
coded/decoded versions of the original channels. Therefore,
data losses because of the intensity stereo coding system
are included in the compatible downmix channels. Astereo-
only decoder, which only decodes the compatible channels
rather than the enhancement intensity stereo encoded chan-
nels, therefore, provides an output signal, which is af-
fected by intensity stereo induced data losses.
Additionally, a full additional channel has to be transmit-
ted besides the two downmix channels. This channel is the
combined channel, which is formed by means of joint stereo
coding of the left channel, the right channel and the cen-
ter channel. Additionally, the intensity stereo information
to reconstruct the original channels L, R, C from the com-
bined channel also has to be transmitted to the decoder. At
the decoder, an inverse matrixing, i.e., a dematrixing op-
eration is performed to derive the surround channels from
the two downmix channels. Additionally, the original left,
right and center channels are approximated by joint stereo
decoding using the transmitted combined channel and the
transmitted joint stereo parameters. It is to be noted that
the original left, right and center channels are derived by
joint stereo decoding of the combined channel.

It has been found out that in case of intensity stereo
techniques, when used in combination with multi-channel
signals, only fully coherent output signals which are based
on the same base channel can be produced.
In BCC techniques, it is quite expensive to reduce the in-
ter-channel coherence in a reconstructed multi-channel out-
put signal, since a pseudo-random number generator for in-
fluencing the weighting sectors is required. Additionally,
it has been shown that this kind of processing is problem-
atic in that artifacts because of randomly manipulating
multiplication factors or time delay factors can be intro-
duced which can become audible under certain circumstances
and, therefore, deteriorate the quality of the recon-
structed multi-channel output signal.
Summary of the Invention
It is, therefore, an object of the present invention to
provide a concept for a bit-efficient and artifact-reduced
processing or inverse processing of a multi-channel audio
signal.
In accordance with the first aspect of the present inven-
tion, this object is achieved by an apparatus for con-
structing a multi-channel output signal using an input sig-
nal and parametric side information, the input signal in-
cluding a first input channel and a second input channel
derived from an original multi-channel signal, the original
multi-channel signal having a plurality of channels, the
plurality of channels including at least two original chan-
nels, which are defined as being located at one side of an

assumed listener position, wherein a first original channel
is a first one of the at least two original channels, and
wherein a second original channel is a second one of the at
least two original channels, and the parametric side infor-
mation describing interrelations betweens original channels
of the multi-channel original signal, comprising: original
multi-channel signal; means for determining a first base
channel by selecting one of the first and the second input
channels or a combination of the first and the second input
channels, and for determining a second base channel by se-
lecting the other of the first and the second input chan-
nels or a different combination of the first and the second
input channels, such that the second base channel is dif-
ferent from the first base channel; and means for synthe-
sizing a first output channel using the parametric side in-
formation and the first base channel to obtain a first syn-
thesized output channel which is a reproduced version of
the first original channel which is located at the one side
of the assumed listener position, and for synthesizing a
second output channel using the parametric side information
and the second base channel, the second output channel be-
ing a reproduced version of the second original channel
which is located at the same side of the assumed listener
position.
In accordance with the second aspect of the present inven-
tion, this object is achieved by a method of constructing a
multi-channel output signal using an input signal and para-
metric side information, the input signal including a first
input channel and a - second input channel derived from an
original multi-channel signal, the original multi-channel
signal having a plurality of channels, the plurality of
channels including at least two original channels, which

are defined as being located at one side of an assumed lis-
tener position, wherein a first original channel is a first
one of the at least two original channels, and wherein a
second original channel is a second one of the at least two
original channels, and the parametric side information de-
scribing interrelations betweens original channels of the
multi-channel original signal, comprising: determining a
first base channel by selecting one of the first and the
second input channels or a combination of the first and the
second input channels, and determining a second base chan-
nel by selecting the other of the first and the second in-
put channels or a different combination of the first and
the second input channels, such that the second base chan-
nel is different from the first base channel; and synthe-
sizing a first output channel using the parametric side in-
formation and the first base channel to obtain a first syn-
thesized output channel which is a reproduced version of
the first original channel which is located at the one side
of the assumed listener position, and synthesizing a second
output channel using the parametric side information and
the second base channel, the second output channel being a
reproduced version of the second original channel which is
located at the same side of the assumed listener position.
In accordance with the third aspect of the present inven-
tion, this object is achieved by an apparatus for generat-
ing a downmix signal from a multi-channel original signal,
the downmix signal having a number of channels being
smaller than a number of original channels, comprising:
means for calculating a first downmix channel and a second
downmix channel using a downmix rule; means for calculating
parametric level information representing an energy distri-
bution among the channels in the multi-channel original

signal; means for determining a coherence measure between
two original channels, the two original channels being lo-
cated at one side of an assumed listener position; and
means for forming an output signal using the first and the
second downmix channels, the parametric level information
and only at least one coherence measure between two origi-
nal channels located at the one side or a value derived
from the at least one coherence measure, but not using any
coherence measure between channels located at different
sides of the assumed listener position.
In accordance with a fourth aspect of the present inven-
tion, this object is achieved by a method for generating a
downmix signal from a multi-channel original signal, the
downmix signal having a number of channels being smaller
than a number of original channels, comprising: calculating
a first downmix channel and a second downmix channel using
a downmix rule; calculating parametric level information
representing an energy distribution among the channels in
the multi-channel original signal; determining a coherence
measure between two original channels, the two original
channels being located at one side of an assumed listener
position; and forming an output signal using the first and
the second downmix channels, the parametric level informa-
tion and only at least one coherence measure between two
original channels located at the one side or a value de-
rived from the at least one coherence measure, but not us-
ing any coherence measure between channels located at dif-
ferent sides of the assumed listener position.
In accordance with a fifth aspect and a sixth aspect of the
present invention, this object is achieved by a computer
program including the method for constructing the multi-

channel output signal or the method of generating a downmix
signal.
The present invention is based on the finding that an effi-
cient and artifact-reduced reconstruction of a multi-
channel output signal is obtained, when there are two or
more channels, which can be transmitted from an encoder to
a decoder, wherein the channels which are preferably a left
and a right stereo channel, show a certain degree of inco-
herence. This will normally be the case, since the left and
right stereo channels or the left and right compatible ste-
reo channels as obtained by downmixing a multi-channel sig-
nal will usually show a certain degree of incoherence,
i.e., will not be fully coherent or fully correlated.
In accordance with the present invention, the reconstructed
output channels of the multi-channel output signal are de-
correlated from each other by determining different base
channels for the different output channels, wherein the
different base channels are obtained by using varying de-
grees of the uncorrelated transmitted channels.
In other words, a reconstructed output channel having, for
example, the left transmitted input channel as a base chan-
nel would be - in the BCC subband domain - fully correlated
with another reconstructed output channel which has the
same e.g. left channel as the base channel assuming no ex-
tra "correlation synthesis". In this context, it is to be
noted that deterministic delay and level settings do not
reduce coherence between these channels. In accordance with
the present invention, the coherence between these chan-
nels, which is 100 % in the above example is reduced to a
certain coherence degree or coherence measure by using a

first base channel for constructing the first output chan-
nel and for using a second base channel for constructing
the second output channel, wherein the first and second
base channels have different "portions" of the two trans-
mitted (de-correlated) channels. This means that the first
base channel is influenced stronger by the first transmit-
ted or is even identical to the first transmitted channel,
compared to the second base channel which is influenced
less by the first channel, i.e., which is more influenced
by the second transmitted channel.
In accordance with the present invention, inherent de-
correlation between the transmitted channels is used for
providing de-correlated channels in a multi-channel output
signal.
In a preferred embodiment, a coherence measure between re-
spective channel pairs such as front left and left surround
or front right and right surround is determined in an en-
coder in a time-dependent and frequency-dependent way and
transmitted as side information, to an inventive decoder
such that a dynamic determination of base channels and,
therefore, a dynamic manipulation of coherence between the
reconstructed output channels can be obtained.
Compared to the above mentioned prior art case, in which
only an ICC cue for the two strongest channels is transmit-
ted, the inventive system is easier to control and provides
a better quality reconstruction, since no determination of
the strongest channels in an encoder or a decoder are nec-
essary, since the inventive coherence measure always re-
lates to the same channel pair irrespective of the fact,
whether this channel pair includes the strongest channels

or not. Higher quality compared to the prior art systems is
obtained in that two downmixed channels are transmitted
from an encoder to a decoder such that the left/right co-
herence relation is automatically transmitted such that no
extra information on a left/right coherence is required.
A further advantage of the present invention has to be seen
in the fact that a decoder-side computing workload can be
reduced, since the normal decorrelation processing load can
be reduced or even completely eliminated.
Preferably, parametric channel side information for one or
more of the original channels are derived such that they
relate to one of the downmix channels rather than, as in
the prior art, to an additional "combined" joint stereo
channel. This means that the parametric channel side infor-
mation are calculated such that, on a decoder side, a chan-
nel reconstructor uses the channel side information and one
of the downmix channels or a combination of the downmix
channels to reconstruct an approximation of the original
audio channel, to which the channel side information is as-
signed.
This concept is advantageous in that it provides a bit-
efficient multi-channel extension such that a multi-channel
audio signal can be played at a decoder.
Additionally, the concept is backward compatible, since a
lower scale decoder, which is only adapted for two-channel
processing, can simply ignore the extension information,
i.e., the channel side information. The lower scale decoder
can only play the two downmix channels to obtain a stereo
representation of the original multi-channel audio signal.

