Apparatus, Method and Computer Program for Providing One or More Adjusted
Parameters for Provision of an Upmix Signal Representation on the Basis of a
Downmix Signal Representation and a Parametric Side Information Associated with
the Downmix Signal Representation, Using an Average Value
Description
Technical Field
An embodiment according to the invention is related to an apparatus for providing one or
more adjusted parameters for a provision of an upmix signal representation on the basis of
a downmix signal representation and a parametric side information associated with the
downmix signal representation.
Another embodiment according to the invention is related to an apparatus for providing an
upmix signal representation on the basis of the downmix signal representation and the
parametric side information.
Another embodiment according to the invention is related to a method for providing one or
more adjusted parameters for a provision of an upmix signal representation on the basis of
a downmix signal representation and a parametric side information associated with the
downmix signal representation.
Another embodiment according to the invention is related to a computer program for
performing said method.
Some embodiments according to the invention are related to a parameter limiting scheme
for distortion control in MPEG SAOC.
Background of the Invention
In the art of audio processing, audio transmission and audio storage, there is an increasing
desire to handle multi-channel contents in order to improve the hearing impression. Usage
of multi-channel audio content brings along significant improvements for the user. For
example, a 3-dimensional hearing impression can be obtained, which brings along an
improved user satisfaction in entertainment applications, However, multi-channel audio
contents are also useful in professional environments, for example in telephone

conferencing applications, because the speaker intelligibility can be improved by using a
multi-channel audio playback.
However, it is also desirable to have a good tradeoff between audio quality and bitrate
requirements in order to avoid an excessive resource load caused by multi-channel
applications.
Recently, parametric techniques for the bitrate-efficient transmission and/or storage of
audio scenes containing multiple audio objects has been proposed, for example, Binaural
Cue Coding (Type I) (see, for example, reference [1]), Joint Source Coding (see, for
example, reference [2]), and MPEG Spatial Audio Object Coding (SAOC) (see, for
example, references [3], [4], [5]).
In combination with user interactivity at the receiving side, such techniques may lead to a
low audio quality of the output signals if extreme object rendering is performed (see, for
example, reference [6]).
These techniques aim at perceptually reconstructing the desired output audio scene rather
than by a waveform match.
Fig. 8 shows a system overview of such a system (here: MPEG SAOC). The MPEG SAOC
system 800 shown in Fig. 8 comprises an SAOC encoder 810 and an SAOC decoder 820.
The SAOC encoder 810 receives a plurality of object signals x1 to XN, which may be
represented, for example, as time-domain signals or as time-frequency-domain signals (for
example, in the form of a set of transform coefficients of a Fourier-type transform, or in the
form of QMF subband signals). The SAOC encoder 810 typically also receives downmix
coefficients d1 to dN, which are associated with the object signals x1 to XN. Separate sets of
downmix coefficients may be available for each channel of the downmix signal. The
SAOC encoder 810 is typically configured to obtain a channel of the downmix signal by
combining the object signals x1 to xN in accordance with the associated downmix
coefficients d1 to dN. Typically, there are less downmix channels than object signals x1 to
XN. In order to allow (at least approximately) for a separation (or separate treatment) of the
object signals at the side of the SAOC decoder 820, the SAOC encoder 810 provides both
the one or more downmix signals (designated as downmix channels) 812 and a side
information 814. The side information 814 describes characteristics of the object signals X1
to XN, in order to allow for a decoder-sided object-specific processing.

The SAOC decoder 820 is configured to receive both the one or more downmix signals
812 and the side information 814. Also, the SAOC decoder 820 is typically configured to
receive a user interaction information and/or a user control information 822, which
describes a desired rendering setup. For example, the user interaction information/user
control information 822 may describe a speaker setup and the desired spatial placement of
the objects which provide the object signals x1 to XN
The SAOC decoder 820 is configured to provide, for example, a plurality of decoded
upmix channel signals ŷ1 toŷM The upmix channel signals may for example be associated
with individual speakers of a multi-speaker rendering arrangement. The SAOC decoder
820 may, for example, comprise an object separator 820a, which is configured to
reconstruct, at least approximately, the object signals X1 to xN on the basis of the one or
more downmix signals 812 and the side information 814, thereby obtaining reconstructed
object signals 820b. However, the reconstructed object signals 820b may deviate
somewhat from the original object signals x1 to XN, for example, because the side
information 814 is not quite sufficient for a perfect reconstruction due to the bitrate
constraints. The SAOC decoder 820 may further comprise a mixer 820c, which may be
configured to receive the reconstructed object signals 820b and the user interaction
information/user control information 822, and to provide, on the basis thereof, the upmix
channel signals ŷ1 to ŷM The mixer 820c may be configured to use the user interaction
information /user control information 822 to determine the contribution of the individual
reconstructed object signals 820b to the upmix channel signals ŷ1 to ŷM. The user
interaction information/user control information 822 may, for example, comprise rendering
parameters (also designated as rendering coefficients), which determine the contribution of
the individual reconstructed object signals 822 to the upmix channel signals ŷ1 to ŷM
However, it should be noted that in many embodiments, the object separation, which is
indicated by the object separator 820a in Fig. 8, and the mixing, which is indicated by the
mixer 820c in Fig. 8, are performed in one single step. For this purpose, overall parameters
may be computed which describe a direct mapping of the one or more downmix signals
812 onto the upmix channel signals ŷ1 to ŷM- These parameters may be computed on the
basis of the side information and the user interaction information/user control information
820.
Taking reference now to Figs. 9a, 9b and 9c, different apparatus for obtaining an upmix
signal representation on the basis of a downmix signal representation and object-related
side information will be described. It should be noted that the object-related side
information is an example of a side information associated with the downmix signal. Fig.

9a shows a block schematic diagram of an MPEG SAOC system 900 comprising an SAOC
decoder 920. The SAOC decoder 920 comprises, as separate functional blocks, an object
decoder 922 and a mixer/renderer 926. The object decoder 922 provides a plurality of
reconstructed object signals 924 in dependence on the downmix signal representation (for
example, in the form of one or more downmix signals represented in the time domain or in
the time-frequency-domain) and object-related side information (for example, in the form
of object meta data). The mixer/renderer 926 receives the reconstructed object signals 924
associated with a plurality of N objects and provides, on the basis thereof and on the
rendering information, one or more upmix channel signals 928. In the SAOC decoder 920,
the extraction of the object signals 924 is performed separately from the mixing/rendering
which allows for a separation of the object decoding functionality from the
mixing/rendering functionality but brings along a relatively high computational
complexity.
Taking reference now to Fig. 9b, another MPEG SAOC system 930 will be briefly
discussed, which comprises an SAOC decoder 950. The SAOC decoder 950 provides a
plurality of upmix channel signals 958 in dependence on a downmix signal representation
(for example, in the form of one or more downmix signals) and an object-related side
information (for example, in the form of object meta data). The SAOC decoder 950
comprises a combined object decoder and mixer/renderer, which is configured to obtain
the upmix channel signals 958 in a joint mixing process without a separation of the object
decoding and the mixing/rendering, wherein the parameters for said joint upmix process
are dependent both on the object-related side information and the rendering information.
The joint upmix process depends also on the downmix information, which is considered to
be part of the object-related side information.
To summarize the above, the provision of the upmix channel signals 928, 958 can be
performed in a one step process or a two step process.
Taking reference now to Fig. 9c, an MPEG SAOC system 960 will be described. The
SAOC system 960 comprises an SAOC to MPEG Surround transcoder 980, rather than an
SAOC decoder.
The SAOC to MPEG Surround transcoder comprises a side information transcoder 982,
which is configured to receive the object-related side information (for example, in the form
of object meta data) and, optionally, information on the one or more downmix signals and
the rendering information. The side information transcoder is also configured to provide an
MPEG Surround side information (for example, in the form of an MPEG Surround

bitstream) on the basis of a received data. Accordingly, the side information transcoder 982
is configured to transform an object-related (parametric) side information, which is
received from the object encoder, into a channel-related (parametric) side information,
taking into consideration the rendering information and, optionally, the information about
the content of the one or more downmix signals.
Optionally, the SAOC to MPEG Surround transcoder 980 may be configured to manipulate
the one or more downmix signals, described, for example, by the downmix signal
representation, to obtain a manipulated downmix signal representation 988. However, the
downmix signal manipulator 986 may be omitted, such that the output downmix signal
representation 988 of the SAOC to MPEG Surround transcoder 980 is identical to the input
downmix signal representation of the SAOC to MPEG Surround transcoder. The downmix
signal manipulator 986 may, for example, be used if the channel-related MPEG Surround
side information 984 would not allow to provide a desired hearing impression on the basis
of the input downmix signal representation of the SAOC to MPEG Surround transcoder
980, which may be the case in some rendering constellations.
Accordingly, the SAOC to MPEG Surround transcoder 980 provides the downmix signal
representation 988 and the MPEG Surround bitstream 984 such that a plurality of upmix
channel signals, which represent the audio objects in accordance with the rendering
information input to the SAOC to MPEG Surround transcoder 980 can be generated using
an MPEG Surround decoder which receives the MPEG Surround bitstream 984 and the
downmix signal representation 988.
To summarize the above, different concepts for decoding SAOC-encoded audio signals can
be used. In some cases, an SAOC decoder is used, which provides upmix channel signals
(for example, upmix channel signals 928, 958) in dependence on the downmix signal
representation and the object-related parametric side information. Examples for this
concept can be seen in Figs. 9a and 9b. Alternatively, the SAOC-encoded audio
information may be transcoded to obtain a downmix signal representation (for example, a
downmix signal representation 988) and a channel-related side information (for example,
the channel-related MPEG Surround bitstream 984), which can be used by an MPEG
Surround decoder to provide the desired upmix channel signals.
In the MPEG SAOC system 800, a system overview of which is given in Fig. 8, the
general processing is carried out in a frequency selective way and can be described as
follows within each frequency band:

• N input audio object signals X1 to XN are downmixed as part of the SAOC encoder
processing. For a mono downmix, the downmix coefficients are denoted by d1 to dN. In
addition, the SAOC encoder 810 extracts side information 814 describing the
characteristics of the input audio objects. For MPEG SAOC, the relations of the object
powers with respect to each other are the most basic form of such a side information.
• Downmix signal (or signals) 812 and side information 814 are transmitted and/or
stored. To this end, the downmix audio signal may be compressed using well-known
perceptual audio coders such as MPEG-1 Layer II or III (also known as ".mp3"),
MPEG Advanced Audio Coding (AAC), or any other audio coder.
• On the receiving end, the SAOC decoder 820 conceptually tries to restore the original
object signal ("object separation") using the transmitted side information 814 (and,
naturally, the one or more downmix signals 812). These approximated object signals
(also designated as reconstructed object signals 820b) are then mixed into a target scene
represented by M audio output channels (which may, for example, be represented by
the upmix channel signals ŷ1 to ŷM) using a rendering matrix. For a mono output, the
rendering matrix coefficients are given by r1 to rN .
• Effectively, the separation of the object signals is rarely executed (or even never
executed), since both the separation step (indicated by the object separator 820a) and
the mixing step (indicated by the mixer 820c) are combined into a single transcoding
step, which often results in an enormous reduction in computational complexity.
It has been found that such a scheme is tremendously efficient, both in terms of
transmission bitrate (it is only necessary to transmit a few downmix channels plus some
side information instead of N discrete object audio signals or a discrete system) and
computational complexity (the processing complexity relates mainly to the number of
output channels rather than the number of audio objects). Further advantages for the user
on the receiving end include the freedom of choosing a rendering setup of his/her choice
(mono, stereo, surround, virtualized headphone playback, and so on) and the feature of
user interactivity: the rendering matrix, and thus the output scene, can be set and changed
interactively by the user according to will, personal preference or other criteria. For
example, it is possible to locate the talkers from one group together in one spatial area to
maximize discrimination from other remaining talkers. This interactivity is achieved by
providing a decoder user interface.

For each transmitted sound object, its relative level and (for non-mono rendering) spatial
position of rendering can be adjusted. This may happen in real-time as the user changes the
position of the associated graphical user interface (GUI) sliders (for example: object level
= +5dB, object position = -30deg).
However, it has been found that the decoder-sided choice of parameters for the provision
of the upmix signal representation (e.g. the upmix channel signals ŷ1 to ŷM) brings along
audible degradations in some cases.
In view of this situation, it is the objective of the present invention to create a concept
which allows for reducing or even avoiding audible distortion when providing an upmix
signal representation (for example, in the form of upmix channel signals ŷ1 to ŷM)-
Summary of the Invention
This problem is solved by an apparatus for providing one or more adapted parameters for a
provision of an upmix signal representation on the basis of a downmix signal
representation and a parametric side information associated with the downmix signal
representation. The apparatus comprises a parameter adjuster configured to receive one or
more parameters (which may be input parameters in some embodiments) and to provide,
on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured
to provide the one or more adjusted parameters in dependence on an average value of a
plurality of parameter values (which may be input parameter values in some
embodiments), such that the distortion of the upmix signal representation caused by the use
of non-optimal parameters is reduced at least for parameters (or input parameters)
deviating from optimal parameters by more than a predetermined deviation.
This embodiment according to the invention is based on the idea that an average value of a
plurality of input parameter values constitutes a meaningful quantity which allows for an
adjustment of parameters, which are used for a provision of an upmix signal representation
on the basis of a downmix signal representation and a parametric side information
associated with the downmix signal representation, because distortions are often caused by
excessive deviations from such an average value. The usage of an average value allows for
an adjustment of one or more parameters, to avoid such excessive deviations from the
average value (also sometimes designated as a mean value), consequently bringing along
the possibility to avoid an excessively degraded audio quality.

The above-discussed embodiment provides a concept for safeguarding the subjective sound
quality of the rendered SAOC scene for which all processing may be carried out entirely
within an SAOC decoder/transcoder, because the SAOC decoder/transcoder comprises the
full information required for the adjustment of the parameters. Also, the above-described
embodiment does not involve the explicit calculation of sophisticated measures of
perceived audio quality of the rendered scene, because it has been found that a limitation of
a deviation between a parameter value and an average value typically results in a good
hearing impression while large deviations between a parameter value and an average value
typically result in audible distortions. Thus, the above-discussed embodiment provides for
a particularly efficient mechanism, namely the use of the average value, for appropriately
adjusting the parameters which are considered for the provision of the upmix signal
representation.
In a preferred embodiment, the parameter adjuster of the apparatus is configured to provide
the one or more adjusted parameters in dependence on an average value which is a
weighted average of a plurality of parameter values. Using a weighted average provides a
high degree of freedom, because t is possible to allocate different weights to different of
the parameter values. However, allocating identical weights to the parameter values is also
possible.
In a preferred embodiment, the parameter adjuster of the apparatus is configured to provide
the one or more adjusted parameters such that the one or more adjusted parameters deviate
from the average value less than corresponding received parameters. By bringing the
adjusted parameters close to the average value, or by even setting the adjusted parameters
to be equal to the average value, a significant reduction of distortions can be achieved.
In a preferred embodiment, the apparatus is configured to receive one or more rendering
coefficients (also designated as rendering parameters) describing contributions of audio
objects to one or more channels of the upmix signal representation. In this case, the
apparatus is preferably configured to provide one or more adjusted rendering coefficients
as the adjusted parameters. It has been found that adjusting rendering parameters in
dependence on an average value of a plurality of rendering parameters, which serve as
input parameter values, brings along the possibility to obtain well-suited adjusted rendering
parameters, which avoid excessive audible distortions.
In a preferred embodiment, the parameter adjuster is configured to receive, as the input
parameters, a plurality of rendering coefficients. In this case, the parameter adjuster is
configured to compute an average over rendering coefficients associated with a plurality of

audio objects. Also, the parameter adjuster is configured to provide the adjusted rendering
coefficients such that a deviation of an adjusted rendering coefficient from the average
over rendering coefficients associated with a plurality of audio objects is restricted. This
embodiment according to the invention is based on the finding that a distortion of the
upmix signal representation caused by the use of non-optimal rendering parameters is
typically reduced, at least for rendering parameters deviating from optimal rendering
parameters by more than a predetermined deviation, if a deviation of an adjusted rendering
coefficient from the average over rendering coefficients associated with a plurality of audio
objects is restricted. Thus, a simple mechanism, namely the adjustment of the rendering
coefficients such that the deviation of the adjusted rendering coefficients from the average
over rendering coefficients associated with a plurality of audio objects is restricted, allows
to avoid excessive audible distortions.
In a preferred embodiment, the parameter adjuster is configured to leave a rendering
coefficient, which is within a tolerance interval determined in dependence on the average
over the rendering coefficients, unchanged, and to selectively set a rendering coefficient,
which is larger than an upper boundary value of the tolerance interval to a value which is
smaller than or equal to the upper boundary value, and to selectively set a rendering
coefficient, which is smaller than a lower boundary value of the tolerance interval to a
value which is larger than or equal to the lower boundary value. Accordingly, a very
simple mechanism is established for adjusting the rendering coefficients, wherein this
simple mechanism still allows to obtain adjusted rendering coefficients, which avoid an
excessive distortion of the upmix signal representation which would be caused by the use
of non-optimal rendering parameters that are strongly different from the average value.
In a preferred embodiment, the parameter adjuster is configured to iteratively select a
respective one of the rendering coefficients, which comprises a maximum deviation from
the average over the rendering coefficients in the respective iteration, and to bring the
selected one of the rendering coefficients closer to the average over the rendering
coefficients. Accordingly, the rendering parameters which are outside of a tolerance
interval determined in dependence on the average over the rendering coefficients are
iteratively brought into the tolerance interval. Thus, the rendering parameters are adjusted
in dependence on the average value such that a distortion of the upmix signal
representation caused by the use of non-optimal rendering parameters is typically reduced
(at least for input rendering parameters deviating from optimal rendering parameters by
more than a predetermined deviation).

