Efficient Use of Phase Information in Audio Encoding and
Decoding
Description
The present invention relates to audio encoding and audio
decoding, in particular to an encoding and decoding scheme,
selectively extracting and/or transmitting phase
information, when reconstruction of such information is
perceptually relevant.
Recent parametric multi-channel coding-schemes like
binaural cue coding (BCC), parametric stereo (PS) or MPEG
surround (MPS) use a compact parametric representation of
the humans auditory system's cues for spatial perception.
This allows for a rate efficient representation of an audio
signal having two or more audio channels. To this end, an
encoder performs a down-mix from M-input channels to N-
output channels and transmits the extracted cues together
with the down-mix signal. The cues are furthermore
quantized according to the principles of human perception,
that is, information which is not audible or
distinguishable by the human auditory system may be deleted
or coarsely quantized.
As the downmix-signal is a "generic" audio signal, the
bandwidth consumed by such an encoded representation of an
original audio signal may be further decreased by
compacting the down-mix signal or the channels of the
downmix signal using single channel audio compressors.
Various types of those single channel audio compressors
will be summarized as core coders within the following
paragraphs.
Typical cues used to describe the spatial interrelation
between two or more audio channels are interchannel level

differences (ILD) parametrizing level relations between
input channels, interchannel cross correlations/coherences
(ICC) parametrizing the statistical dependency between
input channels and interchannel time/phase differences (ITD
or IPD) parametrizing the time or phase difference between
similar signal segments of input channels.
To maintain a high perceptual, quality of the signals
represented by a down-mix and the previously described
cues, individual cues are normally calculated for different
frequency bands. That is, for a given time segment of the
signal, multiple cues parametrizing the same property are
transmitted, each cue-parameter representing a
predetermined frequency band of the signal.
The cues may be calculated time- and frequency dependent on
a scale close to the human frequency resolution. Whenever
multi-channel audio signals are represented, a
corresponding decoder performs an upmix from M to N
channels based on the transmitted spatial cues and the
downmix transmitted signals (the transmitted downmix
therefore often being called the carrier signal).
Generally, a resulting upmix channel may be described as a
level- and phase weighted version of the transmitted
downmix. The decorrelation derived while encoding the
signals may be synthesized by mixing and weighting the
transmitted downmix signal (the "dry" signal) with a
decorrelated signal (the "wet" signal) derived from the
downmix signal as indicated by the transmitted correlation
parameters (ICC). The upmixed channels then have a similar
correlation with respect to each other than the original
channels had. A decorrelated signal (i.e. a signal having a
cross correlation coefficient close to zero when cross-
correlated with the transmitted signal) may be produced by
feeding the downmix to a chain of filters, as for example,
all-pass filters and delay lines. However, further ways of
deriving a decorrelated signal may be used.

Apparently, in a particular implementation of the above
encoding/decoding scheme, a trade-off between the
transmitted bitrate (ideally being as low as possible) and
the achievable quality (ideally being as high as possible)
of the encoded signal, has to be performed.
It may, therefore, be decided to not transmit a full set of
spatial cues, but to omit transmission of one particular
parameter. This decision may additionally be influenced by
the selection of an appropriate upmix. An appropriate upmix
could, for example, reproduce a spatial cue not transmitted
on the average. That is, at least for a long-term segment
of the full bandwidth signal, the average spatial property
is preserved.
In particular, not all of the parametric multi-channel
schemes make use of interchannel time or interchannel phase
differences, thus avoiding the respective calculation and
synthesis. Schemes like MPEG surround rely on synthesis of
ILDs and ICCs only. The interchannel phase-differences are
implicitly approximated by the decorrelation synthesis,
which mixes two representations of the decorrelated signal
to the transmitted downmix signal, wherein the two
representations have a relative phase shift of 180°. A
transmission of IPDs is omitted, thus reducing the
necessary amount of parametric information, at the same
time, accepting a degradation in reproduction quality.
It, therefore, exists the need to provide for a better
reconstruction quality of a signal, without increasing the
required bitrate significantly.
One embodiment of the present invention achieves this goal
by using a phase estimator, which derives a phase
information indicating a phase relation between a first and
a second input audio signal, when a phase shift between the
input audio signals exceeds a predetermined threshold. An

associated output interface, which includes the spatial
parameters and a downmix signal into the encoded
representation of the input audio signals, does only
include the derived phase information, when the
transmission of phase information is, from a perceptional
point of view, necessary.
To do this, the determination of the phase information may
be performed continuously and only the decision, whether
the phase information is to be included or not, may be
taken based on the threshold. The threshold could, for
example, describe a maximum allowable phase shift, for
which additional phase information processing is
unnecessary to achieve an acceptable quality of the
reconstructed signal.
Alternatively, the phase shift between the input audio
signals may be derived independently from the actual
generation of the phase information, such that a decent
phase analysis to derive the phase information is only
taking place when the phase threshold is exceeded.
Alternatively, a spatial output mode decider may be
implemented, which receives the continuously generated
phase information, and which steers the output interface to
include the phase information only when a phase information
condition is met, that is, for example, when the phase
difference between the input signals exceeds a
predetermined threshold.
That is to say, the output interface predominantly includes
the ICC and ILD parameters as well as the downmix signal
into the encoded representation of the input audio signals
only. On occurrence of a signal having particular signal
characteristics, the determined phase information is
additionally included, such that the signal reconstructed
using the encoded representation may be reconstructed with
higher quality. However, this may be achieved by only a

minimum amount of additional transmitted information, since
the phase information is indeed only transmitted for those
signal parts, which are critical.
This allows, on the one hand, for a high quality
reconstruction and, on the other hand, for a low bitrate
implementation.
A further embodiment of the invention analyzes the signal
to derive a signal characterization information, the signal
characterization information distinguishing between input
audio signals having different signal types or
characteristics. This could, for example, be the different
characteristics of speech and of music signals. The phase
estimator may only be required, when the input audio
signals have a first characteristic, whereas, when the
input audio signals have a second characteristic, phase
estimation might be obsolete. The output interface does
therefore only include the phase information, when a signal
is encoded which requires phase synthesis in order to
provide an acceptable quality of the reconstructed signal.
Other spatial cues, such as, for example, the correlation
information (for example ICC parameters) are permanently
included in the encoded representation, since their
presence may be important for both signal types or signal
characteristics. This may, for example, also be true for
the interchannel level difference, which essentially
describes an energy relation between two reconstructed
channels.
In a further embodiment, the phase estimation may be
performed based on other spatial cues, such as on the
correlation ICC between the first and the second input
audio signal. This may become feasible when the
characterization information is present, which includes
some additional constraints on the signal characteristics.