A higher scale decoder, however, which is enabled for
multi-channel operation, can use the transmitted channel
side information to reconstruct approximations of the
original channels.
The present embodiment is advantageous in that it is bit-
efficient, since, in contrast to the prior art, no addi-
tional carrier channel beyond the first and second downmix
channels Lc, Rc is required. Instead, the channel side in-
formation are related to one or both downmix channels. This
means that the downmix channels themselves serve as a car-
rier channel, to which the channel side information are
combined to reconstruct an original audio channel. This
means that the channel side information are preferably pa-
rametric side information, i.e., information which do not
include any subband samples or spectral coefficients. In-
stead, the parametric side information are information used
for weighting (in time and/or frequency) the respective
downmix channel or the combination of the respective down-
mix channels to obtain a reconstructed version of a se-
lected original channel.
In a preferred embodiment of the present invention, a back-
ward compatible coding of a multi-channel signal based on a
compatible stereo signal is obtained. Preferably, the com-
patible stereo signal (downmix signal) is generated using
matrixing of the original channels of multi-channel audio
signal.
Preferably, channel side information for a selected origi-
nal channel is obtained based on joint stereo techniques
such as intensity stereo coding or binaural cue coding.
Thus, at the decoder side, no dematrixing operation has to

be performed. The problems associated with dematrixing,
i.e., certain artifacts related to an undesired distribu-
tion of quantization noise in dematrixing operations, are
avoided. This is due to the fact that the decoder uses a
channel reconstructor, which reconstructs an original sig-
nal, by using one of the downmix channels or a combination
of the downmix channels and the transmitted channel side
information.
Preferably, the inventive concept is applied to a multi-
channel audio signal having five channels. These five chan-
nels are a left channel L, a right channel R, a center
channel C, a left surround channel Ls, and a right surround
channel Rs. Preferably, downmix channels are stereo com-
patible downmix channels Ls and Rs, which provide a stereo
representation of the original multi-channel audio signal.
In accordance with the preferred embodiment of the present
invention, for each original channel, channel side informa-
tion are calculated at an encoder side packed into output
data. Channel side information for the original left chan-
nel are derived using the left downmix channel. Channel
side information for the original left surround channel are
derived using the left downmix channel. Channel side infor-
mation for the original right channel are derived from the
right downmix channel. Channel side information for the
original right surround channel are derived from the right
downmix channel.
In accordance with the preferred embodiment of the present
invention, channel information for the original center
channel are derived using the first downmix channel as well
as the second downmix channel, i.e., using a combination of

the two downmix channels. Preferably, this combination is a
summation.
Thus, the groupings, i.e., the relation between the channel
side information and the carrier signal, i.e., the used
downmix channel for providing channel side information for
a selected original channel are such that, for optimum
quality, a certain downmix channel is selected, which con-
tains the highest possible relative amount of the respec-
tive original multi-channel signal which is represented by
means of channel side information. As such a joint stereo
carrier signal, the first and the second downmix channels
are used. Preferably, also the sum of the first and the
second downmix channels can be used. Naturally, the sum of
the first and second downmix channels can be used for cal-
culating channel side information for each of the original
channels. Preferably, however, the sum of the downmix chan-
nels is used for calculating the channel side information
of the original center channel in a surround environment,
such as five channel surround, seven channel surround, 5.1
surround or 7.1 surround. Using the sum of the first and
second downmix channels is especially advantageous, since
no additional transmission overhead has to be performed.
This is due to the fact that both downmix channels are pre-
sent at the decoder such that summing of these downmix
channels can easily be performed at the decoder without re-
quiring any additional transmission bits.
Preferably, the channel side information forming the multi-
channel extension are input into the output data bit stream
in a compatible way such that a lower scale decoder simply
ignores the multi-channel extension data and only provides
a stereo representation of the multi-channel audio signal.

Nevertheless, a higher scale encoder not only uses two
downmix channels, but, in addition, employs the channel
side information to reconstruct a full multi-channel repre-
sentation of the original audio signal.
Brief Description of the Accompanying Drawings
Preferred embodiments of the present invention are subse-
quently described by referring to the enclosed drawings, in
which:
Fig. 1A is a block diagram of a preferred embodiment of
the inventive encoder;
Fig. 1B is a block diagram of an inventive encoder for
providing a coherence measure for respective in-
put channel pairs.
Fig. 2A is a block diagram of a preferred embodiment of
the inventive decoder;
Fig. 2B is a block diagram of an inventive decoder having
different base channels for different output
channels;
Fig. 2C is a block diagram of a preferred embodiment of
the means for synthesizing of Fig. 2B;
Fig. 2D is a block diagram of a preferred embodiment of
apparatus shown in Fig. 2C for a 5-channel sur-
round system;

Fig. 2E is a schematic representation of a means for de-
termining a coherence measure in an inventive en-
coder;
Fig. 2F is a schematic representation of a preferred ex-
ample for determining a weighting factor for cal-
culating a base channel having a certain coher-
ence measure with respect to another base chan-
nel;
Fig. 2G is a schematic diagram of a preferred way to ob-
tain a reconstructed output channel based on a
certain weighting factor calculated by the scheme
shown in Fig. 2F;
Fig. 3A is a block diagram for a preferred implementation
of the means for calculating to obtain frequency
selective channel side information;
Fig. 3B is a preferred embodiment of a calculator imple-
menting joint stereo processing such as intensity
coding or binaural cue coding;
Fig, 4 illustrates another preferred embodiment of the
means for calculating channel side information,
in which the channel side information are gain
factors;
Fig. 5 illustrates a preferred embodiment of an imple-
mentation of the decoder, when the encoder is im-
plemented as in Fig. 4;

Fig. 6 illustrates a preferred implementation of the
means for providing the downmix channels;
Fig. 7 illustrates groupings of original and downmix
channels for calculating the channel side infor-
mation for the respective original channels;
Fig. 8 illustrates another preferred embodiment of an
inventive encoder;
Fig. 9 illustrates another implementation of an inven-
tive decoder; and
Fig. 10 illustrates a prior art joint stereo encoder.
Fig. 11 is a block diagram representation of a prior art
BCC encoder/decoder chain?;
Fig. 12 is a block diagram of a prior art implementation
of a BCC synthesis block of Fig. 11;
Fig. 13 is a representation of a well-known scheme for
determining ICLD, ICTD and ICC parameters;
Fig. 14A is a schematic representation of the scheme for
attributing different base channels for the re-
production of different output channels;
Fig. 14B is a representation of the channel pairs neces-
sary for determining ICC and ICTD parameters;

Fig. 15A a schematic representation of a first selection
of base channels for constructing a 5-channel
output signal; and
Fig. 15B a schematic representation of a second selection
of base channels for constructing a 5-channel
output signal.
Detailed Description of Preferred Embodiments
Fig. 1A shows an apparatus for processing a multi-channel
audio signal 10 having at least three original channels
such as R, L and C. Preferably, the original audio signal
has more than three channels, such as five channels in the
surround environment, which is illustrated in Fig. 1A. The
five channels are the left channel L, the right channel R,
the center channel C, the left surround channel Ls and the
right surround channel Rs. The inventive apparatus includes
means 12 for providing a first downmix channel Lc and a
second downmix channel Rc, the first and the second downmix
channels being derived from the original channels. For de-
riving the downmix channels from the original channels,
there exist several possibilities. One possibility is to
derive the downmix channels Lc and Rc by means of matrixing
the original channels using a matrixing operation as illus-
trated in Fig. 6. This matrixing operation is performed in
the time domain.
The matrixing parameters a, b and t are selected such that
they are lower than or equal to 1. Preferably, a and b are
0.7 or 0.5. The overall weighting parameter t is preferably
chosen such that channel clipping is avoided.

Alternatively, as it is indicated in Fig. 1A, the downmix
channels Lc and Rc can also be externally supplied. This
may be done, when the downmix channels Lc and Rc are the
result of a "hand mixing" operation. In this scenario, a
sound engineer mixes the downmix channels by himself rather
than by using an automated matrixing operation. The sound
engineer performs creative mixing to get optimized downmix
channels Lc and Rc which give the best possible stereo rep-
resentation of the original multi-channel audio signal.
In case of an external supply of the downmix channels, the
means for providing does not perform a matrixing operation
but simply forwards the externally supplied downmix chan-
nels to a subsequent calculating means 14.
The calculating means 14 is operative to calculate the
channel side information such as li, lsi, ri or rsi for se-
lected original channels such as L, Ls, R or Rs, respec-
tively. In particular, the means 14 for calculating is op-
erative to calculate the channel side information such that
a downmix channel, when weighted using the channel side in-
formation, results in an approximation of the selected
original channel.
Alternatively or additionally, the means for calculating
channel side information is further operative to calculate
the channel side information for a selected original chan-
nel such that a combined downmix channel including a combi-
nation of the first and second downmix channels, when
weighted using the calculated channel side information re-
sults in an approximation of the selected original channel.