In a preferred embodiment, the parameter adjuster is configured to repeat the iterative
selection of a respective one of the rendering coefficients and the iterative modification of
a selected one of the rendering coefficients until all rendering parameters are adjusted to be
within applicable tolerance intervals. Accordingly, it is ensured that audible distortions in
the upmix signal representation are kept sufficiently small.
In a preferred embodiment, the apparatus is configured to receive one or more transcoding
coefficients describing a mapping of one or more channels of the downmix signal
representation onto one or more channels of the upmix signal representation. In this case,
the apparatus is configured to provide one or more adjusted transcoding coefficients as the
adjusted parameters. This embodiment according to the invention is based on the finding
that transcoding parameters are also well-suited for an adjustment in dependence on an
average value, because large deviations of the transcoding coefficients from the average
value typically cause audible distortions. Accordingly, it is possible to reduce distortions of
the upmix signal representation caused by the use of non-optimal transcoding parameters
(at least for input transcoding parameters deviating from optimal transcoding parameters
by more than a predetermined deviation) by an adjustment or a limitation of the
transcoding parameters in dependence on the average value.
In a preferred embodiment, the parameter adjuster is configured to receive, as the input
parameters, a temporal sequence of transcoding coefficients (also designated as
transcoding parameters). In this case, the parameter adjuster is configured to compute a
temporal mean (also designated as a temporal average) in dependence on a plurality of
transcoding coefficients. Also, the parameter adjuster is configured to provide the adjusted
transcoding coefficients such that a deviation of the adjusted transcoding coefficients from
the temporal mean is restricted. Again, a simple mechanism for avoiding excessive audible
distortions of an upmix signal representation caused by the use of non-optimal transcoding
coefficients is created.
In a preferred embodiment, the parameter adjuster is configured to leave a transcoding
coefficient, which is within a tolerance interval determined in dependence on the temporal
mean (which constitutes the average value) unchanged. Also, the parameter adjuster is
configured to selectively set a transcoding coefficient, which is larger than an upper
boundary value of the tolerance interval, to a value which is smaller than or equal to the
upper boundary value of the tolerance interval, and to selectively set a transcoding
coefficient, which is smaller than a lower boundary value of the tolerance interval, to a
value which is larger than or equal to the lower boundary value. Accordingly, the
transcoding coefficients can be brought into a well-defined tolerance interval, which allows

to reduce distortions of an upmix signal representation caused by the use of non-optimal
transcoding coefficients at least for transcoding coefficients deviating from optimal
transcoding coefficients by more than a predetermined deviation. The tolerance interval is
chosen in an adaptive manner, as the temporal mean is used. This concept is based on the
finding that strong temporal changes of the transcoding coefficients typically bring along
audible distortions and should therefore be limited to some degree.
In a preferred embodiment, the parameter adjuster is configured to calculate the temporal
mean using a recursive low pass filtering of the sequence of transcoding coefficients. This
concept has shown to bring along a very well-defined temporal mean, which takes into
account a long-term evolution of the transcoding coefficients. Also, it has been found that
such a recursive low pass filtering of the sequence of transcoding coefficients can be
effected with little computational effort and memory effort, which helps to reduce the
memory requirements. In particular, it is possible to obtain a meaningful temporal mean
without storing the transcoding coefficient history for an extended period of time.
In a preferred embodiment, the parameter adjuster is configured to provide a given one of
the one or more adjusted parameters such that the given one of the adjusted parameters is
within a tolerance interval, boundaries of which are defined in dependence on the average
value of the plurality of input parameter values and one or more tolerance parameters, and
such that a deviation between an input parameter and a corresponding adjusted parameter
is minimized or kept within a predetermined maximal allowable range. It has been found
that adjusted parameters bringing along a good hearing impression can be obtained by
restricting the adjusted parameters to a tolerance interval while also considering the
objective to avoid excessively large differences between an input parameter and a
corresponding adjusted parameter. Accordingly, a distortion of the upmix signal
representation caused by the use of non-optimal parameters can be reduced without
unnecessarily compromising desired auditory settings defined by the input parameters.
In a preferred embodiment, the parameter adjuster is configured to selectively set an input
parameter, which is found to be outside of the tolerance interval, boundaries of which
tolerance interval are defined in dependence on the average value of the plurality of input
parameter values, to an upper boundary value or a lower boundary value of the tolerance
interval, in order to obtain an adjusted version of the input parameter.
In another preferred embodiment, the parameter adjuster is configured to iteratively select
a respective one of the input parameters, Which comprises a maximum deviation from the
average value in a respective iteration, and to bring the selected one of the input parameters

closer to the average value, in order to iteratively bring input parameters, which are outside
of a tolerance interval (boundaries of which are defined in dependence on the average
value) into the tolerance interval.
In a preferred embodiment, the parameter adjuster is configured to choose a step size used
to bring the selected one of the input parameters closer to the average value to be a
predetermined fraction of a difference between the selected one of the input parameters
and the average value.
Another embodiment according to the invention creates an apparatus for providing an
upmix signal representation on the basis of a downmix signal representation and a
parametric side information. Said apparatus comprises an apparatus for providing one or
more adjusted parameters on the basis of one or more input parameters, as discussed
before. The apparatus for providing an upmix signal representation also comprises a signal
processor configured to obtain the upmix signal representation on the basis of the downmix
signal representation and a parametric side information. The apparatus for providing one or
more adjusted parameters is configured to provide adjusted versions of one or more
processing parameters of the signal processor, for example, of rendering parameters input
to the signal processor or of transcoding parameters computed in the signal processor and
applied by the signal processor to obtain the upmix signal representation.
This embodiment is based on the finding that there is a large number of parameters, which
are applied by the signal processor and either input into the signal processor or even
calculated in the signal processor, and which can benefit from the above-discussed
parameter adjustment on the basis of the average value. It has been found that the signal
processor typically provides a good quality upmix signal representation, with small
distortions, if a set of parameters (for example, a set of rendering coefficients associated
with different audio objects, or a set of transcoding parameter values associated with
different instances in time) is well-balanced, such that the individual values of such a set of
values do not comprise excessively large deviations from an average value. Thus, by
applying the apparatus for providing one or more adjusted parameters in combination with
an apparatus for providing an upmix signal representation, the benefits of the inventive
concept can be realized.
In a preferred embodiment, the signal processor is configured to provide the upmix signal
representation in dependence on adjusted rendering coefficients describing contributions of
audio objects to one or more channels of the upmix signal representation. The apparatus
for providing one or more adjusted parameters is configured to receive a plurality of user-

specified rendering parameters as input parameters and to provide, on the basis thereof,
one or more adjusted rendering parameters for use by the signal processor (preferably to
the signal processor). It has been found that well-balanced rendering parameters, which can
be obtained using the apparatus for providing one or more adjusted parameters, typically
result in a good hearing impression.
In another embodiment, the apparatus for providing the one or more adjusted parameters is
configured to receive one or more mix matrix elements of a mix matrix as the one or more
input parameters, and to provide, on the basis thereof, one or more adjusted mix matrix
elements of the mix matrix for use by the signal processor. In this case, the signal
processor is configured to provide the upmix signal representation in dependence on the
adjusted mix matrix elements of the mix matrix, wherein the mix matrix describes a
mapping of one or more audio channel signals of the downmix signal representation
(represented, for example, in the form of a time domain representation or in the form of a
time-frequency-domain representation) onto one or more audio channel signals of the
upmix signal representation. It has been found that the mix matrix elements should also be
well-adapted to the average value, for example, in that temporal changes of the mix matrix
elements are limited.
In another embodiment according to the invention, the audio processor is configured to
obtain an MPEG surround arbitrary-downmix-gain value. In this case, the apparatus for
providing one or more adjusted parameters is configured to receive a plurality of arbitrary-
downmix-gain values as input parameters, and to provide a plurality of adjusted arbitrary-
downmix-gain values. It has been found that an application of the apparatus for providing
adjusted parameters to arbitrary-downmix-gain values also results in a good hearing
impression and allows to limit audible distortions.
Further embodiments according to the invention create a method and a computer program
for providing one or more adjusted parameters. Said embodiments are based on the same
findings as the above-discussed apparatus and can be extended by any of the features and
functionalities discussed herein with respect to the inventive apparatus.
Brief Description of the Figures
Fig. 1 shows a block schematic diagram of an apparatus for providing one or more
adjusted parameters, according to an embodiment of the invention;