Then, the ICC parameter may be used to extract, apart from
statistical information, also phase information.
According to a further embodiment, the phase information
may be included extremely bit efficient in that only one
phase-switch is implemented, signalling the application of
a phase shift of predetermined size. Nonetheless, the rough
reconstruction of the phase relation in reproduction may be
enough for certain signal types, as elaborated in more
detail below. In further embodiments the phase information
may be signalled in a much higher resolution (for example,
10 or 20 different phase shifts) or even as a continuous
parameter, giving possible relative phase angles between -
180° and +180°.
When the signal characteristic is known, phase information
may only be transmitted for a small number of frequency
bands, which may be much smaller than the number of
frequency bands used for the derivation of the ICC and/or
ILD parameters. When it is for example known that the audio
input signals have a speech characteristic, only one single
phase information may be necessary for the whole bandwith.
In a further embodiment, a single phase information may be
derived for a frequency range between, say, 100Hz and 5
kHz, since it is assumed that the signal energy of a
speaker is mainly distributed in this frequency range. A
common phase information parameter for the full bandwith
may, for example, be feasible when a phase shift exceeds
more than 90 degrees or more than 60 degrees.
When the signal characteristic is known, the phase
information may furthermore directly be derived from
already existent ICC parameters or correlation parameters,
by applying a threshold criterion to said parameters. For
example, when the ICC parameter is smaller than -0.1, it
may be concluded that this correlation parameter
corresponds to a fixed phase shift, as the speech

characteristic of the input audio signals constrains other
parameters as described in more detail below.
In a further embodiment of the present invention, an ICC
parameter (correlation parameter) derived from the signal
is furthermore modified or postprocessed, when the phase
information is included into the bitstream. This utilizes
the fact, that an ICC (correlation) parameter may actually
comprise information about two characteristics, namely
about the statistical dependence between the input audio
signals and about a phase shift between those signals. When
additional phase information is transmitted, the
correlation parameter may therefore be modified, such that
phase and correlation are, separately, considered as best
as possible while reconstructing the signal.
In a fully backwards compatible scenario, such correlation
modification may also be performed by an embodiment of an
inventive decoder. It could be activated, when the decoder
receives additional phase information.
To allow for such a perceptually superior reconstruction,
embodiments of inventive audio decoders may comprise an
additional signal processor operating on the intermediate
signals generated by an internal upmixer of the audio
decoder. The upmixer does, for example, receive the downmix
signal and all spatial cues other than the phase
information (ICC and ILD). The upmixer derives a first and
a second intermediate audio signal, having signal
properties as described by the spatial cues. To this end,
the generation of an additional reverberation
(decorrelated) signal may be foreseen in order to mix
decorrelated signal portions (wet signals) and the
transmitted downmix channel (dry signal).
However, the intermediate signal post processor does apply
an additional phase shift to at least one of the
intermediate signals, when phase information is received by

the audio decoder. That is, the intermediate signal post
processor is only operative when the additional phase
information is transmitted. That is, embodiments of
inventive audio decoders are fully compatible with a
conventional audio decoder.
The processing in some embodiments of decoders may, as well
as on the encoder side, be performed in a time and
frequency selective manner. That is, a consecutive series
of neighbouring time slices having multiple frequency bands
may be processed. Therefore, some embodiments of audio
encoders incorporate a signal combiner in order to combine
the generated intermediate audio signals and post processed
intermediate audio signals, such that the encoder outputs
time-continuous audio signal
That is, for a first frame (time segment), the signal
combiner may use the intermediate audio signals derived by
the upmixer and, for a second frame, the signal combiner
may use the post processed intermediate signal, as it is
derived by the intermediate signal post processor. Further
to introducing a phase shift, it is, of course, also
possible to implement a more sophisticated signal
processing into the intermediate signal post processor.
Alternatively, or additionally, embodiments of audio
decoders may comprise a correlation information processor,
such as to post-process a received correlation information
ICC, when phase information is additionally received. The
post processed correlation information may then be used by
a conventional upmixer, to generate the intermediate audio
signals, such that, in combination with the phase shift
introduced by the signal post processor, a naturally
sounding reproduction of the audio signals may be achieved.
Several embodiments of the present invention will in the
following be described, referencing the enclosed figures,
wherein

Fig. 1 shows an upmixer generating two output signals
from a downmix signal;
Fig. 2 shows an example for a use of ICC parameters by
the upmixer of Fig. 1;
Fig. 3 shows examples for signal characteristics of
audio input signals to be encoded;
Fig. 4 shows an embodiment of an audio encoder;
Fig. 5 shows a further embodiment of an audio encoder;
Fig. 6 shows an example for an encoded representation of
an audio signal generated by one of the encoders
of Figs. 4 and 5;
Fig. 7 shows a further embodiment of an encoder;
Fig. 8 shows a further embodiment of an encoder for
speech/music encoding;
Fig. 9 shows an embodiment of a decoder;
Fig. 10 shows a further embodiment of a decoder;
Fig. 11 shows a further embodiment of a decoder;
Fig. 12 shows an embodiment of a speech/music decoder;
Fig. 13 shows an embodiment of a method for encoding; and
Fig. 14 shows an embodiment of a method for decoding.
Fig. 1 shows an upmixer as it may be used within an
embodiment of a decoder to generate a first intermediate
audio signal 2 and a second intermediate audio signal 4,

using a downmix signal 6. Furthermore, an additional
interchannel correlation information and an interchannel
level difference information is used as steering parameters
of amplifiers to control the upmix.
The upmixer comprises a decorrelator 10, three correlation
related amplifiers 12a to 12c, a first mixing node 14a, a
second mixing note 14b, as well as first and second level
related amplifiers 16a and 16b. The downmix audio signal 6
is a mono signal, which is distributed to the decorrelator
10 as well as to the input of decorrelation related
amplifiers 12a and 12b. The decorrelator 10 creates, using
the downmix audio signal 6, a decorrelated version of same
by means of a decorrelation algorithm. The decorrelated
audio channel (decorrelated signal) is input into the third
of the correlation related amplifiers 12c. It may be noted
that signal components of the upmixer which only comprise
samples of the downmix audio signals are often also called
"dry" signals, whereas signal components only comprising
samples of the decorrelated signal are often called "wet"
signals.
The ICC related amplifiers 12a to 12c scale the wet and the
dry signal components, according to a scaling rule
depending on the transmitted ICC parameter. Basically, the
energy of those signals is adjusted prior to a summation of
the dry and wet signal components by the summation nodes
14a and 14b. To this end, the output of the correlation
related amplifier 12a is provided to a first input of the
first summation node 14a and the output of the correlation
related amplifier 12b is provided to a first input of
summation node 14b. The output of the correlation related
amplifier 12c associated to the wet signal is provided to a
second input of the first summation node 14a as well as to
a second input of the second summation node 14b. However,
as indicated in Fig. 1, the sign of the wet signal at the
summation nodes differs, in that it is input into the first
summation node 14a with negative sign, whereas the wet

signal with its original sign is input into the second
summation node 14b. That is, the decorrelated signal is
mixed with the first dry signal component with its original
phase, whereas it is mixed with the second dry signal
component with an inverted phase, i.e. with an phaseshift
of 180°.
The energy ratio was, as already explained, preecedingly
adjusted in dependence of the correlation parameter, such
that the signals output from the summation nodes 14a and
14b have a correlation similar to correlation of the
originally encoded signals (which is parametrized by the
transmitted ICC parameter). Finally, an energy relation
between the first channel 2 and the second channel 4 is
adjusted, using the energy related amplifiers 16a and 16b.
The energy relation is parametrized by the ILD parameter,
such that both amplifiers are steered by a function
depending on the ILD parameter.
That is, the so generated left and right channels 2 and 4
have a statistical dependence being similar to the
statistical dependence of the originally encoded signals.
However, the contributions to the generated first (left)
and second (right) output signals 2 and 4 originating
directly from the transmitted downmix audio signal 6 have
identical phases.
Although Fig. 1 assumes a broadband implementation of the
upmix, further implementations may perform the upmix
individually for multiple parallel frequency bands, such
that the upmixer of Fig. 4 may operate on a bandwith
limited representation of the original signal. The
reconstructed signal with the full band with could then be
gained by adding all bandwith limited output signals in a
final synthesis mixture.