To show this feature in the figure, an adder 14a and a com-
bined channel side information calculator 14b are shown.
It is clear for those skilled in the art that these ele-
ments do not have to be implemented as distinct elements.
Instead, the whole functionality of the blocks 14, 14a, and
14b can be implemented by means of a certain processor
which may be a general purpose processor or any other means
for performing the required functionality.
Additionally, it is to be noted here that channel signals
being subband samples or frequency domain values are indi-
cated in capital letters. Channel side information are, in
contrast to the channels themselves, indicated by small
letters. The channel side information Ci is, therefore, the
channel side information for the original center channel C.
The channel side information as well as the downmix chan-
nels Lc and Rc or an encoded version Lc' and Rc' as pro-
duced by an audio encoder 16 are input into an output data
formatter 18. Generally, the output data formatter 18 acts
as means for generating output data, the output data in-
cluding the channel side information for at least one
original channel, the first downmix channel or a signal de-
rived from the first downmix channel (such as an encoded
version thereof) and the second downmix channel or a signal
derived from the second downmix channel (such as an encoded
version thereof).
The output data or output bitstream 20 can then be trans-
mitted to a bitstream decoder or can be stored or distrib-
uted. Preferably, the output bitstream 20 is a compatible
bitstream which can also be read by a lower scale decoder

not having a multi-channel extension capability. Such lower
scale encoders such as most existing normal state of the
art mp3 decoders will simply ignore the multi-channel ex-
tension data, i.e., the channel side information. They will
only decode the first and second downmix channels to pro-
duce a stereo output. Higher scale decoders, such as multi-
channel enabled decoders will read the channel side infor-
mation and will then generate an approximation of the
original audio channels such that a multi-channel audio im-
pression is obtained.
Fig. 8 shows a preferred embodiment of the present inven-
tion in the environment of five channel surround / mp3.
Here, it is preferred to write the surround enhancement
data into the ancillary data field in the standardized mp3
bit stream syntax such that an "mp3 surround" bit stream is
obtained.
Fig. 1B illustrates a more detailed representation of ele-
ment 14 in Fig. 1A. In a preferred embodiment of the pre-
sent invention, a calculator 14 includes means 141 for cal-
culating parametric level information representing an en-
ergy distribution among the channels in the multi channel
original signal shown at 10 in Fig. 1A. Element 141 there-
fore is able to generate output level information for all
original channels. In a preferred embodiment, this level
information includes ICLD parameters obtained by regular
BCC synthesis as has been described in connection with
Figs. 10 to 13.
Element 14 further comprises means 142 for determining a
coherence measure between two original channels located at
one side of an assumed listener position. In case of the 5-

channel surround example shown in Fig. 1A, such a channel
pair includes the right channel R and the right surround
channel Rs or, alternatively or additionally the left chan-
nel L and the left surround channel Ls. Element 14 alterna-
tively further comprises means 143 for calculating the time
difference for such a channel pair, i.e., a channel pair
having channels which are located at one side of an assumed
listener position.
The output data formatter 18 from Fig. 1A is operative to
input into the data stream at 20 the level information rep-
resenting an energy distribution among the channels in the
multi channel original signal and a coherence measure only
for the left and left surround channel pair and/or the
right and the right surround channel pair. The output data
formatter, however, is operative to not include any other
coherence measures or optionally time differences into the
output signal such that the amount of side information is
reduced compared to the prior art scheme in which ICC cues
for all possible channel pairs were transmitted.
To illustrate the inventive encoder as shown in Fig. 1B in
more detail, reference is made to Fig. 14A and Fig. 14B. In
Fig. 14A, an arrangement of channel speakers for an example
5-channel system is given with respect to a position of an
assumed listener position which is located at the center
point of a circle on which the respective speakers are
placed. As outlined above, the 5-channel system includes a
left surround channel, a left channel, a center channel, a
right channel and a right surround channel. Naturally, such

It is to be noted here that the left surround channel can
also be termed as "rear left channel". The same is true for
the right surround channel. This channel is also known as
the rear right channel.
In contrast to state of the art BCC with one transmission
channel, in which the same base channel, i.e., the trans-
mitted mono signal as shown in Fig. 11 is used for generat-
ing each of the N output channels, the inventive system
uses, as a base channel, one of the N transmitted channels
or a linear combination thereof as the base channel for
each of the N output channels.
Therefore, Fig. 14 shows a NtoM scheme, i. e. a scheme, in
which N original channels are downmixed to two downmix
channels. In the example of Fig. 14, N is equal to 5 while
M is equal to 2. In particular, for the front left channel
reconstruction, the transmitted left channel Lc is used.
Analogously, for the front right channel reconstruction,
the second transmitted channel Rc is used as the base chan-
nel. Additionally, an equal combination of Lc and Rc is
used as the base channel for reconstructing the center
channel. In accordance with an embodiment of the present
invention, correlation measures are additionally transmit-
ted from an encoder to a decoder. Therefore, for the left
surround channel, not only the transmitted left channel Lc
is used but the transmitted channel Lc + α1Rc such that the
base channel for reconstructing the left surround channel
is not fully coherent to the base channel for reconstruct-
ing the front left channel. Analogously, the same procedure
is performed for the right side (with respect to the as-
sumed listener position), in that the base channel for re-

constructing the right surround channel is different from
the base channel for reconstructing the front right chan-
nel, wherein the difference is dependent on the coherence
measure α2 which is preferable transmitted from an encoder
to a decoder as side information.
The inventive process, therefore, is unique in that for the
reproduction of preferable each output channel, a different
base channel is used, wherein the base channels are equal
to the transmitted channels or a linear combination
thereof. This linear combination can depend on the trans-
mitted base channels on varying degrees, wherein these de-
grees depend on coherence measures which depends on the
original multi-channel signal.
The process of obtaining the N base channels given the M
transmitted channels is called "upmixing". This upmixing
can be implemented by multiplying a vector with the trans-
mitted channels by a NxM matrix to generate N base chan-
nels. By doing so, linear combinations of transmitted sig-
nal channels are formed to produce the base signals for the
output channel signals. A specific example for upmixing is
shown in Fig. 14A, which is a 5 to 2-scheme applied for
generating a 5-channel surround output signal with a 2-
channel stereo transmission. Preferably, the base channel
for an additional subwoofer output channel is the same as
the center channel L+R. In a preferred embodiment of the
present invention, a time-varying and - optionally - fre-
quency-varying coherence measure is provided such that a
time-adaptive upmixing matrix, which is - optionally - also
frequency-selective is obtained.

In the following, reference is made to Fig. 14B showing a
background for the inventive encoder implementation illus-
trated in Fig. 1B. In this context, it is to be noted that
ICC and ICTD cues between left and right and left surround
and right surround are the same as in the transmitted ste-
reo signal. Thus, there is, in accordance with the present
invention, no need for using ICC and ICTD cues between left
and right and left surround and right surround for synthe-
sizing or reconstructing an output signal. Another reason
for not synthesizing ICC and ICTD cues between left and
right and left surround and right surround is the general
objective stating that the base channels have to be modi-
fied as little as possible to maintain maximum signal qual-
ity. Any signal modification potentially introduces arti-
facts or non-naturalness.
Therefore, only a level representation of the original
multi-channel signal which is obtained by providing the
ICLD cues is provided, while, in accordance with the pre-
sent invention, ICC and ICTD parameters are only calculated
and transmitted for channel pairs to one side of the as-
sumed listener position. This is illustrated by the dotted
line 144 for the left side and the dotted line 145 for the
right side in Fig. 14B. In contrast to ICC and ICTD, ICLD
synthesis is rather non-problematic with respect to arti-
facts and non-naturalness because it just involves scaling
of subband signals. Thus, ICLDs are synthesized as gener-
ally as in regular BCC, i.e., between a reference channel
and all other channels. More generally speaking, in a N 2 M
scheme, ICLDs are synthesized between channel pairs similar
to regular BCC. ICC and ICTD cues, however, are, in accor-
dance with the present invention, only synthesized between
channel pairs which are on the same side with respect to

the assumed listener position, i.e., for the channel pair
including the front left and the left surround channel or
the channel pair including the front right and the right
surround channel.
In case of 7-channel or higher surround systems, in which
there are three channels on the left side and three chan-
nels on the right side, the same scheme can be applied,
wherein only for possible channel pairs on the left side or
the right side, coherence parameters are transmitted for
providing different base channels for the reconstruction of
the different output channels on one side of the assumed
listener position. The inventive NtoM encoder as shown in
Fig. 1A and Fig. 1B is, therefore, unique in that the input
signals are downmixed not into one single channel but into
M channels, and that ICTD and ICC cues are estimated and
transmitted only between the channel pairs for which this
is necessary.
In a 5-channel surround system, the situation is shown in
Fig. 14B from which it becomes clear that at least one co-
herence measure between left and left surround has to be
transmitted. This coherence measure can also be used for
providing decorrelation between right and right surround.
This is a low side information implementation. In case one
has more available channel capacity, one can also generate
and transmit a separate coherence measure between the right
and the right surround channel such that, in an inventive
decoder, also different degrees of decorrelation on the
left side and on the right side can be obtained.
Fig. 2A shows an illustration of an inventive decoder act-
ing as an apparatus for inverse processing input data re-