Fig. 2 shows a block schematic diagram of an apparatus for providing an upmix
signal representation, according to an embodiment of the invention;
Fig. 3 shows a block schematic diagram of an apparatus for providing an upmix
signal representation, according to another embodiment of the invention;
Fig. 4 shows a schematic representation of parameter limiting schemes using an
indirect control and a direct control;
Fig. 5 a shows a table representing listening test conditions;
Fig. 5b shows a table representing audio items of listening test;
Fig. 6 shows a table representing tested extreme rendering conditions;
Fig. 7 shows a graphical representation of MUSHRA listening test results for
different parameter limiting schemes (PLS);
Fig. 8 shows a block schematic diagram of a reference MPEG SAOC system;
Fig. 9a shows a block schematic diagram of a reference SAOC system using a
separate decoder and mixer;
Fig. 9b shows a block schematic diagram of a reference SAOC system using an
integrated decoder and mixer;
Fig. 9c shows a block schematic diagram of a reference SAOC system using an
SAOC-to-MPEG transcoder; and
Fig. 10 shows a table describing which transcoding coefficients can be modified by
the proposed parameter limiting scheme.
Detailed Description of the Embodiments
1. Apparatus for providing one or more adjusted parameters, according to Fig. 1
In the following, an apparatus for providing one or more adjusted parameters for a
provision of an upmix signal representation on the basis of a downmix signal

representation and a parametric side information associated with the downmix signal
representation will be described. Fig. 1 shows a block schematic diagram of such an
apparatus 100.
The apparatus 100 is configured to receive one or more input parameters 110 and to
provide, on the basis thereof, one or more adjusted parameters 120. The apparatus 100
comprises a parameter adjuster 130 which is configured to receive the one or more input
parameters 110 and to provide, on the basis thereof, the one or more adjusted parameters
120. The parameter adjuster 130 is configured to provide the one or more adjusted
parameters 120 in dependence on an average value 132 of a plurality of input parameter
values, such that a distortion of an upmix signal representation caused by the use of non-
optimal parameters (for example, the one or more input parameters 110) is reduced at least
for input parameters (for example, input parameters 110) deviating from optimal
parameters by more than a predetermined deviation. For example, the parameter adjuster
130 may have the effect that the one or more adjusted parameters 120 are "closer" (in the
sense of causing smaller distortions) to optimal parameters (which would result in a
distortion-free upmix signal representation) than the one or more input parameters 110.
For this purpose, the parameter adjuster 130 implements an average value computation, to
obtain the average value 132 (for example, as a temporal average or an inter-object
average) of a set of related input parameters 110 (for example, input parameters associated
with a common time interval, or input parameters of the same parameter type associated
with different time instances). Regarding the operation of the apparatus 100, it should be
noted that the provision of the one or more adjusted parameters 120 on the basis of the one
or more input parameters 110 is made in dependence on the average value 132, because it
has been found that the average value 132 is a meaningful quantity for adjusting the
parameters. In particular, it has been found that moderate parameters (with respect to the
average value) typically bring along moderate distortions.
Further details will be described subsequently.
2. Apparatus for providing an upmix signal representation, according to Fig. 2
In the following, an apparatus for providing an upmix signal representation according to
Fig. 2 will be described. Fig. 2 shows a block schematic diagram of such an apparatus 200,
which can be considered as an audio signal decoder. For example, the apparatus 200 may
comprise the functionality of an SAOC decoder or an SAOC transcoder.

The apparatus 200 is configured to receive a downmix signal representation 210 and a
parametric side information 212. Also, the apparatus 200 is configured to receive user-
specified rendering parameters 214. The apparatus is configured to provide an upmix
signal representation 220.
The downmix signal representation 210 may, for example, be a representation of a one-
channel audio signal or of a two-channel audio signal. The downmix signal representation
210 may, for example, be a time domain representation or an encoded representation. In
some embodiments, the downmix signal representation 210 may be a time-frequency-
domain representation, in which the one or more channels of the downmix signal
representation 210 are represented by subsequent sets of spectral values.
The upmix signal representation 220 may, for example, be a representation of individual
audio channels, for example, in the form of a time domain representation or a time-
frequency-domain representation. Alternatively, the upmix signal representation 220 may
be an encoded representation, comprising both a downmix signal representation and a
channel-related side information, for example, an MPEG Surround side information.
The user-specified rendering parameters 214 may be provided in the form of rendering
matrix entries describing desired contributions of a plurality of audio objects to the one or
more channels of the upmix signal representation 220. Alternatively, the user-specified
rendering parameters 214 may be provided in any other appropriate form, for example,
specifying a desired rendering position and rendering volume of the audio objects.
The apparatus 200 comprises a signal processor 230, which is configured to provide the
upmix signal representation 220 on the basis of the downmix signal representation 210 and
the parametric side information 212. The signal processor 230 comprises a remixing
functionality 232 in order to provide the upmix signal representation 220 on the basis of
the downmix signal representation 210. For example, the remixing functionality 232 may
be configured to linearly combine a plurality of channels of the downmix signal
representation 212 in order to obtain the one or more channels of the upmix signal
representation 220. In this remixing, contributions of the channels of the downmix signal
representation 210 to the channels of the upmix signal representation 220 may be
determined by mix matrix elements of a mix matrix G, wherein a first dimension (for
example, a number of rows) of the mix matrix G may be determined by the number of
channels of the upmix signal representation 220, and wherein a second dimension (for
example, a number of columns) of the mix matrix G may be determined by a number of
channels of the downmix signal representation 210.

For example, the remixing process 232 may be used to provide one or more vectors
comprising spectral values associated with one or more channels of the upmix signal
representation 220 by multiplying one or more vectors comprising spectral values of one or
more channels of the downmix signal representation 210 with the mix matrix G.
The signal processor 230 may also comprise a mixing parameter computation 236 which
provides the mix matrix G (or equivalently, the elements thereof). The mix matrix
elements are determined in dependence on the parametric side information 212 and
modified rendering parameters 252 by the mixing parameter computation 236. The mix
matrix elements of the mix matrix G are, for example, provided such that the one or more
channels of the upmix signal representation 220 describe audio objects, which are
represented by the one or more channels of the downmix signal representation 210, in
accordance with the modified rendering parameters 252. For this purpose, the parametric
side information 212 is evaluated by the mixing parameter computation 236, wherein the
parametric side information 212 comprises, for example, an object-level difference
information OLD, an inter-object-correlation information IOC, a downmix gain
information DMG and (optionally) a downmix-channel-level-difference information
DCLD. The object-level difference information may describe, for example, in a frequency-
band-wise manner, level differences between a plurality of audio objects. Similarly, the
inter-object-correlation information may describe, for example, in a frequency-band-wise
manner, correlations between a plurality of audio objects. The downmix-gain information
and the (optional) downmix-channel-level-difference information may describe the
downmix, which is performed to combine audio object signals from a plurality of audio
objects into the one or more channels of the downmix signal representation, wherein there
are typically more audio objects than channels of the downmix signal representation 210.
Accordingly, the mixing parameter computation 236 may evaluate how the mix matrix
elements should be chosen in order to obtain an upmix signal representation 220
comprising expected statistic properties on the basis of the parametric side information 212
and the modified rendering parameters 252.
The signal processor 230 may optionally comprise a side information modification or side
information transformation 240, which is configured to receive the parametric side
information 212 and to provide a modified side information (for example, an MPEG
Surround side information), such that the modified side information and the associated
remixed downmix signal representation provided by the remixing process 232 describe a
desired audio scene.

To summarize, the signal processor 230 may, for example, fulfill the functionality of the
SAOC decoder 820, wherein the downmix signal representation 210 takes the role of the
one or more downmix signals 812, wherein the parametric side information 212 takes the
role of the side information 814, and wherein the upmix signal representation 220 is
equivalent to the output channel signals ŷ1 to ŷM
Alternatively, the signal processor 230 may comprise the functionality of the separate
decoder and mixer 920, wherein the downmix signal representation 210 may take the role
of the one or more downmix signals, wherein the parametric side information 212 may
take the role of the object meta data, and wherein the upmix signal representation 220 may
take the role of the one or more output channel signals 928.
Alternatively, the signal processor 230 may comprise the functionality of the integrated
decoder and mixer 950, wherein the downmix signal representation 210 may take the role
of the one or more downmix signals, wherein the parametric side information 212 may
take the role of the object meta data, and wherein the upmix signal representation 220 may
take the role of the one or more output channel signals 958.
Alternatively, the signal processor 230 may comprise the functionality of the SAOC-to-
MPEG surround transcoder 980, wherein the downmix signal representation 210 may take
the role of the one or more downmix signals, wherein the parametric side information 212
may take the role of the object meta data, and wherein the upmix signal representation may
be equivalent to the one or more downmix signals 988 when taken in combination with the
MPEG surround bitstream 984.
In any case, the modified rendering parameters 252 may take the role of the user
interaction/control information 822 or of the rendering information.
The apparatus 200 also comprises an apparatus 250 for providing adjusted rendering
parameters. The apparatus 250 for providing the adjusted rendering parameters receives the
user-specified rendering parameters 214 and provides, on the basis thereof, the modified
rendering parameters 252. The apparatus 250 is typically configured to calculate an
average value over a plurality of user-specified rendering parameters associated with
different audio objects, to obtain an average value. Also, the apparatus 250 is configured to
perform a rendering parameter limitation in dependence on the average value, to obtain the
modified rendering parameters 252 by limiting the user-specified rendering parameters
214. A tolerance interval, to which the modified rendering parameters 252 are limited, is