Fig. 2 shows an example of a ICC parameter dependent
function used to steer the correlation related amplifiers
12a to 12C. Using that function and appropriately deriving
a ICC parameter from original channels to be encoded, the
phaseshift between the originally encoded signals may be
coarsely reproduced (on the average). For this discussion,
an understanding of the generation of the transmitted ICC
parameter is essential. The basis for this discussion may
be a complex inter-channel coherence parameter, derived
between two corresponding signal segments of two input
audio signals to be encoded, which is defined as follows:

In the preceding equation, 1 indexes the number of samples
within the signal segment processed, whereas the optional
index k denotes one of several subbands, which may,
according to some specific embodiments, be represented by
one single ICC parameter. In other words, X1 and X2 are the
complex-valued subband samples of the two channels, k is
the subband index and 1 is the time index.
The complex-valued subband samples may be derived by
feeding the originally sampled input signals into a QMF-
filterbank, deriving for example 64 subbands, wherein the
samples within each of the subbands are represented by a
complexe-valued number. Calculating a complex cross
correlation using the previous formula, two corresponding
signal segments are characterized by one complex-valued
parameter, the parameter ICCcomplex, which has the following
properties:

Its length |ICCcomplex| represents the coherence of the two
signals. The longer the vector, the more statistical
dependence is between the two signals.
That is, whenever the length or the absolute value of
ICCcomplex equals 1, both signals are, apart from one global
scaling factor, identical. However, they may have a
relative phase difference, which is then given by the phase
angle of ICCcomplex. In that case, the angle of ICCcomplex with
respect to the real axis represents the phase angle between
the two signals. However, when the derivation of ICCcomplex
is performed using more than one subband (that is, k>=2),
the phase angle is consequently an average angle for all
the processed parameter bands.
In other words, when the two signals are statistically
strongly dependent ( ICCcomplex≈1), the real part Re
(ICCcomplex) is approximately the cosine of the phase angle,
and thus the cosine of the phase difference between the
signals.
When the absolute value of ICCcomcomplex is significantly lower
than 1, the angle 9 between the vector ICCcompcomplex and the
real axis can no longer be interpreted to be a phase angle
between identical signals. It is then rather a best
matching phase between statistically fairly independent
signals.
Fig. 3 gives three examples 20a, 20b and 20c of possible
vectors ICCcomplex. The absolute value (length) of vector 20a
is close to unity, meaning that the two signals represented
by the vector 20a are nearly the same but phase shifted
with respect to each other. In other words, both signals
are highly coherent. In that case, the phase angle 30 (0)
directly corresponds to a phase shift between the almost
identical signals.

However, if an evaluation of ICCcomplex results in vector
20b, the meaning of the phase angle 0 is no longer that
well determined. Since the complex vector 20b has an
absolute value significantly lower than 1, both analyzed
signal portions or signals are statistically fairly
independent. That is, the signal within the observed time
segments have no common shape. Still, the phase angle 30
represents somewhat of a phase shift corresponding to the
best match of both signals. However, when the signals are
incoherent, a common phase shift between the two signals is
hardly of any significance.
Vector 20c, again, has an absolute value close to unity,
such that its phase angle 32 (O) may again be unambiguously
identified as a phase difference between two similar
signals. Furthermore, it is apparent that a phase shift
greater than 90° corresponds to a real part of the vector
ICCcomplex, which is smaller than 0.
In audio coding schemes focusing on the correct
construction of the statistical dependence of two or more
coded signals, a possible upmix procedure to create a first
and a second output channel from a transmitted downmix
channel is illustrated in Fig. 1.
As an ICC dependent function to control the correlation
related amplifiers 20a-20c, the function illustrated in
Fig. 2 is often used, to allow for a smooth transition from
totally correlated to total decorrelated signals, without
introducing any discontinuities. Fig. 2 shows how the
signal energies are distributed between the dry signal
components (by steering amplifiers 12a and 12b) and the wet
signal component (by steering amplifier 12c). To achieve
this, the real part of ICC complex is transmitted as a
measure for the length of ICCcomplex and thus for the
similarity between signals.

In Fig. 2, the x-axis gives the value of the transmitted
ICC parameter and the y-axis gives the amount of energy of
the dry signal (solid line 30a) and of the wet signal
(dashed line 30b) mixed together by the summation nodes 14a
and 14b of the upmixer. That is, when the signals are
perfectly correlated (same signal shape, same phase), the
ICC parameter transmitted will be unity. Therefore, the
upmixer distributes the received downmix audio signal 6 to
the outputs, without adding any wet signal parts. As the
downmix audio signal is essentially the sum of the original
channels encoded, the reproduction is correct with respect
to the phase and to the correlation.
If, however, the signals are anti-correlated (phase = 180°,
same signal shape), the transmitted ICC parameter is -1.
Therefore, the reconstructed signal will comprise no signal
portions of the dry signal, but only signal components of
the wet signal. As the wet signal portion is added to the
first audio channel and substracted from the second audio
channel generated, the phase shift between the signals is
correctly reconstructed to be 180°. However, the signal
comprises no dry signal portions at all. This is
unfortunate, since the dry signal actually comprises the
whole direct information transmitted to the decoder.
Therefore, the signal quality of the reconstructed signal
may be decreased. However, the decrease may be dependent on
the signal type encoded, i.e., on the signal characteristic
of the underlying signal. In general terms, the correlated
signals provided by decorrelator 10 have a reverberation-
like sound characteristic. That is, for example, the
audible distortion from only using the decorrelated signal
is rather low for music signals as compared to speech
signals, where a reconstruction from a reverberated-audio
signal leads to an unnatural sounding.
In summarizing, the previously described decoding scheme
does only coarsely approximate the phase properties, since

these are, at best, restored on the average. This is an
extremely coarse approximation, since it is only achieved
by varying the energy of the signal added, wherein the
signal portions added have a relative phase difference of
180°. For signals that are clearly decorrelated or even
anti-correlated (ICC ≤ 0), a significant amount of
decorrelated signal is necessary to restore this
decorrelation, i.e., the statistical independence between
the signals. As, generally, the decorrelated signal as
output of allpass filters has a "reverb-like" sound, the
overall achievable quality is strongly degraded.
As already mentioned, for some signal types, the
restoration of the phase relation may be less important,
but for other signal types, the correct restoration may be
perceptually relevant. In particular, the reconstruction of
an original phase relation may be required, when a phase
information derived from the signals satisfies certain
perceptually motivated phase reconstruction criteria.
Several embodiments of the present invention do, therefore,
include phase information into a encoded representation of
audio signals, when certain phase properties are
fullfilled. That is, phase information is only occasionally
transmitted, when the benefit (in a rate-distortion
estimation) is significant. Moreover, the transmitted phase
information may be coarsely quantized, such that only an
insignificant amount of additional bit rate is required.
Given the transmitted phase information, it is possible to
reconstruct the signal with a correct phase relation
between the dry signal components, that is, between the
signal components directly derived from the original
signals, which are, therefore, perceptually highly
relevant.
If, for example, signals are encoded with an ICCcomplex-
vector 20c, the transmitted ICC parameter (the real part of