ceived at an input data port 22. The data received at the
input data port 22 is the same data as output at the output
data port 20 in Fig. 1A. Alternatively, when the data are
not transmitted via a wired channel but via a wireless
channel, the data received at data input port 22 are data
derived from the original data produced by the encoder.
The decoder input data are input into a data stream reader
24 for reading the input data to finally obtain the channel
side information 26 and the left downmix channel 28 and the
right downmix channel 30. In case the input data includes
encoded versions of the downmix channels, which corresponds
to the case, in which the audio encoder 16 in Fig. 1A is
present, the data stream reader 24 also includes an audio
decoder, which is adapted to the audio encoder used for en-
coding the downmix channels. In this case, the audio de-
coder, which is part of the data stream reader 24, is op-
erative to generate the first downmix channel Lc and the
second downmix channel Rc, or, stated more exactly, a de-
coded version of those channels. For ease of description, a
distinction between signals and decoded versions thereof is
only made where explicitly stated.
The channel side information 26 and the left and right
downmix channels 28 and 30 output by the data stream reader
24 are fed into a multi-channel reconstructor 32 for pro-
viding a reconstructed version 34 of the original audio
signals, which can be played by means of a multi-channel
player 36. In case the multi-channel reconstructor is op-
erative in the frequency domain, the multi-channel player
36 will receive frequency domain input data, which have to
be in a certain way decoded such as converted into the time

domain before playing them. To this end, the multi-channel
player 36 may also include decoding facilities.
It is to be noted here that a lower scale decoder will only
have the data stream reader 24, which only outputs the left
and right downmix channels 28 and 30 to a stereo output 38.
An enhanced inventive decoder will, however, extract the
channel side information 26 and use these side information
and the downmix channels 28 and 30 for reconstructing re-
constructed versions 34 of the original channels using the
multi-channel reconstructor 32.
Fig. 2B shows an inventive implementation of the multi-
channel reconstructor 32 of Fig. 2A. Therefore, Fig. 2B
shows an apparatus for constructing a multi-channel output
signal using an input signal and parametric side informa-
tion, the input signal including a first input channel and
a second input channel derived from an original multi-
channel signal, and the parametric side information de-
scribing interrelations between channels of the multi-
channel original signal. The inventive apparatus shown in
Fig. 2B includes means 320 for providing a coherence meas-
ure depending on a first original channel and a second
original channel, the first original channel and the second
original channel being included in the original multi-
channel signal. In case the coherence measure is included
in the parametric side information, the parametric side in-
formation is input into means 320 as illustrated in Fig.
2B. The coherence measure provided by means 320 is input
into means 322 for determining base channels. In particu-
lar, the means 322 is operative for determining a first
base channel by selecting one of the first and the second
input channels or a predetermined combination of the first

and the second input channels. Means 322 is further opera-
tive to determine a second base channel using the coherence
measure such that the second base channel is different from
the first base channel because of the coherence measure. In
the example shown in Fig. 2B, which is related to the 5-
channel surround system, the first input channel is the
left compatible stereo channel Lc; and the second input
channel is the right compatible stereo channel Rc. The
means 322 is operative to determine the base channels which
have already been described in connection with Fig. 14A.
Thus, at the output of means 322, a separate base channel
for each of the to be reconstructed output channels is ob-
tained, wherein, preferably, the base channels output by
means 322 are all different from each other, i.e., have a
coherence measure between themselves, which is different
for each pair.
The base channels output by means 322 and parametric side
information such as ICLD, ICTD or intensity stereo informa-
tion are input into means 324 for synthesizing the first
output channel such as L using the parametric side informa-
tion and the first base channel to obtain a first synthe-
sized output channel L, which is a reproduced version of
the corresponding first original channel, and for synthe-
sizing a second output channel such as Ls using the para-
metric side information and the second base channel, the
second output channel being a reproduced version of the
second original channel. In addition, means 324 for synthe-
sizing is operative to reproduce the right channel R and
the right surround channel Rs using another pair of base
channels, wherein the base channels in this other pair are
different from each other because of the coherence measure

or because of an additional coherence measure which has
been derived for the right/right surround channel pair.
A more detailed implementation of the inventive decoder is
shown in Fig. 2C. It can be seen that in the preferred em-
bodiment which is shown in Fig. 2C, the general structure
is similar to the structure which has already been de-
scribed in connection with Fig. 12 for a state of the art
prior art BCC decoder. Contrary to Fig. 12, the inventive
scheme shown in Fig. 2C includes two audio filter banks,
i.e., one filter bank for each input signal. Naturally, a
single filter bank is also sufficient. In this case, a con-
trol is required which inputs into the single filter bank
the input signals in a sequential order. The filter banks
are illustrated by blocks 319a and 319b. The functionality
of elements 320 and 322 - which are illustrated in Fig. 2B
- is included in an upmixing block 323 in Fig. 2C.
At the output of the upmixing block 323, base channels,
which are different from each other, are obtained. This is
in contrast to Fig. 12, in which the base channels on node
130 are identical to each other. The synthesizing means 324
shown in Fig. 2B includes preferably a delay stage 324a, a
level modification stage 324b and, in some cases, a proc-
essing stage for performing additional processing tasks
324c as well as a respective number of inverse audio filter
banks 324d. In one embodiment, the functionality of ele-
ments 324a, 324b, 324c and 324d can be the same as in the
prior art device described in connection with Fig. 12.
Fig. 2D shows a more detailed example of Fig. 2C for a 5-
channel surround set up, in which two input channels y1 and
y2 are input and five constructed output channels are ob-

tained as shown in Fig. 2D. In contrast to Fig. 2C, a more
detailed design of the upmixing block 323 is given. In par-
ticular, a summation device 330 for providing the base
channels for reconstructing a center output channel is
shown. Additionally, two blocks 331, 332 titled "W" are
shown in Fig. 2D. These blocks perform the weighted combi-
nation of the two input channels based on the coherence
measure K which is input at a coherence measure input 334.
Preferably, the weighting block 331 or 332 also performs
respective post processing operations for the base channels
such as smoothing in time and frequency as will be outlined
below. Thus, Fig. 2C is a general case of Fig. 2D, wherein
Fig. 2C illustrates how the N output channels are gener-
ated, given the decoder's M input channels. The transmitted
signals are transformed to a sub band domain.
The process of computing the base channels for each output
channel is denoted upmixing, because each base channel is
preferably a linear combination of the transmitted chan-
nels. The upmixing can be performed in the time domain or
in the sub band or frequency domain.
For computing each base channel, a certain processing can
be applied to reduce cancellation/amplification effects
when the transmitted channels are out-of-phase or in-phase.
ICTD are synthesized by imposing delays on the sub band
signals and ICLD are synthesized by scaling the sub band
signals. Different techniques can be used for synthesizing
ICC such as manipulating the weighting factors or the time
delays by means of a random number sequence. It is, how-
ever, to be noted here that preferably, no coher-
ence/correlation processing between output channels except
the inventive determination of the different base channels

for each output channel is performed. Therefore, a pre-
ferred inventive device processes ICC cues received from an
encoder for constructing the base channels and ICTD and
ICLD cues received from an encoder for manipulating the al-
ready constructed base cha nnel. Thus, ICC cues or — more
generally speaking - coherence measures are not used for
manipulating a base channel but are used for constructing
the base channel which is manipulated later on.
In the specific example shown in Fig. 2D, a 5-channel sur-
round signal is decoded from a 2-channel stereo transmis-
sion. A transmitted 2-channel stereo signal is converted to
a sub band domain. Then, upmixing is applied to generate
five preferable different base channels. ICTD cues are only
synthesized between left and left surround, and right and
right surround by applying delays di (k) as has been dis-
cussed in connection with Fig. 14B. Also, the coherence
measures are used for constructing the base channels
(blocks 331 and 332) in Fig. 2D rather than for doing any
post processing in block 324c.
Inventively, the ICC and ICTD cues between left and right
and left surround and right surround are maintained as in
the transmitted stereo signal. Therefore, a single ICC cue
and a single ICTD cue parameter will be sufficient and
will, therefore, be transmitted from an encoder to a de-
coder.
In another embodiment, ICC cues and ICTD cues for both
sides can be calculated in an encoder. These two values can
be transmitted from an encoder to a decoder. Alternatively,
the encoder can compute a resulting ICC or ICTD cue by in-
putting the cues for both sides into a mathematical func-