typically determined in dependence on the average value, such that strong deviations of the
modified rendering parameters 252 from the average value are avoided, even if one or
more of the user-specified rendering parameters 214 comprises such a strong deviation
from the average value. In this manner, excessive distortions within the upmix signal
representation 220 are typically avoided, because the modified rendering parameters 252,
which comprise limited inter-object deviation, will result in an upmix signal representation
with low-distortions, while a large difference between rendering parameters associated
with different audio objects would typically result in audible artifacts.
It should be noted here that the apparatus 250 for providing adjusted rendering coefficients
may comprise the same overall functionality as apparatus 100 for providing one or more
adjusted parameters, wherein the user-specified rendering parameters 214 may take the
role of one or more input parameters 110, and wherein the adjusted rendering parameters
252 may take the role of the one or more adjusted parameters 120.
Details regarding the provision of the modified rendering parameters 252 will be discussed
below, taking reference to Fig. 4.
3. Apparatus for providing an upmix signal representation, according to Fig. 3
In the following, an apparatus for providing an upmix signal representation according to
another embodiment of the invention will be described taking reference to Fig. 3, which
shows a block schematic diagram of such an apparatus 300.
The apparatus 300 typically receives the same type of input signals and provides the same
type of output signals as the apparatus 200, such that identical reference numerals are used
herein to describe identical or equivalent signals. To summarize, the apparatus 300
receives a downmix signal representation 210, parametric side information 212 and user-
specified rendering parameters 214, and the apparatus 300 provides, on the basis thereof,
an upmix signal representation 220.
The apparatus 300 comprises a signal processor 330, which may be substantially
equivalent in the functionality to the signal processor 230. The signal processor 330
comprises a remixing functionality 332,which is identical to the remixing functionality 232
of the signal processor 230 in that it provides remixed audio channel signals on the basis of
the downmix signal representation. However, the remixing 332 uses an adjusted mix
matrix, rather than a mix matrix obtained directly from a mixing parameter computation.

The signal processor 330 also comprises a mixing parameter computation 336, which may
be identical in function to the mixing parameter computation 236 of the signal processor
230. Accordingly, the mixing parameter computation 336 receives the parametric side
information 212 and the user-specified rendering parameters 214, and provides, on the
basis thereof, a mix matrix G (or equivalently, mix matrix elements of the mix matrix G,
which are also designated with 337).
The signal processor 330 optionally also comprises a side information modification 338,
the functionality of which is identical to the side information modification 240.
In addition, the apparatus 300 comprises an apparatus 350 for providing adjusted mix
matrix elements. The apparatus 350 may or may not be part of the signal processor 330.
The apparatus 350 is configured to receive the mix matrix 337, G (or, equivalently, the mix
matrix elements thereof), which are provided by the mixing parameter computation 336,
and to provide, on the basis thereof, an adjusted mix matrix 352 G' (or, equivalently,
adjusted mix matrix elements thereof). For example, one set of mix matrix elements and
one set of adjusted mix matrix elements may be provided per frequency band and per audio
frame. In other words, the mix matrix G and the modified mix matrix G' may be updated
once per audio frame of the downmix signal representation 210, if a frame-wise processing
is chosen. However, the update interval may be different in some cases. Also, it is not
necessary that there are multiple mix matrices and adjusted mix matrices G, G' for
different frequency bands.
However, the apparatus 350 is configured to provide adjusted mix matrix elements of the
adjusted mix matrix 352 on the basis of the mix matrix elements of the mix matrix 337
provided by the mixing parameter computation 336. For example, the processing may be
performed individually per position of the mix matrix (or adjusted mix matrix), such that a
sequence of adjusted mix matrix elements of a given mix matrix position may be
dependent on a sequence of mix matrix elements of the mix matrix 337 at the same mix
matrix position, but independent from mix matrix elements at different mix matrix
positions.
The apparatus 350 for providing an adjusted mix matrix element is configured to provide
the one or more adjusted mix matrix elements of the adjusted mix matrix 352 in
dependence on one or more average values (for example, one or more matrix-position-
individual average values) computed on the basis of the mix matrix 337. The apparatus 350
for providing the adjusted mix matrix elements of the adjusted mix matrix 352 is

preferably configured to calculate an average value of mix matrix elements at a given mix
matrix position over time. Thus, for a given mix matrix position, an average value
(preferably, but not necessarily, a temporal average value, like, for example, a floating
average or a quasi-infinite-impulse-response average value or an average value obtained by
a recursive low pass filtering or similar mathematical operations well-known for time
averaging) may be computed on the basis of a sequence of mix matrix elements of the
given mix matrix position. For example, a sequence of mix matrix elements describing a
contribution of a given channel of the downmix signal representation 210 onto a given
channel of the upmix signal representation 220, which mix matrix elements are associated
with a plurality of audio frames, may be used in order to obtain such an average value (also
designates as mean value), which average value may be a finite-impulse-response average
value or a (quasi) infinite-impulse-response average value (obtained, for example, using a
recursive low pass filtering or similar mathematical operations well-known for time
averaging). A current adjusted mix matrix element of the given mix matrix position
(describing the contribution of the given channel of the downmix signal representation 210
onto the given channel of the upmix signal representation 220) may be limited by the
apparatus 350 to a tolerance interval which is defined in dependence on the average value
associated to the given mix matrix position.
Accordingly, excessive temporal fluctuations of mix matrix elements are avoided, because
adjusted mix matrix elements are restricted to a tolerance interval which is determined, for
example, by an average (finite-impulse-response average or infinite-impulse-response
average) of previous mix matrix elements at the same mix matrix position. It has been
found that such a restriction of the adjusted mix matrix elements of the adjusted mix matrix
352 typically brings along a limitation of the distortions of the upmix signal 220 caused by
the use of non-optimal parameters (for example non-optimal user-specified rendering
parameters) at least if the non-optimal user-specified rendering parameters deviate from
optimal user-specified rendering parameters by more than a predetermined deviation.
It should be noted here that the apparatus 350 for providing adjusted mix matrix elements
may comprise the same overall functionality as apparatus 100 for providing one or more
adjusted parameters, wherein the mix matrix elements of the mix matrix 337 may take the
role of one or more input parameters 110, and wherein the adjusted mix matrix elements of
the adjusted mix matrix 352 may take the role of the one or more adjusted parameters 120.
4. Parameter limiting schemes according to Fig. 4

In the following, parameter limiting schemes according to the invention will be described
taking reference to Fig. 4, which shows a schematic representation of such parameter
limiting schemes.
Fig. 4 shows the application of parameter limiting schemes in combination with an SAOC
decoder 410. However, the parameter limiting schemes may be applied in combination
with different types of audio decoders or audio transcoders, like, for example, an SAOC
transcoder.
SAOC decoder 410 receives a downmix 420 and an SAOC bitstream 422. Also, the SAOC
decoder provides one or more output channels 430a to 430M.
In a first implementation, designated with (a), the parameter limiting scheme 440
implements an indirect control. The parameter limiting scheme 440 receives an input
rendering matrix R, for example, a user specified rendering matrix, and provides, on the
basis thereof, an adjusted rendering matrix R to the SAOC decoder. In this case, the
SAOC decoder uses the adjusted rendering matrix R for a derivation of the mix matrix G,
as described above. The parameter limiting scheme 440 may also receive parameters AR.,
AR+, which may determine boundaries of a tolerance interval.
Alternatively, or in addition, a second parameter limiting scheme 450 may be applied. The
second parameter limiting scheme receives transcoding parameters T and provides, on the
basis thereof, adjusted transcoding parameters T . The transcoding parameters T may be
computed in the SAOC decoder 410, and the adjusted transcoding parameters 7* may be
applied by the SAOC decoder 410. For example, the transcoding parameters T may be
equivalent to the mix matrix elements of the mix matrix G, as discussed before, and the
adjusted transcoding parameters f may be equivalent to the adjusted mix matrix elements
of the adjusted mix matrix G'.
The parameter limiting scheme 450 may receive one or more parameters AT-, AT+, which
parameters may determine boundaries of tolerance intervals.
4.1 Overview
In the following, an overview will be given over the parameter limiting scheme for
distortion control.