ICCcomplex) is approximately -0.4. That is, in the upmix,
more than 50% of the energy will be derived from the
decorrelated signal. However, as an audible amount of
energy is still originating from the downmix audio channel,
the phase relation between the signal components
originating from the downmix audio channel is still
important, since audible. That is, it may be desirable to
approximate the phase relation between the dry signal
portions of the reconstructed signal more closely.
Therefore, additional phase information is transmitted,
once it is determined that a phase shift between the
original audio channels is greater than a predetermined
threshold. Examples for such a threshold may be 60°, 90° or
120°, depending on the specific implementation. Depending
on the threshold, the phase relation may be transmitted
with high resolution, i.e., one of multiple predetermined
phase shifts is signaled, or a continuously varying phase
angle is transmitted.
In some embodiments of the present invention, only a single
phase shift indicator or phase information is transmitted,
indicating that the phase of the reconstructed signals
shall be shifted by a predetermined phase angle. According
to one embodiment, this phase shift applies only when the
ICC parameter is within a predetermined negative range.
This range could, for example, be the range from -1 to -0.3
or from -0.8 to -0.3 dependent on that phase threshold
criterion. That is, one single bit of phase information may
be required.
When the real part of ICCcomplex is positive, the phase
relation between the reconstructed signals are, on the
average, approximated correctly by the upmixer of Fig. 1
due to the phase-identical processing of the dry signal
components.

If, however, the transmitted ICC parameter is below 0, the
phase shift of the original signals is, on the average,
greater than 90°. At the same time, still audible signal
portions of the dry signal are used by the upmixer.
Therefore, in an area starting from ICC = 0 to, say, ICC
approximately -0.6, a fixed phase shift (corresponding for
example to the phase shift corresponding to the middle of
the previously introduced interval) may provide for a
significantly increased perceptual quality of the
reconstructed signal, at the cost of only one single
transmitted bit. When the ICC parameter proceeds to ever
smaller values, for example, lower than -0.6, only small
amounts of signal energy in the first and second output
channels 2 and 4 originate from the dry signal component.
Therefore, restoring the correct phase properties between
those perceptually less relevant signal portions may again
be skipped, since the dry signal portions are hardly
audible at all.
Fig. 4 shows one embodiment of an inventive encoder for
generating an encoded representation of a first input audio
signal 40a and a second input audio signal 40b. The audio
encoder 42 comprises a spatial parameter estimator 44, a
phase estimator 4 6, an output operation mode decider 48 and
an output interface 50.
The first and second input audio signals 40a and 40b are
distributed to the spatial parameter estimator 44 as well
as to the phase estimator 46. The spatial parameter
estimator is adapted to derive spatial parameters,
indicating a signal characteristic of the two signals with
respect to each other, such as for example an ICC parameter
and an ILD parameter. The estimated parameters are provided
to the output interface 50.
The phase estimator 46 is adapted to derive phase
information of the two input audio signals 40a and 4 0b.
Such phase information could, for example, be a phase shift

between the two signals. The phase shift could, for
example, be directly estimated by performing a phase
analysis of the two input audio signals 40a and 40b
directly. In a further alternative embodiment, the ICC
parameters derived by the spatial parameter estimator 44
may be provided to the phase estimator via an optional
signal line 52. The phase estimator 46 could then perform
the phase difference determination using the ICC parameters
anyway derived. This may lead to an implementation with
lower complexity, as compared to an embodiment with full
phase analysis of the two audio input signals.
The phase information derived is provided to the output
operation mode decider 48, which is able to switch the
output interface 50 between a first output mode and a
second output mode. The phase information derived is
provided to the output interface 50, which creates an
encoded representation of the first and the second input
audio signals 40a and 40b by including specific subsets of
the generated ICC, ILD or PI (phase information) parameters
into the encoded representation. In the first mode of
operation, the output interface 50 includes the ICC, the
ILD and the phase information PI into the encoded
representation 54. In the second mode of operation, the
output interface 50 includes only the ICC and the ILD
parameter into the encoded representation 54.
The output mode decider 48 decides for the first output
mode, when the phase information indicates a phase
difference between the first and the second audio signals
40a and 40b, which is greater than a predetermined
threshold. The phase difference could, for example, be
determined by performing a complete phase analysis of the
signal. This could, for example, be performed by shifting
the input audio signals with respect to each other and by
calculating the cross-correlation for each of the signal
shifts. The cross-correlation with the highest value
corresponds to the phaseshift.

The phase information introduced into the representation
could, for example, be a single bit indicating a
predetermined phase shift. Alternatively, the transmitted
phase information could be more precise by transmitting
phase shifts in a finer quantization, up to a continuous
representation of a phase shift.
Furthermore, the audio encoder could operate on a band
limited copy of the input audio signals, such that several
audio encoders 43 of Fig. 4 are implemented in parallel,
each audio encodux operating on a bandwidth filtered
version of an original broadband signal.
Fig. 5 shows a further embodiment of an inventive audio
encoder, comprising a correlation estimator 62, a phase
estimator 46, a signal characteristic estimator 66 and an
output interface 68. The phase estimator 46 corresponds to
the phase estimator introduced in Fig. 4. A further
discussion of the properties of the phase estimator is
therefore omitted to avoid unnecessary redundancies.
Generally, components having the same or similar
functionalities are given the same references. The first
input audio signal 40a and the second input audio signal
40b are distributed to the signal characteristic estimator
66, the correlation estimator 62 and the phase estimator
46.

The signal characteristic estimator is adapted to derive
signal characterization information, which indicates a
first or a second different characteristic of the input
audio signal. For example, a speech signal could be
detected as a first characteristic and a music signal could
be detected as a second signal characterization. The
additional signal characteristic information can be used to
determine the need for the transmission of phase
information or, additionally, to interpret the correlation
parameter in terms of a phase relation.
In one embodiment, the signal characterization estimator 66
is a signal classifier, used to derive the information,
whether the current extract of the audio signal, i.e. of
the first and second input audio channels 40a and 40b, is
speech-like or non-speech. Dependent on the derived signal
characteristic, phase estimation by the phase estimator 46
could be switched on and off via an optional control link
70. Alternatively, phase estimation could be performed all
the time, while the output interface is steered via an
optional second control link 72, such as to include the
phase information 74 only, when the first characteristic of
the input audio signal, i.e. for example, the speech-
characteristic, is detected.
To the contrary, ICC-determination is performed all the
time, such as to provide a correlation parameter required
for an upmix of an encoded signal.
A further embodiment of an audio encoder may, optionally,
comprise a downmixer 7 6, adapted to derive a Downmix audio
signal 78, which could, optionally be included into the
encoded representation 54 provided by the audio encoder 60.
In an alternative embodiment, the phase information could
be based on an analysis of the correlation information ICC,
as already discussed for the embodiment of Fig. 4. To this
end, the output of the correlation estimator 62 may be

provided to the phase estimator 4 6 via an optional signal
line 52.
Such determination could, for example, be based on ICCcomplex
according to the following considerations, when the signal
is discriminated between being a speech-signal and a music-
signal.
When it is known from the signal characteristic information
66, that the signal is a speech-signal, one could evaluate
ICCcomplex

according to the following considerations. When a speech-
signal is determined, it may be concluded that the signal
received by the human auditory signal is strongly
correlated, since the origin of a speech-signal is point-
like. Therefore, the absolute value of ICCcomplex is close to
1. Therefore, the phase angle 0 (IPD) of Fig. 3 can be
estimated by using only the information on the real part of
ICCcomplex according to the following formula, without even
valuating the complex vector ICCcomplex:

Phase information may be gained based on the real part of
ICCcomplex, which could be determined without ever
calculating the imaginary part of ICCcomplex.
In short, one could conclude


In the above equation, please note that cos(IPD)
corresponds to cos(Θ) of Fig. 3.
The necessity to perform a phase-synthesis on the decoder
side could, more generally, also be derived according to
the following considerations:
Coherence (abs (ICCcomplex) significantly >0, Correlation
(Real (ICCcomplex) ) significantly < 1, or Phase angle
(arg(ICCcomplex)) significantly different from 0.
Please note that these are general criteria, wherein at the
presence of speech, it is implicitly assumed that abs
(ICCcomplex) is significantly greater than 0.
Fig. 6 gives an example of an encoded representation
derived by the encoder 60 of Fig. 5. Corresponding to a
time segment 80a and to a first time segment 80b, the
encoded representation only comprises correlation
information, wherein for the second time segment 80c, the
encoded representation generated by the output interface 68
comprises correlation information as well as phase
information PI. In short, an encoded representation
generated by the audio encoder may be characterized in that
it comprises a downmix signal (not shown for simplicity),
which is generated using a first and a second original
output channel. The encoded representation further
comprises a first correlation information 82a indicating a
correlation between the first and the second original audio
channels within a first time segment 80b. The
representation does furthermore comprise a second
correlation information 82b indicating a decorrelation
between the first and the second audio channels within a

second time segment 80c and first phase information 84,
indicating a phase relation between the first and the
second original audio channel for the second time segment,
wherein no phase information is included for the first time
segment 80b. Please note that for the ease of illustration,
Fig. 6 only illustrates the side information, whereas the
downmix channel which is also transmitted, is not shown.
Fig. 7 schematically shows a further embodiment of the
present invention, in which an audio encoder 90 furthermore
comprises a correlation information modifier 92. The
illustration of Fig. 7 assumes, that the spatial parameter
extraction of, for example, the parameters ICC and ILD, has
already been performed, such that the spatial parameters 94
are provided together with the audio signal 96. The audio
encoder 90 furthermore comprises a signal characteristic
estimator 66 and a phase estimator 46, operating as
indicated above. Dependent on the result of the signal
classification and/or the phase analysis, phase parameters
are extracted and submitted according to a first mode of
operation, indicated by the upper signal path.
Alternatively, a switch 98, which is steered by the signal
classification and/or the phase analysis may activate a
second mode of operation, where the provided spatial
parameters 94 are transmitted without modification.
However, when the first mode of operation requiring the
transmission of phase information is chosen, the
correlation information modifier 92 derives a correlation
measure from the received ICC-parameters, which is
transmitted instead of the ICC-parameters. The correlation
measure is chosen such that it is greater than the
correlation information, when a relative phase shift
between the first and the second input audio signals is
determined, and when the audio signal is classified to be a
speech-signal. Additionally, phase parameters are extracted
and transmitted by phase parameter extractor 100.

The optional ICC adjustment or the determination of a
correlation measure, which is to be submitted instead of
the originally derived ICC-parameter, may have the effect
of an even better perceptual quality, since it accounts for
the fact that for ICC s smaller than 0, the reconstructed
signal would comprise only less than 50% of the dry signal,
which are actually the only signals derived directly from
the original audio signals. That is, although one knows,
that the audio signals can only differ significantly by a
phase shift, the reconstruction provides a signal, which is
dominated by the decorrelated signal (the wet signal). When
the ICC-parameter (the real part of ICCcomplex) is increased
by the correlation information modifier, the upmix will
automatically use more signal energy from the dry signal,
such using more of the "genuine" audio information, such
that the reproduced signal is even closer to the original,
when the necessity of a phase reproduction is derived.
In other words, the transmitted ICC-parameters are modified
in a way that the decoder upmix adds less decorrelated
signal. One possible modification of the ICC parameter is
to use the interchannel coherence (absolute value of
ICCcomplex) instead of the interchannel cross-correlation
usually used as the ICC-parameter. Interchannel cross-
correlation is defined as:

and depends on the phase relation of the channels.
Interchannel coherence, however, is independent of the
phase relation and defined as follows:


The interchannel phase difference is calculated and
transmitted to the decoder together with the remaining
spatial side information. The representation can be very
coarse in quantization of the actual phase values and may
furthermore have a coarse frequency resolution, wherein
even a broadband phase information may be beneficial, as it
will be apparent from the embodiment of Fig. 8.
The phase difference may be derived from the complex
interchannel relations as follows:

If the phase information is included in the bit stream,
i.e. into the encoded representation 54, a decoder's
decorrelation synthesis may use the modified ICC-parameters
(the correlation measures) to produce an upmix signal with
reduced reverberation.
If, for example, the signal classifier discriminates
between speech and music signals, a decision whether the
phase synthesis is required, could be taken according to
the following rules, once a predominant speech-
characteristic of the signal is determined.
First of all, a broad-band indication value or phase shift
indicator may be derived, for several of the parameter
bands, used to generate the ICC and ILD parameters. That
is, for example, a frequency range predominantly populated
by speech signals could be evaluated (for example between
100Hz and 2KHz). One possible evaluation would be to
calculate the mean correlation within this frequency range,
based on the already derived ICC-parameters of the
frequency bands. If it turns out that this mean correlation
is smaller than a predetermined threshold, the signal may
be assumed to be out of phase and a phase shift is
triggered. Furthermore, multiple thresholds may be used to
signal different phase shifts, depending on the desired

granularity of the phase reconstruction. Possible threshold
values could, for example, be 0, -0.3 or -0.5.
Fig. 8 shows a further embodiment of the present invention,
in which the encoder 150 is operative to encode speech and
music signals. The first and second input audio signals 40a
and 40b are provided to the encoder 150, which comprises a
signal characteristic estimator 66, a phase estimator 46, a
downmixer 152, a music core-coder 154, a speech core-coder
156 and a correlation information modifier 158. The signal
characteristic estimator 66? is adapted to discriminate
between a speech characteristic as first signal
characteristic and a music characteristic as a second
signal characteristic. Via control link 160, the signal
characteristic estimator 66 is operative to steer the
output interface 68, depending on the signal characteristic
derived.
The phase estimator estimates phase information, either
directly from the input audio channels 40a and 4 0b or from
the ICC-parameter derived by the downmixer 152. The
downmixer creates a downmix audio channel M (162) and
correlation information ICC (164). According to the
previously described embodiments, the phase information
estimator 4 6 may alternatively derive the phase information
directly from the provided ICC-parameters 164. The downmix
audio channel 162 can be provided to the music core coder
154 as well as to the speech core coder 156, both of which
are connected to the output interface 68 to provide the
encoded representation of the audio downmix channel. The
correlation information 164 is, on the one hand, directly
provided to the output interface 68. On the other hand, it
is provided to the input of a correlation information
modifier 158, adapted to modify the provided correlation
information and to provide the so derived correlation
measure to the output interface 68.