tion such as an averaging function etc for deriving the re-
sulting value from the two coherence measures.
In the following, reference is made to Fig. 15A and 15B to
show a low-complexity implementation of the inventive con-
cept. While a high-complexity implementation requires an
encoder-side determination of the coherence measure at
least between a channel pair on one side of the assumed
listener position, and transmitting of this coherence meas-
ure preferably in a quantized and entropy-encoded form, the
low-complexity version does not require any coherence meas-
ure determination on the encoder-side and any transmission
from the encoded to the decoder of such information. In or-
der to, nevertheless, obtain a good subjective quality of
the reconstructed multi channel output signal, a predeter-
mined coherence measure or, stated in other words, prede-
termined weighting factors for determining a weighted com-
bination of the transmitted input channels using such a
predetermined weighting factor is provided by the means 324
in Fig. 2D. There exist several possibilities to reduce co-
herence in base channels for the reconstruction of output
channels. Without the inventive measure, the respective
output channels would be, in a base line implementation, in
which no ICC and ICTD are encoded and transmitted, fully
coherent. Therefore, any use of any predetermined coherence
measure will reduce coherence in reconstructed output sig-
nals such that the reproduced output signals are better ap-
proximations of the corresponding original channels.
To therefore prevent that base channels are fully coherent,
the upmixing is done as shown for example in Fig. 15A as
one alternative or Fig. 15B as another alternative. The
five base channels are computed such that none of them are

fully coherent, if the transmitted stereo signal is also
not fully coherent. This results in that an inter-channel
coherence between the left channel and the left surround
channel or between the right channel and the right surround
channel is automatically reduced, when the inter-channel
coherence between the left channel and the right channel is
reduced. For example, for an audio signal which is inde-
pendent between all channels such as an applause signal,
such upmixing has the advantage that a certain independence
between left and left surround and right and right surround
is generated without a need for synthesizing (and encoding)
inter-channel coherence explicitly. Of course, this second
version of upmixing can be combined with a scheme which
still synthesizes ICC and ICTD.
Fig. 15A shows an upmixing optimized for front left and
front right, in which most independence is maintained be-
tween the front left and the front right.
Fig. 15B shows another example, in which front left and
front right on the one hand and left surround and right
surround on the other hand are treated in the same way in
that the degree of independence of the front and rear chan-
nels is the same. This can be seen in Fig. 15B by the fact
that an angle between front left/right is the same as the
angle between left surround/right.
In accordance with the preferred embodiment of the present
invention, dynamic upmixing instead of a static selection,
is used. To this end, the invention also relates to an en-
hanced algorithm which is able to dynamically adapt the up-
mixing matrix in order to optimize a dynamic performance.
In the example illustrated below, the upmixing matrix can

be chosen for the back channels such that optimum reproduc-
tion of front-rear coherence becomes possible. The inven-
tive algorithm comprises the following steps:
For the front channels, a simply assignment of base chan-
nels is used, as the one described in Fig. 14A or 15A. By
this simple choice, coherence of the channels along the
left/right axis is preserved.
In the encoder, the front-back coherence values such as ICC
cues between left/left surround and preferably between
right/right surround pairs are measured.
In the decoder, the base channels for the left rear and
right rear channels are determined by forming linear combi-
nations of the transmitted channel signals, i.e., a trans-
mitted left channel and a transmitted right channel. Spe-
cifically, upmixing coefficients are determined such that
the actual coherence between left and left surround and
right and right surround achieves the values measured in
the encoder. For practical purposes, this can be achieved
when the transmitted channel signals exhibit sufficient
decorrelations, which is normally the case in usual 5-
channel scenarios.
In the preferred embodiment of dynamic upmixing, an example
of an implementation which is regarded as the best mode of
carrying out the present invention, will be given with re-
spect to Fig. 2E as to an encoder implementation and Fig.
2F and Fig. 2G with respect to a decoder implementation.
Fig. 2E shows one example for measuring front/back coher-
ence values (ICC values) between the left and the left sur-
round channel or between the right and the right surround

channel, i.e., between a channel pair located at one side
with respect to an assumed listener position.
The equation shown in the box in Fig. 2E gives a coherence
measure cc between the first channel x and the second chan-
nel y. In one case, the first channel x is the left chan-
nel, while the second channel y is the left surround chan-
nel. In another case, the first channel x is the right
channel, while the second channel y is the right surround
channel. Xi stands for a sample of the respective channel x
at the time instance i, while yi stands for a sample at a
time instance of the other original channel y. It is to be
noted here that the coherence measure can be calculated
completely in the time domain. In this case, the summation
index i runs from a lower border to an upper border,
wherein the other border normally is the same as the number
of samples in one frame in case of a frame-wise processing.
Alternatively, coherence measures can also be calculated
between band pass signals, i.e., signals having reduced
band widths with respect to the original audio signal. In
the latter case, the coherence measure is not only time-
dependent but also frequency-dependent. The resulting
front/back ICC cues, i.e., CCi for the left front/back co-
herence and CCr for the right front/back coherence are
transmitted to a decoder as parametric side information
preferably in quantized and encoded form.
In the following, reference will be made to Fig. 2F for
showing a preferred decoder upmixing scheme. In the illus-
trated case, the transmitted left channel is kept as the
base channel for the left output channel. In order to de-
rive the base channel for the left rear output channel, a

linear combination between the left (1) and the right (r)
transmitted channel, i.e., 1 + αr, is determined. The
weighting factor a is determined such that the cross-
correlation between 1 and 1 + ar is equal to the transmit-
ted desired value CC1 for the left side and CCr for the
right side or generally the coherence measure k.
The calculation of the appropriate a value is described in
Fig. 2F. In particular, a normalized cross-correlation of
two signals 1 and r is defined as shown in the equation in
the block of Fig. 2E.
Given two transmitted signals 1 and r, the weighting factor
a has to be determined such that the normalized cross-
correlation of the signal 1 and 1 + ar is equal to a de-
sired value k, i.e., the coherence measure. This measure is
defined between -1 and +1.
Using the definition of the cross-correlation for the two
channels, one obtains the equation given in Fig. 2F for the
value k. By using several abbreviations which are given in
the bottom of Fig. 2F, the condition for k can be rewritten
as a quadratic equation, the solution of which gives the
weighting factor a.
It can be shown that the equation always has real-valued
solutions, i.e., that the discriminant is guaranteed to be
non-negative.
Depending on the basic cross-correlation of the signal 1
and r, and on the desired cross-correlation k, one of both
delivered solutions may in fact lead to the negative of the

desired cross-correlation value and is, therefore, dis-
carded for all further calculation.
After calculating the base channel signal as a linear com-
bination of the 1 signal and the r signal, the resulting
signal is normalized (re-scaled) to the original signal en-
ergy of the transmitted 1 or r channel signal.
Similarly, the base channel signal for the right output
channel can be derived by swapping the role of the left and
right channels, i.e., considering the cross-correlation be-
tween r and r + α1.
In practice, it is preferred to smooth the results of the
calculation process for the a value over time and fre-
quency in order to obtain maximum signal quality. Also
front/back correlation measurements other than left/left
rear and right/right rear can be used to further maximize
signal quality.
Subsequently, a step-by-step description of the functional-
ity performed by the multi-channel reconstructor 32 from
Fig. 2A will be given, referring to Fig. 2G.
Preferably, a weighting factor a is calculated (200) based
on a dynamic coherence measure provided from an encoder to
a decoder or based on a static provision of a coherence
measure as described in connection with Fig. 15A and Fig.
15B. Then, the weighting factor is smoothed over time
and/or frequency (step 202) to obtain a smoothed weighting
factor αs. Then, a base channel b is calculated to be for
example 1 + αsr (step 204) . The base channel b is then

used, together with other base channels, to calculate raw
output signals.
As it becomes clear from box 206, the level representation
ICLD as well as the delay representation ICTD are required
for calculating raw output signals. Then, the raw output
signals are scaled to have the same energy as a sum of the
individual energies of the left and right input channels.
Stated in other words, the raw output signals are scaled by
means of a scaling factor such that a sum of the individual
energies of the scaled raw output signals is the same as
the sum of the individual energies of the transmitted left
and right input channels.
Alternatively, one could also calculated the sum of the
left and right transmitted channels and to use the energy
of the resulting signal. Additionally, one could also cal-
culate a sum signal by sample wise summing the raw output
signals and to use the energy of the resulting signal for
scaling purposes.
Then, at an output of box 208, the reconstructed output
channels are obtained, which are unique in that none of the
reconstructed output channels is fully coherent to another
of the reconstructed output channels such that a maximum
quality of the reproduced output signal is obtained.
To summarize, the inventive concept is advantageous in that
an arbitrary number of transmitted channels (M) and an ar-
bitrary number of output channels (N) can be used.