The general SAOC processing is carried out in a time/frequency selective way and will be
described in the following.
The SAOC encoder extracts the psychoacoustic characteristics (for example, object power
relations and correlations) of several input audio object signals and then downmixes them
into a combined mono or stereo channel (which may be designated, for example, as a
downmix signal representation). This downmix signal and extracted side information are
transmitted (or stored) in compressed format using the well-known perceptual audio
coders. On the receiving end, the SAOC decoder conceptually tries to restore the original
object signal (i.e., separate downmixed objects) using the transmitted side information (for
example, object-level-difference information OLD, inter-object-correlation information
IOC, downmix-gain information DMG and downmix-channel-level-difference information
DCLD). These approximated object signals are then mixed into a target scene using a
rendering matrix (wherein the rendering matrix typically describes contributions of
different audio objects to different channels of the upmix signal representation). The
rendering matrix is composed of the relative rendering coefficients RCs (or object gains)
specified for each transmitted audio object and upmix setup loudspeaker. These object
gains determine the spatial position of all separated/rendered objects. Effectively, the
separation of the object signals is rarely executed (or even never executed) since the
separation and the mixing is performed in a single combined processing step, which results
in an enormous reduction of computational complexity. The single combined processing
step may, for example, be performed using transcoding coefficients, which describe the
combination of the object separation and mixing of the separated objects.
It has been found that this scheme is tremendously efficient, both in terms of transmission
bitrate (it is only required to transmit one or two downmix channels plus some side
information instead of a number of individual object audio signals) and computational
complexity (the processing complexity relates mainly to the number of output channels
rather than the number of audio objects).
The SAOC decoder transforms (on a parametric level) the object gains and other side
information directly into the transcoding coefficients (TCs) which are applied to the
downmix signal to create the corresponding signals for the rendered output audio scene (or
a preprocessed downmix signal for a further decoding operation, i.e. typically multi-
channel MPEG Surround rendering).
It has been found that the subjectively perceived audio quality of the rendered output scene
can be improved by application of distortion control measures or DCMs, as described in

non-pre-published US 61/173,456. This improvement can be achieved for the price of
accepting a moderate dynamic modification of the target rendering settings. The
modification of the rendering information has time and frequency variant nature which
under specific circumstances may result in unnatural sound colorations and temporal
fluctuation artifacts.
In an alternative to the distortion control measures (DCMs) described in reference [6],
embodiments according to the present invention use a number of parameter limiting
schemes which focus on the reduction of audio artifacts (sound colorations, temporal
fluctuations, etc.) and at the same time preserving a natural sound quality.
The proposed parameter limiting scheme concepts described herein do not adjust rendering
coefficients (RCs) based on a distortion measure calculated using sophisticated algorithms
based on psychoacoustic models. Instead, the proposed parameter limiting scheme
concepts show a low computational and structural complexity and are therefore attractive
for integration into SAOC technology. Nevertheless, they can also be advantageously
combined with the schemes described in reference [6] in order to achieve better overall
output quality by complementing each other.
Within the overall SAOC system, the parameter limiting schemes can be incorporated into
the SAOC decoder processing chain in two ways. For example, that parameter limiting
scheme can be placed at the front-end for indirect (external) modification of the SAOC
output by controlling the rendering coefficients (RCs) R, which is shown as alternative (a)
in Fig. 4. Alternatively, the inherent transcoding coefficients (TCs) T are directly
(internally) modified at the back-end of the SAOC decoder, before the coefficients are
applied to the downmix signal to yield the output upmix channel signals, which is shown
as the alternative (b) of Fig. 4.
4.2. Indirect control
In the following, the concept of indirect control will be discussed in more detail.
The underlying hypothesis of the indirect control method considers a relationship between
distortion level and deviations of the RCs from their object-averaged value. This is based
on the observation that the more specific attenuation/boosting is applied by the RCs to a
particular object with respect to the other objects, the more aggressive modification of the
transmitted downmix signal is to be performed by the SAOC decoder/transcoder. In other
words: the higher the deviation of the "object gain" values are relative to each other, the

higher the chance for unacceptable distortion to occur (assuming identical downmix
coefficients). It has been found that this can be tested by examining the deviation of the
RCs from the average of the RCs across all objects (e.g. mean rendering value).
Without loss of generality, the subsequent description is based on the configuration
considering a mono downmix with unity downmix gains for all objects. For the case of
nontrivial downmixes (with different and/or dynamic object gains) the algorithm can be
appropriately modified. In addition, the RCs are assumed to be frequency invariant to
simplify the notation.
Based on the user specified rendering scenario represented by the coefficients R(i) with
object index i, the PLS prevents extreme rendering values by producing modified RC
values R(f) that are actually used by the SAOC rendering engine. They can be derived as
the following function

where A is a PLS control parameter (i.e. threshold value). The PLS control parameter may
be considered as a tolerance parameter.
The deviation Rd (/') of rendering coefficient R(i) from an averaged rendering value R
(e.g. the arithmetic mean) can be obtained as

where


Accordingly, is a ratio between a rendering coefficient and an averaged
rendering value The averaged rendering value is an average value, averaged over
the audio objects having audio object indices i, of the rendering coefficients
The limited deviation is restricted to a certain tolerance A range as

Note that this corresponds to an RC limiting operation which is carried out relative to a
reference value, for example R which is computed dynamically from the input RCs rather
than a specific pre-defined value.
R(i)
For the described PLS approach the optimal solution can be formulated as a minimization
problem for which the difference between given RC R(i) and modified (limited)
value is minimized

In the following, some algorithmic solutions for providing the adjusted rendering
coefficients will be described, wherein the adjusted rendering coefficients can
be considered as adjusted parameters.
The following two algorithmic solutions are based on the deviation of those rendering
values which lie outside the tolerance range, i.e.

4.2.1 One-step solution
A simple and fast one-step solution can be employed to limit all rendering values outside
the tolerance range by


In contrast, the rendering values inside the tolerance range may be left unaffected, such
that

for such rendering values
4.2.2 Iterative solution
Another straightforward method can be employed in which the out-of-range rendering
values with associated deviations are limited gradually. In each iteration of this
algorithm, the maximal rendering deviation is defined as

The corresponding rendering coefficient is restricted such that

This processing can be performed until all values are inside the tolerance region or with a
pre-determined number of iterations.
Accordingly, in each iteration, a rendering coefficient is selected for which the
deviation (for example, from the average value takes the maximum value
In other words, the rendering coefficient is selected, which comprises a
maximum deviation (in terms of the deviation value from the average over the

rendering coefficients in the respective iteration. In addition, the selected rendering
coefficient R(imax) is brought closer to the average over the rendering coefficients using
the above mentioned linear combination of R(i) and (which may be applied selectively
for i = imax). In each step of the iterative procedure, a new selection of the rendering
coefficient having the maximum deviation from the average value may be performed, such
that different rendering coefficients may be modified in different steps of the iterative
algorithm. In other words, imax is typically updated in every iteration. Also, the average
value may optionally be recomputed for every step of the iterative algorithm, considering a
previously modified rendering coefficient.
4.3 Direct Control
The underlying hypothesis of the direct control method considers a relationship between
distortion level and deviations of the TCs from their time-averaged value. This is based on
the observation that the more specific attenuation/boosting is applied to a particular object
with respect to the other objects, the more aggressive modification of the transmitted
downmix signal by the TCs is to be performed by the SAOC decoder/transcoder. In other
words: if the value of a TC is unusually large, it can be concluded that the SAOC algorithm
attempts to modify an object signal with small power into an output dominated by other
object signal(s) with a large power by applying a strong boost. Conversely, if a TC is
unusually small, it can be concluded that the SAOC algorithm attempts to modify an object
signal with large power into an output dominated by other object signal(s) with a small
power by applying a strong attenuation. In both cases, there is a high risk of producing an
unacceptably low signal quality at the SAOC output. Thus, the central idea is to prevent
large deviations of TCs from an average value.
This PLS can be considered as time and frequency variant, since it includes all
dependencies on the SAOC signal parameters (e.g. OLD, IOC) and heuristic elements of
the transcoding/decoding process.
Without loss of generality, the subsequent description is based on the configuration
considering a mono upmix.
Based on the SAOC output TC T[k) with frequency index k, the PLS prevents extreme
values of the TCs by replacing them (e.g., transcoding coefficients outside of a tolerance
interval) with modified TC values which are then used by the actual SAOC rendering
process. The modified TC values T(k) can be derived with the following function


where A is a PLS control parameter (i.e. threshold value). The PLS control parameter may
be considered as a tolerance parameter.
Since the TCs are time-variant, a recursive low pass filter is applied to calculate the mean

The mean is considered as an average value, wherein a weighting of the individual
transcoding values is introduced by the application of the recursive low pass filtering.
Here, n represents the time index of TCs and is the averaging parameter. The
tolerance range for the modified TC value is defined as

Note that this corresponds to a TC limiting operation which is carried out relative to a
reference value which is computed dynamically from the TCs rather than a specific pre-
defined value.
For the described PLS approach the optimal solution can be formulated as minimization
problem for which the difference between given TC T(k) and modified (limited) TC
value is minimized

In the following, a possible solution algorithm for this problem will be described.
4.3.1 Solution algorithm
The modified TC value can be obtained as