The output interface includes different subsets of
parameters into the decoded representation, depending on
the signal characteristic estimated by the signal
characteristic estimator 66. In a first (speech) mode of
operation, the output interface 68 includes the encoded
representation of the downmix audio channel 106; encoded by
the speech core-coder 156, as well as phase information PI
derived from the phase estimator 46 and the correlation
measure. The correlation measure may either be the
correlation parameter ICC derived by the downmixer 152, or,
alternatively, a correlation measure modified by the
correlation information modifier 158. To this end, the
correlation information modifier 158 may be steered and/or
activated by the phase information estimator 46.
In a music mode of operation, the output interface includes
the downmix audio channel 162 as encoded by the music core-
coder 154 and the correlation information ICC as derived
from the downmixer 152.
It goes without saying that the inclusion of the different
parameter subsets may be implemented different as in the
particular embodiment described above. For example, the
music and/or speech coders may be deactivated,, until a
activation signal switches them into the signal path,
depending on the signal characteristic derived from the
signal characteristic estimator 66.
Fig. 9 shows an embodiment of a decoder according to the
present invention. The audio decoder 200 is adapted to
derive a first audio channel 202a and a second audio
channel 202b from an encoded representation 204, the
encoded representation 204 comprising a downmix audio
signal 206a, first correlation information 208 for the
first time segment of the downmix signal and second
correlation information 210 for a second time segment of
the downmix signal, wherein phase information 212 is only
included for the first or second time segment.

In other words, the first time segment is reconstructed
using decorrelation information ICC1 and the second time
segment is reconstructed usin, ICC2. The first and second
intermediate signals 222a and 222b are provided to an
intermediate signal postprocessor 224, adapted to derive a
postprocessed intermediate signal 226 for the first time
segment using the corresponding phase information 212. To
this end, the intermediate signal postprocessor 224
receives the phase information 212, together with the
intermediate signals generated by the upmixer 220. The
intermediate signal postprocessor 224 is adapted to add a
phase shift to at least one of the audio channels of the
intermediate audio signals, when phase information
corresponding to the particular audio signal is present.
That is, the intermediate signal postprocessor 224 adds a
phase shift to the first intermediate audio signal 222a
wherein the intermediate postprocessor does not add any
phase shift to the intermediate audio signal 222b. The
intermediate signal postprocessor 224 outputs
postprocessed intermediate signal 226 instead of the first

intermediate audio signal and an unaltered second
intermediate audio signal 222b.
The audio decoder 200 further comprises a signal combiner
230, to combine the signals output from the intermediate
signal postprocessor 224, and to thus derive the first and
second audio channels 202a and 202b generated by the audio
decoder 200.
In one particular embodiment, the signal combiner
concatenates the signals as output from the intermediate
signal postprocessor, to finally derive an audio signal for
the first and second time segments. In a further
embodiment, the signal combiner may implement some cross
fading, such as to derive the first and second audio
signals 202a and 202b by fading between the signals
provided from the intermediate signal postprocessor. Of
course, further implementations of the signal combiners 230
are feasible.
Using an embodiment of an inventive decoder as illustrated
in Fig. 9 provides for the flexibility to add a additional
phase shift, as it may be signaled by an encoder signal, or
decode the signal in a backwards compatible manner.
Fig. 10 shows a further embodiment of the present
invention, in which the audio decoder comprises a
decorrelation circuit 243, capable of operating according
to a first decorrelation rule and according to a second
decorrelation rule, depending on the transmitted phase
information. According to the embodiment of Fig. 10, the
decorrelation rule, according to which a decorrelated
signal 242 is derived from the transmitted downmix audio
channel 240 can be switched, wherein the switching depends
on the existing phase information.
In a first mode, in which phase information is transmitted,
a first decorrelation rule is used in order to derive the

decorrelated signal 242. In a second mode, in which phase
information is not received, a second decorrelation rule is
used, creating a decorrelated signal, which is more
decorrelated than the signal created using the first
decorrelation rule.
That is, when phase synthesis is required, a decorrelated
signal may be derived, which is not as highly decorrelated
as the signal used when no phase synthesis is required.
That is, a decoder may then use a decorrelated signal,
which is more similar to the dry signal, as such
automatically creating a signal having more dry-signal
components in the upmix. This is achieved by making the
decorrelated signal more similar to the dry signal.
In a further embodiment, an optional phase shifter 246 may
be applied to the decorrelated signal generated for a
reconstruction with phase synthesis. This provides a closer
reconstruction of the phase properties of the reconstructed
signal, by providing a decorrelated signal already having
the correct phase relation with respect to the dry signal.
Fig. 11 shows a further embodiment of an inventive audio
decoder, comprising an analysis filter bank 260 and a
synthesis filter bank 262. The decoder receives a downmix
audio signal 206 together with the related ICC-parameters
(ICCo ... ICCn) . However, in Fig. 11, the different ICC-
parameters are not only associated to different time
segments but also to different frequency bands of the audio
signal. That is, each time segment process has a full set
of associated ICC- parameters (ICC0 ... ICCn) .
As the processing is performed in a frequency selective
manner, the analysis filterbank 260 derives 64 subband
representations of the transmitted downmix audio signal
206. That is, 64 bandwidth limited signals (in the
filterbank representation) are derived, each signal being
associated with one ICC-parameter. Alternatively, several

bandwidth limited signals may share a common ICC parameter.
Each of the subband representations is processed by an
upmixer 264a, 264b, .... Each of the upmixers could, for
example, be an upmixer in accordance with the embodiment of
Fig. 1.
Therefore, for each bandwidth limited representation, a
first a the second audio channel (both bandwidth limited)
are created. At least one of the so created audio channels
per subband is input into an intermediate audio signal
postprocessor 266a, 266b ..., as, for example, the
intermediate audio signal postprocessor described in Fig.
9. According to the embodiment of Fig. 11, the intermediate
audio signal postprocessors 266a, 266b, ... are steered by
the same, common, phase information 212. That is, an
identical phase shift is applied to each subband signal,
before the subband signals are synthesized by the synthesis
filterbank 262 to become the first and second audio
channels 202a and 202b output by the decoder.
A phase synthesis may thus be performed, requiring only one
additional common phase information to be transmitted. In
the embodiment of Fig. 11, the correct restoration of the
phase properties of the original signal can, therefore, be
performed without a reasonable increase in bit rate.
According to further embodiments, the number of subbands,
for which the common phase information 212 is used, is
signal dependent. Therefore, the phase information may only
be evaluated for subbands, for which an increase in
perceptual quality can be achieved, when a corresponding
phase shift is applied. This may further increase the
perceptual quality of the decoded signal.
Fig. 12 shows a further embodiment of an audio decoder,
adapted to decode an encoded representation of an original
audio signal, which could be both, a speech signal or a
music signal. That is, either a signal characterization