Additionally, the conversion between the transmitted chan-
nels and the base channels for the output channels is done
via preferably dynamic upmixing.
In an important embodiment, upmixing consists of a multi-
plication by an upmixing matrix, i.e., forming linear com-
binations of the transmitted channels, wherein front chan-
nels are preferably synthesized by using the corresponding
transmitted base channels as base channels, while the rear
channels consist of linear combination of the transmitted
channels, the degree of a linear combination depending on a
coherence measure.
Additionally, this upmixing process is preferably performed
signal adaptive in a time-varying fashion. Specifically,
the upmixing process preferably depends on a side informa-
tion transmitted from a BCC encoder such as inter-channel
coherence cues for a front/rear coherence.
Given the base channel for each output channel, a process-
ing similar to a regular binaural cue coding is applied to
synthesize spatial cues, i.e., applying scalings and delays
in subbands and applying techniques to reduce coherence be-
tween channels, wherein ICC cues are additionally, or al-
ternatively, used for constructing respective base channels
to obtain optimal reproduction of front/rear coherence.
Fig. 3A shows an embodiment of the inventive calculator 14
for calculating the channel side information, which an au-
dio encoder on the one hand and the channel side informa-
tion calculator on the other hand operate on the same spec-
tral representation of multi-channel signal. Fig. 1, how-
ever, shows the other alternative, in which the audio en-

coder on the one hand and the channel side information cal-
culator on the other hand operate on different spectral
representations of the multi-channel signal. When computing
resources are not as important as audio guality, the Fig.
1A alternative is preferred, since filterbanks individually
optimized for audio encoding and side information calcula-
tion can be used. When, however, computing resources are an
issue, the Fig. 3A alternative is preferred, since this al-
ternative requires less computing power because of a shared
utilization of elements.
The device shown in Fig. 3A is operative for receiving two
channels A, B. The device shown in Fig. 3A is operative to
calculate a side information for channel B such that using
this channel side information for the selected original
channel B, a reconstructed version of channel B can be cal-
culated from the channel signal A. Additionally, the device
shown in Fig. 3A is operative to form frequency domain
channel side information, such as parameters for weighting
(by multiplying or time processing as in BCC coding e. g.)
spectral values or subband samples. To this end, the inven-
tive calculator includes windowing and time/frequency con-
version means 140a to obtain a frequency representation of
channel A at an output 14 0b or a frequency domain represen-
tation of channel B at an output 140c.
In the preferred embodiment, the side information determi-
nation (by means of the side information determination
means 140f) is performed using quantized spectral values.
Then, a quantizer 140d is also present which preferably is
controlled using a psychoacoustic model having a psycho-
acoustic model control input 140e. Nevertheless, a quan-
tizer is not required, when the side information determina-

tion means 140c uses a non-quantized representation of the
channel A for determining the channel side information for
channel B.
In case the channel side information for channel B are cal-
culated by means of a frequency domain representation of
the channel A and the frequency domain representation of
the channel B, the windowing and time/frequency conversion
means 140a can be the same as used in a filterbank-based
audio encoder. In this case, when AAC (ISO/IEC 13818-3) is
considered, means 140a is implemented as an MDCT filter
bank (MDCT = modified discrete cosine transform) with 50%
overlap-and-add functionality.
In such a case, the quantizer 140d is an iterative quan-
tizer such as used when mp3 or AAC encoded audio signals
are generated. The frequency domain representation of chan-
nel A, which is preferably already quantized can then be
directly used for entropy encoding using an entropy encoder
140g, which may be a Huffman based encoder or an entropy
encoder implementing arithmetic encoding.
When compared to Fig. 1, the output of the device in Fig.
3A is the side information such as li for one original
channel (corresponding to the side information for B at the
output of device 140f). The entropy encoded bitstream for
channel A corresponds to e.g. the encoded left downmix
channel Lc' at the output of block 16 in Fig. 1. From Fig.
3A it becomes clear that element 14 (Fig. 1), i.e., the
calculator for calculating the channel side information and
the audio encoder 16 (Fig. 1) can be implemented as sepa-
rate means or can be implemented as a shared version such
that both devices share several elements such as the MDCT

filter bank 140a, the quantizer 140e and the entropy en-
coder 140g. Naturally, in case one needs a different trans-
form etc. for determining the channel side information,
then the encoder 16 and the calculator 14 (Fig. 1) will be
implemented in different devices such that both elements do
not share the filter bank etc.
Generally, the actual determinator for calculating the side
information (or generally stated the calculator 14) may be
implemented as a joint stereo module as shown in Fig.3B,
which operates in accordance with any of the joint stereo
techniques such as intensity stereo coding or binaural cue
coding.
In contrast to such prior art intensity stereo encoders,
the inventive determination means 140f does not have to
calculate the combined channel. The "combined channel" or
carrier channel, as one can say, already exists and is the
left compatible downmix channel Lc or the right compatible
downmix channel Rc or a combined version of these downmix
channels such as Lc + Rc. Therefore, the inventive device
140f only has to calculate the scaling information for
scaling the respective downmix channel such that the en-
ergy/time envelope of the respective selected original
channel is obtained, when the downmix channel is weighted
using the scaling information or, as one can say, the in-
tensity directional information.
Therefore, the joint stereo module 140f in Fig 3B is illus-
trated such that it receives, as an input, the "combined"
channel A, which is the first or second downmix channel or
a combination of the downmix channels, and the original se-
lected channel. This module, naturally, outputs the "com-

bined" channel A and the joint stereo parameters as channel
side information such that, using the combined channel A
and the joint stereo parameters, an approximation of the
original selected channel B can be calculated.
Alternatively, the joint stereo module 140f can be imple-
mented for performing binaural cue coding.
In the case of BCC, the joint stereo module 140f is opera-
tive to output the channel side information such that the
channel side information are quantized and encoded ICLD or
ICTD parameters, wherein the selected original channel
serves as the actual to be processed channel, while the re-
spective downmix channel used for calculating the side in-
formation, such as the first, the second or a combination
of the first and second downmix channels is used as the
reference channel in the sense of the BCC coding/decoding
technique.
Referring to Fig. 4, a simple energy-directed implementa-
tion of element 140f is given. This device includes a fre-
quency band selector 44 selecting a frequency band from
channel A and a corresponding frequency band of channel B.
Then, in both frequency bands, an energy is calculated by
means of an energy calculator 42 for each branch. The de-
tailed implementation of the energy calculator 42 will de-
pend on whether the output signal from block 40 is a sub-
band signal or are frequency coefficients. In other imple-
mentations, where scale factors for scale factor bands are
calculated, one can already use scale factors of the first
and second channel A, B as energy values EA and EB or at
least as estimates of the energy. In a gain factor calcu-
lating device 44, a gain factor gB for the selected fre-

quency band is determined based on a certain rule such as
the gain determining rule illustrated in block 44 in Fig.
4. Here, the gain factor gB can directly be used for
weighting time domain samples or frequency coefficients
- such as will be described later in Fig. 5. To this end, the
gain factor gB, which is valid for the selected frequency
band is used as the channel side information for channel B
as the selected original channel. This selected original
channel B will not be transmitted to decoder but will be
' represented by the parametric channel side information as
calculated by the calculator 14 in Fig. 1.
It is to be noted here that it is not necessary to transmit
gain values as channel side information. It is also suffi-
cient to transmit frequency dependent values related to the
absolute energy of the selected original channel. Then, the
decoder has to calculate the actual energy of the downmix
channel and the gain factor based on the downmix channel
energy and the transmitted energy for channel B.
Fig. 5 shows a possible implementation of a decoder set up
in connection with a transform-based perceptual audio en-
coder. Compared to Fig. 2, the functionalities of the en-
tropy decoder and inverse quantizer 50 (Fig. 5) will be in-
cluded in block 24 of Fig. 2. The functionality of the fre-
quency/time converting elements 52a, 52b (Fig. 5) will,
however, be implemented in item 36 of Fig. 2. Element 50 in
Fig. 5 receives an encoded version of the first or the sec-
ond downmix signal Lc' or Re'. At the output of element 50,
an at least partly decoded version of the first and the
second downmix channel is present which is subsequently
called channel A. Channel A is input into a frequency band
selector 54 for selecting a certain frequency band from

channel A. This selected frequency band is weighted using a
multiplier 56. The multiplier 56 receives, for multiplying,
a certain gain factor gB, which is assigned to the selected
frequency band selected by the frequency band selector 54,
which corresponds to the frequency band selector 40 in Fig.
4 at the encoder side. At the input of the frequency time
converter 52a, there exists, together with other bands, a
frequency domain representation of channel A. At the output
of multiplier 56 and, in particular, at the input of fre-
quency/time conversion means 52b there will be a recon-
structed frequency domain representation of channel B.
Therefore, at the output of element 52a, there will be a
time domain representation for channel A, while, at the
output of element 52b, there will be a time domain repre-
sentation of reconstructed channel B.
It is to be noted here that, depending on the certain im-
plementation, the decoded downmix channel Lc or Rc is not
played back in a multi-channel enhanced decoder. In such a
multi-channel enhanced decoder, the decoded downmix chan-
nels are only used for reconstructing the original chan-
nels. The decoded downmix channels are only replayed in
lower scale stereo-only decoders.
To this end, reference is made to Fig. 9, which shows the
preferred implementation of the present invention in a sur-
round/mp3 environment. An mp3 enhanced surround bitstream
is input into a standard mp3 decoder 24, which outputs de-
coded versions of the original downmix channels. These
downmix channels can then be directly replayed by means of
a low level decoder. Alternatively, these two channels are
input into the advanced joint stereo decoding device 32
which also receives the multi-channel extension data, which