4.3.2 Examples of transcoding coefficients
The above discussed parameter limiting scheme for transcoding coefficients can be applied
to different transcoding coefficients which are used, for example, in the SAOC decoders
and transcoders discussed above.
For example, the parameter limiting scheme for transcoding coefficients can be applied to
limit parameters of the mix matrix G, which is used in the signal processor 330 of the
apparatus 300. In this case, a mix matrix element at a given matrix position of the matrix G
may take the place of a transcoding coefficient T(k), wherein k is a frequency index. A
corresponding mix matrix element of the mix matrix G' may correspond to an adjusted
transcoding coefficient The transcoding parameter limiting scheme may be applied,
for example, individually to the different matrix positions of the mix matrix. For example,
if the mix matrix G comprises mix matrix elements g11, g12, g21 and g22, and the adjusted
mix matrix G' comprises corresponding matrix elements g11; g12 g21' and g22', the
adjusted mix matrix element gn'(n0) may be derived from a sequence g11(l) to g11(n0).
Equivalent derivations may be used for the other mix matrix elements g12', g21' and g22' of
the adjusted mix matrix G'.
The table of Fig. 10 provides a list of transcoding coefficients which can be modified, for
example, limited, by the proposed parameter limiting schemes for all SAOC modes of
operation. The table of Fig. 10 shows, in a first column 1010, different SAOC modes. The
table of Fig. 10 further shows, in a second column 1020, which parameters can be modified
(for example, limited) by the proposed parameter limiting scheme. A third column 1030
shows a reference to the corresponding subclauses of the MPEG SAOC FCD document of
reference [8]. To summarize, the table of Fig. 10 shows a list of transcoding coefficients
which can be modified (for example, limited) by the proposed parameter limiting schemes
for all SAOC modes of operation with references to corresponding subclauses of the
MPEG SAOC FCD document [8],
4.4 Generalized formulation of the parameter limiting scheme for limited relative
deviation

There exists a generalized formulation for the above-discussed PLS. This formulation can
be expressed in the form of the following minimization problem for the general parameter
variable Xl as

Here, the value of X, is initially given and the "reference" value . can be estimated as a
function of the modified variable as
In the above, the parameter variable X, may, for example, be identical to R{i)ox T(i).
Similarly, the adjusted parameter variable may be identical to the adjusted rendering
coefficient or the adjusted transcoding coefficient . The variables may
also, for example, be equivalent to mix matrix elements gmn(i) and gmn'(0-
In the following, two solution algorithms will be discussed.
Generally, the analytical approaches for obtaining the exact solution of such minimization
problems are computationally demanding. Nevertheless, there exist simple and fast
alternative ways providing suboptimal results which are still suitable for the PLS purposes.
Two such simple approaches are described here.
4.4.1 One-step solution
The one-step solution based on assumption that .
limits all values outside the tolerance range to lie inside it by



Values which lie inside the tolerance range (which may be considered as a tolerance
interval) may, for example, be left unchanged.
4.4.2 Iterative solution
The iterative solution modifies in each step one selected out-of-range value

For instance, the processing index / * can be chosen using the condition:

The number of iterations can be set to a certain value or implicitly derived from the
algorithm.
One should note that all these methods can be applied for limiting RCs and TCs as
described above
4.5 Generalized linear formulation
There exists a generalized linear formulation for the above-discussed PLS. In the previous
section the deviation of the general parameter is described as a ratio In contrast, it
can also be defined as leading to the following minimization problem for the
general parameter variable as


Here, the value of is initially given and the "reference" value . can be estimated as a
function of the modified variable as
In the following, two solution algorithms for this problem will be described.
Generally, the analytical approaches for obtaining the exact solution of such minimization
problems are generally computationally demanding. Nevertheless, there exist simple and
fast alternative ways providing suboptimal results which are still suitable for the PLS
purposes. Two such simple approaches are described here:
4.5.1 One-step solution
The one-step solution based on assumption that limits all values outside the
tolerance range to lie inside it by

4.5.2 Iterative solution
The iterative solution modifies in each step one selected value is outside
a tolerance range:

For instance, the processing index can be chosen using the condition:
and the modification step size value as with
The number of iterations can be set to a certain value or implicitly derived from
the algorithm.
This algorithm provides a flexible way of using the tolerance range, i.e. it is dynamically
changing (depending on Xi*).

One should note that all these methods can be applied for limiting RCs and TCs as
described above.
Alternatively, the following algorithm can be used:

This version of the algorithm uses a fix (static) tolerance range
4.6 Further remarks
One should note that all these methods can be applied for limiting rendering coefficients
and transcoding coefficients, as described above.
5. Application of parameter limiting schemes to multichannel downmix/upmix
scenarios
The single TC PLS (e.g. direct control) of a mono downmix/mono upmix scenario extends
to a TC matrix considering any combination of downmix/upmix channels. Consequently,
the direct control can be applied to each TC individually. The multichannel upmix scenario
for the RC PLS (e.g. indirect control) can be realized, for instance, in a simple multiple-
mono approach where all individual rendering coefficients are handled independently.
6. Listening test results
6.1 Test design and items

The subjective listening test has been conducted to assess the perceptual performance of
the proposed distortion control measure (DCM) concepts and compare it to the regular
SAOC reference model (SAOC RM) decoding processing.
The test design includes the cases of individual application of the direct and indirect
control approaches of the proposed parameter limiting scheme as well as their
combination. The output signal of the regular (unprocessed by the parameter limiting
scheme PLS) SAOC decoder is included in the test to demonstrate the baseline
performance of the SAOC. In addition, the case of trivial rendering, which corresponds to
the downmix signal, is used in the listening test for comparison purposes.
The table of Fig. 5a describes listening test conditions.
The four items representing typical and most critical artifact types for the extreme
rendering conditions have been chosen for the current listening test from the call-for-
proposals (CfP) listening test material.
The table of Fig. 5b describes audio items of the listening test.
The rendering object gains according to the table of Fig. 6 have been applied for the
considered upmix scenarios.
Since the proposed PLS operates using the regular SAOC bitstreams and downmixes (no
any PLS related activity on SAOC encoder side is needed) and does not relay on residual
information, no core coder has been applied to the corresponding SAOC downmix signals.
For all test items and considered rendering conditions the global settings for the PLS are
taken as

6.2 Test methodology

The subjective listening tests were conducted in an acoustically isolated listening room that
is designed to permit high-quality listening. The playback was done using headphones
(STAX SR Lambda Pro with Lake-People D/A-Converter and STAX SRM-Monitor).
The test method followed the procedure used in the spatial audio verification tests, based
on the "Multiple Stimulus with Hidden Reference and Anchors" (MUSHRA) method for
the subjective assessment of intermediate quality audio [7], The test method has been
accordantly modified in order to assess the perceptual performance of the proposed DCM
concepts. In accordance with the adopted test methodology, the listeners were instructed to
compare all test conditions against each other according to the following listening test
instructions:
For each audio item please:
• first read the description of the desired sound mixes that you as a system user would
like to achieve:
Item "BlackCoffee": Soft horn section sound within the sound mix
Item "Fanta4": Strong drum sound within the sound mix
Item "LovePop": Soft string section sound within the sound mix
Item "Audition": Soft music and strong vocal sound
• then grade the signals using one common grade to describe both
- achieving the objective of the desired sound mix
- overall scene sound quality (consider distortions, artifacts, unnaturalness...)
A total of 9 listeners participated in each of the performed tests. All subjects can be
considered as experienced listeners. The test conditions were randomized automatically for
each test item and for each listener. The subjective responses were recorded by a
computer-based MUSHRA program on a scale ranging from 0 to 100. An instantaneous
switching between the items under test was allowed.
6.3 Listening test results
A short overview in terms of the diagrams demonstrating the obtained listening test results
can be found in the appendix. These plots show the average MUSHRA grading per item

over all listeners and the statistical mean value over all evaluated items together with the
associated 95% confidence intervals.
The following observations can be made based upon the results of the conducted listening
tests: For all conducted listening tests the obtained MUSHRA scores prove that the
proposed PLS functionality provides better performance in comparison with the regular
SAOC RM system in sense of overall statistical mean values. One should note that the
quality of all items produced by the regular SAOC decoder (showing strong audio artifacts
for the considered extreme rendering conditions) is graded just slightly higher in
comparison to the quality of downmix-identical rendering settings which does not fulfill
the desired rendering scenario at all. Hence, it can be concluded that the proposed PLS lead
to considerable improvement of subjective signal quality for all considered listening test
scenarios. It can be also concluded that the most promising limiting system consists of a
combination of both RC and TC PLS.
Details regarding the listening test results can be seen in the graphic representation of Fig.
7.
7. Implementation Alternatives
Although some aspects have been described in the context of an apparatus, it is clear that
these aspects also represent a description of the corresponding method, where a block or
device corresponds to a method step or a feature of a method step. Analogously, aspects
described in the context of a method step also represent a description of a corresponding
block or item or feature of a corresponding apparatus. Some or all of the method steps may
be executed by (or using) a hardware apparatus, like for example, a microprocessor, a
programmable computer or an electronic circuit. In some embodiments, some one or more
of the most important method steps may be executed by such an apparatus.
The inventive encoded audio signal can be stored on a digital storage medium or can be
transmitted on a transmission medium such as a wireless transmission medium or a wired
transmission medium such as the Internet.
Depending on certain implementation requirements, embodiments of the invention can be
implemented in hardware or in software. The implementation can be performed using a
digital storage medium, for example a floppy disk, a DVD, a Blue-Ray, a CD, a ROM, a
PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable
control signals stored thereon, which cooperate (or are capable of cooperating) with a