information is transmitted within the encoded
representation, indicating which signal characteristic is
transmitted, or, the signal characteristic may implicitly
be derived, depending on the presence of phase information
in the bit stream. To this end, the presence of phase
information would indicate a speech characteristic of the
audio signal. The transmitted downmix audio signal 206 is,
depending on the signal characteristic, either decoded by a
speech decoder 2 66 or by a music decoder 268. The further
processing is performed as illustrated and explained in
Fig. 11. For the further implementation details, reference
is therefore made to the explanation of Fig. 11.
Fig. 13 illustrates an embodiment of an inventive method
for generating an encoded representation of a first and a
second input audio signal. In a spatial parameter
extraction step 300, an ICC- and an ILD-parameter is
derived from the first and the second input audio signals.
In a phase estimation step 302, phase information
indicating a phase relation between the first and the
second input audio signals is derived. In a mode decision
304, a first output mode is selected, when the phase
relation indicates a phase difference between the first and
the second input audio signal, which is greater than a
predetermined threshold and a second output mode is
selected, when the phase difference is smaller than the
threshold. In a representation generation step 306 the ICC-
parameter, the ILD-parameter and the phase information is
included in the encoded representation in the first output
mode, and the ICC- and the ILD-parameters without the phase
relation are included into the encoded representation in
the second output mode.
Fig. 14 shows an embodiment of a method for generating a
first and a second audio channel using an encoded
representation of an audio signal, the encoded
representation comprising a downmix audio signal, first and
second correlation information indicating a correlation

between a first and a second original audio channel used to
generate the downmix signal, the first correlation
information having the information for a first time segment
of the downmix signal and the second correlation
information having the information for a second, different
time segment, and phase information, the phase information
indicating a phase relation between the first and the
second original audio channels for the first time segment.
In an upmixing step 400, a first intermediate audio signal
is derived using the downmix signal and the first
correlation information, the first intermediate audio
signal corresponding to the first time segment and
comprising a first and a second audio channel. In the
upmixing step 400, a second intermediate audio signal using
the downmix audio signal and the second correlation
information is also derived, the second
intermediate audio signal corresponding to the second time
segment and comprising a first and a second audio channel.
In a postprocessing step 402, a postprocessed intermediate
signal is derived for the first time segment, using the
first intermediate audio signal, wherein an additional
phase shift indicated by the phase relation is added to at
least one of the first or the second audio channels of the
first intermediate audio signal.
In a signal combination step 404 , the first and the second
audio channels are generated, using the postprocessed
intermediate signal and the second intermediate audio
signal.
Depending on certain implementation requirements of the
inventive methods, the inventive methods can be implemented
in hardware or in software. The implementation can be
performed using a digital storage medium, in particular a
disk, DVD or a CD having electronically readable control
signals stored thereon, which cooperate with a programmable

computer system such that the inventive methods are
performed. Generally, the present invention is, therefore,
a computer program product with a program code stored on a
machine readable carrier, the program code being operative
for performing the inventive methods when the computer
program product runs on a computer. In other words, the
inventive methods are, therefore, a computer program having
a program code for performing at least one of the inventive
methods when the computer program runs on a computer.
While the foregoing has been particularly shown and
described with reference to particular embodiments thereof,
it will, be understood by those skilled in the art that
various other changes in the form and details may be made
without departing from the spirit and scope thereof. It is
to be understood that various changes may be made in
adapting to different embodiments without departing from
the broader concepts disclosed herein and comprehended by
the claims that follow.

WE CLAIM
1. Audio encoder for generating an encoded representation
of a first and a second input audio signal,
comprising:
a correlation estimator adapted to derive correlation
information indicating a correlation between the first
and the second input audio signals;
a signal characteristic estimator adapted to derive
signal characterization information, the signal
characterization information indicating a first or a
second, different characteristic of the input audio
signal;
a phase estimator adapted to derive phase information
when the input audio signals have the first
characteristic, the phase information indicating a
phase relation between the first and the second input
audio signals; and
an output interface, adapted to include
the phase information and a correlation measure
into the encoded representation when the input
audio signals have the first characteristic; or
the correlation information into the encoded
representation when the input audio signals have
the second characteristic, wherein the phase
information is not included when the input audio
signals have the second characteristic.
2. The audio encoder of claim 1, wherein the first signal
characteristic indicated by the signal estimator is a
speech characteristic; and

the second signal characteristic indicated by the
signal estimator is a music characteristic.
3. The audio encoder of claim 1, wherein the phase
estimator is adapted to derive the phase information
using the correlation information.
4. The audio encoder of claim 1, wherein the phase
information indicates a phase shift between the first
and the second input audio signals.
5. The audio encoder of claim 3, wherein the correlation
estimator is adapted to generate an ICC-parameter as
the decorrelation information, the ICC-parameter
represented by a real part of a complex cross-
correlation ICCcomplex of sampled signal segments of the
first and the second input audio signal, each signal
segment being represented by 1 sample values X(l),
wherein the ICC-parameter can be described by the
following formula:

and wherein the output interface is adapted to include
the phase information into the encoded representation,
when the correlation information is smaller than a
predetermined threshold.
6. The audio encoder of claim 5, wherein the
predetermined threshold is equal to or smaller than
0.3.

7. The audio encoder of claim 5, wherein the
predetermined threshold for the correlation
information corresponds to a phase shift of more than
90°;
8. The audio encoder of claim 1, wherein the correlation
estimator is adapted to derive multiple correlation
parameters as the correlation information, each
correlation parameter being related to a corresponding
subband of the first and the second input audio
signals, and wherein the phase estimator is adapted to
derive a phase information indicating the phase
relation between the first and the second input audio
signals for at least two of the subbands corresponding
to the correlation parameters.
9. The audio encoder of claim 1, further comprising a
correlation information modifier adapted to derive the
correlation measure such that the correlation measure
indicates a higher correlation than the correlation
information; and
wherein the output interface is adapted to include the
correlation measure instead of the correlation
information.
10. The audio encoder of claim 9, wherein the correlation
information modifier is adapted to use the absolute
value of a complex cross-correlation ICCcomplex of two
sampled signal segments of the first and the second
input audio signal as the correlation measure ICC,
each signal segment being represented by 1 complex
value sample values X(l), the correlation measure ICC
being described by the following formula:


11. Audio encoder for generating an encoded representation
of a first and a second input audio signal,
comprising:
a spatial parameter estimator adapted to derive an
ICC-parameter or an ILD-parameter, the ICC-parameter
indicating a correlation between the first and the
second input audio signals, the ILD-parameter
indicating a level relation between the first and the
second input audio signals;
a phase estimator adapted to derive a phase
information, the phase information indicating a phase
relation between the first and the second input audio
signals;
an output operation mode decider adapted to indicate
a first output mode when the phase relation
indicates a phase difference between the first
and the second input audio signals which is
greater than a predetermined threshold, or
a second output mode, when the phase difference
is smaller than the predetermined threshold; and
an output interface, adapted to include
the ICC- or the ILD-parameter and the phase
information into the encoded representation in
the first output mode; and

the ICC- and the ILD-parameter without the phase
information into the encoded representation in
the second output mode.
12. The audio encoder of claim 11, wherein the
predetermined threshold corresponds to a phase shift
of 60°.
13. The audio encoder of claim 11, wherein the spatial
parameter estimator is adapted to derive multiple ICC-
or ILD-parameters, each ICC- or ILD-parameter being
related to a corresponding subband of a subband
representation of the first and the second input audio
signals, and wherein the phase estimator is adapted to
derive a phase information indicating the phase
relation between the first and the second input audio
signals for at least two of the subbands of the
subband representation.
14. The audio encoder of claim 13, wherein the output
interface is adapted to include a single phase
information parameter into the representation as the
phase information, the single phase information
parameter indicating the phase relation for a
predetermined subgroup of the subbands of the subband
representation.
15. The audio encoder of claim 11, wherein the phase
relation is represented by a single bit indicating a
predetermined phase shift.
16. Audio decoder for generating a first and a second
audio channel using an encoded representation of an
audio signal, the encoded representation comprising a
downmix audio signal, first and second correlation
information indicating a correlation between a first
and a second original audio channel used to generate
the downmix audio signal, the first correlation

information having the information for a first time
segment of the downmix signal and the second
correlation information having the information for a
second, different time segment, the encoded
representation further comprising phase information
for the first and the second time segment, the phase
information indicating a phase relation between the
first and the second original audio channels,
comprising:
an upmixer adapted to derive
a first intermediate audio signal using the
downmix audio signal and the first correlation
information, the first intermediate audio signal
corresponding to the first time segment and
comprising a first and a second audio channel;
and
a second intermediate audio signal using the
downmix audio signal and the second correlation
information, the second intermediate audio signal
corresponding to the second time segment and
comprising a first and a second audio channel;
and
an intermediate signal postprocessor adapted to derive
a postprocessed intermediate audio signal for the
first time segment using the first intermediate audio
signal and the phase information, wherein the
intermediate signal postprocessor is adapted to add an
additional phase shift indicated by the phase relation
to at least one of the first or the second audio
channels of the first intermediate audio signal; and
a signal combiner adapted to generate the first and
the second audio channel by combining the

postprocessed intermediate audio signal and the second
intermediate audio signal.
17. The audio decoder of claim 16, wherein the upmixer is
adapted to use multiple correlation parameters as the
correlation information, each correlation parameter
corresponding to one of multiple subbands of the first
and second original audio signals; and
wherein the intermediate signal postprocessor is
adapted to add the additional phase shift indicated by
the phase relation to at least two of the
corresponding subbands of the first intermediate audio
signal.
18. The audio decoder of claim 16, further comprising a
correlation information processor adapted to derive a
correlation measure, the correlation measure
indicating a higher correlation than the first
correlation; and
wherein the upmixer uses the correlation measure
instead of the correlation information, when the phase
information indicates a phase shift between the first
and the second original audio channels, which is
higher than a predetermined threshold.
19. The audio decoder according to claim 16, further
comprising a decorrelator adapted to derive a
decorrelated audio channel from the downmix audio
signal according to a first decorrelation rule for the
first time segment and according to a second
decorrelation rule for the second time segment,
wherein the first decorrelation rule creates a less
decorrelated audio channel than the second
decorrelation rule.

20. The audio decoder of claim 19, wherein the
decorrelator further comprises a phase shifter, the
phase shifter adapted to apply an additional phase
shift to the decorrelated audio channel generated
using the first decorreltion rule, the additional
phase shift depending on the phase information.
21. Method for generating an encoded representation of a
first and a second input audio signal, comprising:
deriving correlation information indicating a
correlation between the first and the second input
audio signals;
deriving signal characterization information, the
signal characterization information indicating a first
or a second, different characteristic of the input
audio signals;
deriving phase information when the input audio
signals have the first characteristic, the phase
information indicating a phase relation between the
first and the second input audio signals; and
including the phase information and a correlation
measure into the encoded representation when the
input audio signals have the first
characteristic; or
including the correlation information into the
encoded representation when the input audio
signals have a second characteristic, wherein the
phase information is not included when the input
audio signals have the second characteristic.
22. Method for generating an encoded representation of a
first and a second input audio signal, comprising:

deriving an ICC-parameter or an ILD-parameter, the
ICC-parameter indicating a correlation between the
first and the second input audio signals, the ILD-
parameter indicating a level relation between the
first and the second input audio signals;
deriving a phase information, the phase information
indicating a phase relation between the first and the
second input audio signals;
indicating a first output mode when the phase relation
indicates a phase difference between the first and the
second input audio signals which is bigger than a
predetermined threshold, or indicating a second output
mode when the phase difference is smaller than the
predetermined threshold; and
including the ICC or the ILD parameter and the phase
relation into the encoded representation in the first
output mode; or
including the ICC or the ILD parameter without the
phase relation into the encoded representation in the
second output mode.
23. Method for deriving a first and a second audio channel
using an encoded representation of an audio signal,
the encoded representation comprising a downmix audio
signal, first and second correlation information
indicating a correlation between a first and a second
original audio channel used to generate the downmix
audio signal, the first correlation information having
the information for a first time segment of the
downmix signal and the second correlation information
having the information for a second, different time
segment, the encoded representation further comprising
phase information for the first and the second time
segment, the phase information indicating a phase

relation between the first and the second original
audio channels, comprising:
deriving a first intermediate audio signal using the
downmix audio signal and the first correlation
information, the first intermediate audio signal
corresponding to the first time segment and comprising
a first and a second audio channel;
deriving a second intermediate audio signal using the
downmix audio signal and the second correlation
information, the second intermediate audio signal
corresponding to the second time segment and
comprising a first and the second audio channel;
deriving a post processed intermediate signal for the
first time segment, using the first intermediate audio
signal and the phase information, wherein the post
processed intermediate signal is derived by adding an
additional phase shift indicated by the phase relation
to at least one of the first or the second audio
channels of the first intermediate signal; and
combining the post processed intermediate signal and
the second intermediate audio signal to derive the
first and the second audio channels.
24. Encoded representation of an audio signal, comprising:
a downmix signal generated using a first and a second
original audio channel;
a first correlation information indicating a
correlation between the first and the second original
audio channels within a first time segment;

a second correlation information indicating a
correlation between the first and the second original
audio channels within a second time segment; and
phase information indicating a phase relation between
the first and the second original audio channels for
the first time segment, wherein the phase information
is the only phase information included in the
representation for the first and for the second time
segments.
25. Computer program having a program code for performing,
when running on a computer, any of the methods of claims 21
to 23.

An efficient encoded representation of a first and a second
input audio signal can be derived using correlation
information indicating a correlation between the first and
the second input audio signals, when a signal
characterization information, indicating at least a first
or a second, different characteristic of the input audio
signal is additionally considered. Phase information
indicating a phase relation between the first and the
second input audio signals is derived, when the input audio
signals have the first characteristic. The phase
information and a correlation measure are included into the
encoded representation when the input audio signals have
the first characteristic, and only the correlation
information is included into the encoded representation
when the input audio signals have the second
characteristic.