are preferably input into the ancillary data field in a mp3
compliant bitstream.
Subsequently, reference is made to Fig. 7 showing the
grouping of the selected original channel and the respec-
tive downmix channel or combined downmix channel. In this
regard, the right column of the table in Fig. 7 corresponds
to channel A in Fig. 3A, 3B, 4 and 5, while the column in
the middle corresponds to channel B in these figures. In
the left column in Fig. 7, the respective channel side in-
formation is explicitly stated. In accordance with the Fig.
7 table, the channel side information li for the original
left channel L is calculated using the left downmix channel
Lc. The left surround channel side information lsi is de-
termined by means of the original selected left surround
channel Ls and the left downmix channel Lc is the carrier.
The right channel side information ri for the original
right channel R are determined using the right downmix
channel Rc. Additionally, the channel side information for
the right surround channel Rs are determined using the
right downmix channel Rc as the carrier. Finally, the chan-
nel side information ci for the center channel C are deter-
mined using the combined downmix channel, which is obtained
by means of a combination of the first and the second down-
mix channel, which can be easily calculated in both an en-
coder and a decoder and which does not require any extra
bits for transmission.
Naturally, one could also calculate the channel side infor-
mation for the left channel e. g. based on a combined down-
mix channel or even a downmix channel, which is obtained by
a weighted addition of the first and second downmix chan-
nels such as 0.7 Lc and 0.3 Rc, as long as the weighting

parameters are known to a decoder or transmitted accord-
ingly. For most applications, however, it will be preferred
to only derive channel side information for the center
channel from the combined downmix channel, i.e., from a
combination of the first and second downmix channels.
To show the bit saving potential of the present invention,
the following typical example is given. In case of a five
channel audio signal, a normal encoder needs a bit rate of
64 kbit/s for each channel amounting to an overall bit rate
of 320 kbit/s for the five channel signal. The left and
right stereo signals require a bit rate of 128 kbit/s.
Channels side information for one channel are between 1.5
and 2 kbit/s. Thus, even in a case, in which channel side
information for each of the five channels are transmitted,
this additional data add up to only 7.5 to 10 kbit/s. Thus,
the inventive concept allows transmission of a five channel
audio signal using a bit rate of 138 kbit/s (compared to
320 (!) kbit/s) with good quality, since the decoder does
not use the problematic dematrixing operation. Probably
even more important is the fact that the inventive concept
is fully backward compatible, since each of the existing
mp3 players is able to replay the first downmix channel and
the second downmix channel to produce a conventional stereo
output.
Depending on the application environment, the inventive
methods for constructing or generating can be implemented
in hardware or in software. The implementation can be a
digital storage medium such as a disk or a CD having elec-
tronically readable control signals, which can cooperate
with a programmable computer system such that the inventive
methods are carried out. Generally stated, the invention

therefore, also relates to a computer program product hav-
ing a program code stored on a machine-readable carrier,
the program code being adapted for performing the inventive
methods, when the computer program product runs on a com-
puter. In other words, the invention, therefore, also re-
lates to a computer program having a program code for per-
forming the methods, when the computer program runs on a
computer.

WE CLAIM :
1. An apparatus for constructing a multi-channel output signal (34) using an
input signal (28, 30) and parametric side information (26), the input signal
comprising a first input channel (Lc, 28) and a second input channel (Rc,
30) derived from an original multi-channel signal (10), the original multi-
channel signal (10) having a plurality of channels, the plurality of channels
comprising at least two original channels, which are defined as being
located at one side of an assumed listener position, wherein a first original
channel is a first one of the at least two original channels, and wherein a
second original channel is a second one of the at least two original
channels, and the parametric side information (26) describing
interrelations between original channels of the multi-channel original
signal, comprising:
means (322) for determining a first base channel by selecting one of the
first and the second input channels (28, 30) or a combination of the first
and the second input channels (28, 30), and for determining a second
base channel by selecting the other of the first and the second input
channels (28, 30)or a different combination of the first and the second
input channels(28, 30), such that the second base channel is different
from the first base channel; and
means (324) for synthesizing a first output channel using the parametric
side information (26) and the first base channel to obtain a first
synthesized output channel (L) which is a reproduced version of the first
original channel which is located at the one side of the assumed listener
position, and for synthesizing a second output channel (R) using the
parametric side information (26) and the second base channel, the second
output channel (R) being a reproduced version of the second original
channel which is located at the same side of the assumed listener
position.

2. The apparatus as claimed in claim 1, comprising:
means (320) for providing a coherence measure, the coherence measure
depending on a coherence between a first original channel and a second
original channel, the first and the second original channels being
comprised of in an original multi-channel signal;
wherein the means (322) for determining is operative to determine the
first and the second base channels different from each other based on the
coherence measure.
3. The apparatus as claimed in claim 1, wherein the at least two original
channels comprise a left original channel (L) and a left surround original
channel (Ls) or a right original channel (R) and a right surround original
channel (Rs).
4. The apparatus as claimed in claim 1, wherein a combination of the first
and the second input channels determined to be the second base channel
is such that one of the two input channels contributes to the second base
channel more than the other input channel.
5. The apparatus as claimed in claim 2, wherein the coherence measure is
time-varying such that the means (320) for determining is operative to
determine the second base channel as a combination of the first input
channel and the second input channel, the combination being variable
over time.
6. The apparatus as claimed in claim 2, wherein parametric side information
(26) comprises the coherence measure, the coherence measure being
determined using the first original channel (L) and the second original
channel (R), and wherein the means (320) for providing is operative to
extract the coherence measure from the parametric side information (26).

7. The apparatus as claimed in claim 6, wherein the input signal has a
sequence of frames and the parametric side information (26) comprises a
sequence of parameters comprising the coherence measure, the
parameters being associated with the frames.
8. The apparatus as claimed in claim 1, wherein the original signal
comprises a center channel (C), and in which the means (322) for
determining is further operative to calculate a third base channel using the
first input channel (L) and the second input channel (R) in equal portions.
9. The apparatus as claimed in claim 1, wherein the parametric side
information (26) are frequency dependent and the means (324) for
synthesizing are operative to perform a frequency-dependent synthesis.
10. The apparatus as claimed in claim 1, wherein the parametric side
information (26) comprise binaural cue coding (BCC) parameters
comprising inter-channel level difference parameters and inter-channel
time delay parameters, and wherein the means (324) for synthesizing is
operative to perform a BCC synthesis using a base channel determined by
the means (322) for determining when synthesizing an output channel.
11. The apparatus as claimed in claim 2, wherein the means (322) for
determining is operative to determine the first base channel as one of the
first and second input channels (L, R) and to determine the second base
channel as a weighted combination of the first and the second input
channels (L, R), a weighting factor depending on the coherence measure.
12. The apparatus as claimed in claim 11, wherein the means (322) for
determining is configured to determine the weighting factor based on the
following relationship:


wherein a is the weighting factor, and wherein A, B, C are
determined as follows,

wherein L, R, C are determined as follows,

and wherein k is the coherence measure, and wherein I is the first
input channel and r is the second input channel.
13. The apparatus as claimed in claim 11, wherein the coherence measure is
given for a frequency band, and in which the means (322) for determining
is operative to determine the second base channel for the frequency band.
14. The apparatus as claimed in claim 11, wherein the means (322) for
determining is configured to determine the coherence measure based on
the following relationship :

wherein cc(x,y) is the coherence measure between two original
channels x, y, wherein Xi is a sample at a time instance i of the first
original channel, and wherein yi is a sample at a time instance i of
the second original channel.
15. The apparatus as claimed in claim 1, wherein the means (322) for
determining is operative to scale the output channels using power
measures derived from the original channels, the power measures being
transmitted within the parametric side information (26).

16. The apparatus as claimed in claim 11, wherein the means (322) for
determining is operative to smooth the weighting factor over time and/or
frequency.
17. The apparatus as claimed in claim 1, wherein the parametric side
information (26) comprise level information representing an energy
distribution of the original channels in the original signal, and wherein the
means (324) for synthesizing is operative to scale the output channels
such that a sum of the energies of the output channels is equal to a sum
of the energies of the first input channel and the second input channel.
18. The apparatus as claimed in claim 17, wherein the means (324) for
synthesizing is operative to calculate raw output channels based on
determined base channels and the level information and to scale the raw
output channels such that a total energy of scaled raw output channels is
equal to a total energy of the first and the second input channels.
19. The apparatus as claimed in claim 1, wherein the input signal comprises a
left channel (Lc) and a right channel (Rc), and the original channel
comprises a front left channel (L), a left surround channel (Ls), a front
right channel (R) and a right surround channel (Rs), and in which the
means (322) for determining is operative to determine
the left channel (Lc) as the base channel for a synthesis of the front
left channel (L),
the right channel (Rc) as the base channel for a synthesis of the
front right channel (R),
a combination of the left channel (Lc) and the right channel (Rc) as
the base channel for the left surround channel (Ls) or the right
surround channel (Rs).

20. The apparatus as claimed in claim 1,
wherein the input signal comprises a left channel (Lc) and a right channel
(Rc) and the original signal comprises a front left channel (L), a left
surround channel (Ls), a front right channel (R) and a right surround
channel (Rs), and in which the means (322) for determining is operative to
determine
the left channel (Lc) as the base channel for a synthesis of the front
left channel,
the right channel (Rc) as the base channel for a synthesis of the
right surround channel, and
a combination of the first and the second input channels (Lc, Rc) as
the base channel for a synthesis of the front right channel or the
left surround channel.
21. A method of constructing a multi-channel output signal (34) using an
input signal (28, 30) and parametric side information (26), the input signal
comprising a first input channel (Lc, 28) and a second input channel (Rc,
30) derived from an original multi-channel signal (10), the original multi-
channel signal (10) having a plurality of channels, the plurality of channels
comprising at least two original channels, which are defined as being
located at one side of an assumed listener position, wherein a first original
channel is a first one of the at least two original channels, and wherein a
second original channel is a second one of the at least two original
channels, and the parametric side information (26) describing
interrelations between original channels of the multi-channel original
signal, comprising:

determining (322) a first base channel by selecting one of the first and the
second input channels (28, 30) or a combination of the first and the
second input channels(28, 30), and determining a second base channel by
selecting the other of the first and the second input channels (28, 30) or a
different combination of the first and the second input channels(28, 30),
such that the second base channel is different from the first base channel;
and
synthesizing (324) a first output channel using the parametric side
information (26) and the first base channel to obtain a first synthesized
output channel (L) which is a reproduced version of the first original
channel which is located at the one side of the assumed listener position,
and synthesizing a second output channel (R) using the parametric side
information (26) and the second base channel, the second output channel
(R) being a reproduced version of the second original channel which is
located at the same side of the assumed listener position.
22. An apparatus for generating a downmix signal (Lc, Rc) from a multi-
channel original signal (10), the downmix signal (Lc, Rc) having a number
of channels being smaller than a number of original channels, comprising:
means (12) for calculating a first downmix channel (Lc) and a second
downmix channel (Rc) using a downmix rule;
means (14) for calculating parametric level information representing an
energy distribution among the channels in the multi-channel original
signal;
means (142) for determining a coherence measure between two original
channels, the two original channels being located at one side of an
assumed listener position; and

means (18) for forming an output signal using the first and the second
downmix channels (Lc, Rc), the parametric level information and only at
least one coherence measure between two original channels (L, Ls)
located at the one side or a value derived from the at least one coherence
measure, but not using any coherence measure between channels (L, R)
located at different sides of the assumed listener position.
23. The apparatus as claimed in claim 22, comprising means (143) for
determining time delay information between two original channels (L, Ls)
located at one side of the assumed listener position; and
wherein the means (18) for forming is operative to only comprise time
level information between two original channels (L, Ls) located at one side
of the assumed listener position but not time level information between
two original channels (L, R) located at different sides of the assumed
listener position.
24. A method of generating a downmix signal (Lc, Rc) from a multi-channel
original signal (10), the downmix signal (Lc, Rc) having a number of
channels being smaller than a number of original channels, comprising:
calculating (12) a first downmix channel (Lc) and a second downmix
channel (Rc) using a downmix rule;
calculating (124) parametric level information representing an energy
distribution among the channels in the multi-channel original signal;
determining (142) a coherence measure between two original channels,
the two original channels being located at one side of an assumed listener
position; and

forming (18) an output signal using the first and the second downmix
channels(Lc, Rc), the parametric level information and only at least one
coherence measure between two original channels (L, Ls) located at the
one side or a value derived from the at least one coherence measure, but
not using any coherence measure between channels (L, R) located at
different sides of the assumed listener position.



ABSTRACT


TITLE "APPARATUS AND METHOD FOR CONSTRUCTING A
MULTI-CHANNEL OUTPUT SIGNAL OR FOR
GENERATING A DOWNMIX SIGNAL"
The invention relates to an apparatus for constructing a multi-channel output
signal (34) using an input signal (28, 30) and parametric side information (26),
the input signal comprising a first input channel (Lc, 28) and a second input
channel (Rc, 30) derived from an original multi-channel signal (10), the original
multi-channel signal (10) having a plurality of channels, the plurality of channels
comprising at least two original channels, which are defined as being located at
one side of an assumed listener position, wherein a first original channel is a first
one of the at least two original channels, and wherein a second original channel
is a second one of the at least two original channels, and the parametric side
information (26) describing interrelations between original channels of the multi-
channel original signal, comprising: means (322) for determining a first base
channel by selecting one of the first and the second input channels (28, 30) or a
combination of the first and the second input channels (28, 30), and for
determining a second base channel by selecting the other of the first and the
second input channels (28, 30)or a different combination of the first and the
second input channels(28, 30), such that the second base channel is different
from the first base channel; and means (324) for synthesizing a first output
channel using the parametric side information (26) and the first base channel to
obtain a first synthesized output channel (L) which is a reproduced version of the
first original channel which is located at the one side of the assumed listener
position, and for synthesizing a second output channel (R) using the parametric
side information (26) and the second base channel, the second output channel
(R) being a reproduced version of the second original channel which is located at
the same side of the assumed listener position.

Documents:

01966-kolnp-2006 abstract.pdf

01966-kolnp-2006 claims.pdf

01966-kolnp-2006 correspondence other.pdf

01966-kolnp-2006 description(compleit).pdf

01966-kolnp-2006 drawings.pdf

01966-kolnp-2006 form 1.pdf

01966-kolnp-2006 form 2.pdf

01966-kolnp-2006 form 3.pdf

01966-kolnp-2006 form 5.pdf

01966-kolnp-2006 international publication.pdf

01966-kolnp-2006 international serch authority report.pdf

01966-kolnp-2006 pct form.pdf

01966-kolnp-2006 priority document.pdf

01966-kolnp-2006-correspondence others-1.1.pdf

01966-kolnp-2006-form-26.pdf

1966-KOLNP-2006-(01-03-2012)-CORRESPONDENCE.pdf

1966-KOLNP-2006-(19-03-2013)-ABSTRACT.pdf

1966-KOLNP-2006-(19-03-2013)-CORRESPONDENCE.pdf

1966-KOLNP-2006-(19-03-2013)-FORM 1.pdf

1966-KOLNP-2006-(19-03-2013)-FORM 2.pdf

1966-KOLNP-2006-(23-09-2011)-CORRESPONDENCE.pdf

1966-KOLNP-2006-(23-09-2011)-OTHERS.pdf

1966-KOLNP-2006-ABSTRACT.pdf

1966-KOLNP-2006-AMANDED CLAIMS.pdf

1966-KOLNP-2006-ASSIGNMENT.pdf

1966-KOLNP-2006-CANCELLED PAGES.pdf

1966-KOLNP-2006-CORRESPONDENCE-1.1.pdf

1966-KOLNP-2006-CORRESPONDENCE.pdf

1966-KOLNP-2006-DESCRIPTION (COMPLETE).pdf

1966-KOLNP-2006-DRAWINGS.pdf

1966-KOLNP-2006-EXAMINATION REPORT.pdf

1966-KOLNP-2006-FORM 1.pdf

1966-KOLNP-2006-FORM 18.pdf

1966-KOLNP-2006-FORM 2.pdf

1966-KOLNP-2006-FORM 26.pdf

1966-KOLNP-2006-FORM 3.pdf

1966-KOLNP-2006-GRANTED-ABSTRACT.pdf

1966-KOLNP-2006-GRANTED-CLAIMS.pdf

1966-KOLNP-2006-GRANTED-DESCRIPTION (COMPLETE).pdf

1966-KOLNP-2006-GRANTED-DRAWINGS.pdf

1966-KOLNP-2006-GRANTED-FORM 1.pdf

1966-KOLNP-2006-GRANTED-FORM 2.pdf

1966-KOLNP-2006-GRANTED-FORM 3.pdf

1966-KOLNP-2006-GRANTED-FORM 5.pdf

1966-KOLNP-2006-GRANTED-SPECIFICATION-COMPLETE.pdf

1966-KOLNP-2006-INTERNATIONAL PUBLICATION.pdf

1966-KOLNP-2006-INTERNATIONAL SEARCH REPORT & OTHERS.pdf

1966-KOLNP-2006-OTHER PATENT DOCUMENT-1.1.pdf

1966-KOLNP-2006-OTHERS.pdf

1966-KOLNP-2006-PETITION UNDER RULE 137.pdf

1966-KOLNP-2006-REPLY TO EXAMINATION REPORT.pdf

abstract-01966-kolnp-2006.jpg


Patent Number 255688
Indian Patent Application Number 1966/KOLNP/2006
PG Journal Number 12/2013
Publication Date 22-Mar-2013
Grant Date 15-Mar-2013
Date of Filing 13-Jul-2006
Name of Patentee AGERE SYSTEMS INC.
Applicant Address 1110 AMERICAN PARKWAY NE, ALLENTOWN, PA 18109
Inventors:
# Inventor's Name Inventor's Address
1 HERRE, JURGEN HALLERSTR. 24 91054 BUCKENHOF
2 FALLER,CHRISTOF GUETRAIN 1, CH-8274 TRAGERWILEN
PCT International Classification Number G10L 19/00,H04S 3/02
PCT International Application Number PCT/EP2005/000408
PCT International Filing date 2005-01-17
PCT Conventions:
# PCT Application Number Date of Convention Priority Country
1 10/762,100 2004-01-20 U.S.A.