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

In some embodiments, a programmable logic device (for example a field programmable
gate array) may be used to perform some or all of the functionalities of the methods
described herein. In some embodiments, a field programmable gate array may cooperate
with a microprocessor in order to perform one of the methods described herein. Generally,
the methods are preferably performed by any hardware apparatus.
The above described embodiments are merely illustrative for the principles of the present
invention. It is understood that modifications and variations of the arrangements and the
details described herein will be apparent to others skilled in the art. It is the intent,
therefore, to be limited only by the scope of the impending patent claims and not by the
specific details presented by way of description and explanation of the embodiments
herein.
8. Conclusions
Embodiments according to the invention create parameter limiting schemes for distortion
control in audio decoders. Some embodiments according to the invention are focused on
spatial audio object coding (SAOC), which provides means for a user interface for a
selection of the desired playback setup (for example, mono, stereo, 5.1, etc.) and
interactive real-time modification of the desired output rendering scene by controlling the
rendering matrix according to a personal preference or other criteria. However, it is a
straightforward task to adapt the proposed method for parametric techniques in general.
Due to the downmix/separation/mix-based parametric approach, the subjective quality of
the rendered audio output depends on the rendering parameter settings. The freedom of
selecting rendering settings of the users choice entails the risk of the user selecting
inappropriate object rendering options, such as extreme gain manipulations of an object
within the overall sound scene.
For a commercial product it is by all means unacceptable to produce bad sound quality
and/or audio artifacts for any settings on the user interface. In order to control excessive
deterioration of the produced SAOC audio output, several computational measures have
been described which are based on the idea of computing a measure of perceptual quality
of the rendered scene, and depending on this measure (and other information), modify the
actually applied rendering coefficients (see, for example, reference [6]).
The present invention creates alternative ideas for safeguarding the subjective sound
quality of the rendered SAOC scene

• for which all processing is carried out entirely within the SAOC decoder/transcoder,
and
• which do not involve the explicit calculation of sophisticated measures of perceived
audio quality of the rendered sound scene.
These ideas can thus be implemented in a structurally simple and extremely efficient way
within the SAOC decoder/transcoder framework. Since the proposed distortion control
mechanisms (DCMs) aim at limiting parameters inherent to the SAOC decoder, namely,
the rendering coefficients (RCs) and the transcoding coefficients (TCs), they are called
parameter limiting schemes (PLS) throughout the present description.
However, the parameter limiting schemes can be applied to any different audio decoders as
well.

9. References
[1] C. Faller and F. Baumgarte, "Binaural Cue Coding - Part II: Schemes and
applications", IEEE Trans, on Speech and Audio Proa, vol. 11, no. 6, Nov. 2003.
[2] C. Faller, "Parametric Joint-Coding of Audio Sources", 120th AES
Convention, Paris, 2006, Preprint 6752.
[3] J. Herre, S. Disch, J. Hilpert, O. Hellmuth: "From SAC To SAOC - Recent
Developments in Parametric Coding of Spatial Audio", 22nd Regional UK AES
Conference, Cambridge, UK, April 2007.
[4] J. Engdegard, B. Resch, C. Falch, O. Hellmuth, J. Hilpert, A. Holzer, L.
Terentiev, J. Breebaart, J. Koppens, E. Schuijers and W. Oomen: "Spatial Audio
Object Coding (SAOC) - The Upcoming MPEG Standard on Parametric Object
Based Audio Coding", 124th AES Convention, Amsterdam 2008, Preprint 7377.
[5] ISO/IEC, "MPEG audio technologies - Part 2: Spatial Audio Object Coding
(SAOC)," ISO/IEC JTC1/SC29/WG11 (MPEG) FCD 23003-2.
[6] US patent application 61/173,456, METHODS, APPARATUS, AND COMPUTER
PROGRAMS FOR DISTORTION AVOIDING AUDIO SIGNAL PROCESSING
[7] EBU Technical recommendation: "MUSHRA-EBU Method for Subjective Listening
Tests of Intermediate Audio Quality", Doc. B/AIM022, October 1999.
[8] ISO/IEC JTC1/SC29/WG11 (MPEG), Document N10843, "Study on ISO/IEC
23003-2:200x Spatial Audio Object Coding (SAOC)", 89th MPEG Meeting,
London, UK, July 2009

Claims
1. An apparatus (100; 250; 350; 440; 450) for providing one or more adjusted
parameters (120; 252; 352; for a provision of an upmix signal
representation (220; 430a-430M) on the basis of a downmix signal representation
(210; 420) and a parametric side information (212; 422) associated with the
downmix signal representation, the apparatus comprising:
a parameter adjuster configured to receive one or more parameters (110; 214; 337)
and to provide, on the basis thereof, one or more adjusted parameters (120; 252;
352), wherein the parameter adjuster is configured to provide the one or more
adjusted parameters in dependence on an average value (132; of a plurality
of parameter values (110; 214; 337; R; T), such that a distortion of the upmix signal
representation caused by the use of non-optimal parameters for the provision of the
upmix signal representation is reduced at least for one or more parameters deviating
from optimal parameters by more than a predetermined deviation.
2. The apparatus (100; 250; 350; 440; 450) according to claim 1, wherein the
parameter adjuster is configured to provide the one or more adjusted parameters in
dependence on an average value which is a weighted average of a plurality of
parameter values.
3. The apparatus (100; 250; 350; 440; 450) according to claim 1 or 2, wherein the
parameter adjuster is configured to provide the one or more adjusted parameters
such that the orie or more adjusted parameters deviate from the average value less
than corresponding received parameters.
4. The apparatus (100; 250; 440) according to one of claims 1 to 3, wherein the
apparatus is configured to receive one or more rendering coefficients (214; R)
describing desired contributions of audio objects to one or more channels of the
upmix signal representation (220; 430a-430M), and wherein the apparatus is
configured to provide one or more adjusted rendering coefficients (252; as the
adjusted parameters.
5. The apparatus (100; 250; 440) according to claim 4, wherein the parameter adjuster
is configured to receive, as the input parameters, a plurality of rendering
coefficients (214; R); and

wherein the parameter adjuster is configured to compute an average over
rendering coefficients associated with a plurality of audio objects; and
wherein the parameter adjuster is configured to provide the adjusted rendering
coefficients (252; such that a deviation of an adjusted rendering coefficient
from the average over rendering coefficients associated with a plurality of audio
objects is restricted.
6. The apparatus (100; 250; 440) according to claim 5, wherein the parameter adjuster
is configured to leave a rendering coefficient (214; R), which is within a tolerance
interval determined in dependence on the average over the rendering
coefficients, unchanged, and to selectively set a rendering coefficient (214; R),
which is larger than an upper boundary value of the tolerance interval, to a
value which is smaller than or equal to the upper boundary value, and
to selectively set a rendering coefficient (214; R), which is smaller than a lower
boundary value of the tolerance interval to a value which is larger than or
equal to the lower boundary value.
7. The apparatus (100; 250; 440) according to claim 5, wherein the parameter adjuster
is configured to iteratively select a respective one (R(imax)) of the rendering
coefficients, which comprises a maximum deviation (Rd,max) from the average
over the rendering coefficients in the respective iteration, and bring the selected one
(R(imax)) of the rendering coefficients closer to the average (R ) over the rendering
coefficients, in order to iteratively bring rendering coefficients, which are outside of
a tolerance interval determined in dependence on the average over the rendering
coefficients, into the tolerance interval.
8. The apparatus (100; 250; 440) according to claim 7, wherein the parameter adjuster
is configured to repeat the iterative selection of a respective one (R(imax)) of the
rendering coefficients and the iterative modification of the selected one of the
rendering coefficients until all rendering coefficients are adjusted to be within
applicable tolerance intervals.
9. The apparatus (100; 350; 450) according to one of claims 1 to 3, wherein the
apparatus is configured to receive one or more transcoding coefficients (337; T)
describing a mapping of one or more channels of the downmix signal representation

(210; 420) onto one or more channels of the upmix signal representation (220;
430a-430M), and
wherein the apparatus is configured to provide one or more adjusted transcoding
coefficients (352; ) as the adjusted parameters.
10. The apparatus (100; 350; 450) according to claim 9, wherein the parameter adjuster
is configured to receive, as the input parameters, a temporal sequence of
transcoding coefficients (337; T); and
wherein the parameter adjuster is configured to compute a temporal mean in
dependence on a plurality of transcoding coefficients; and
wherein the parameter adjuster is configured to provide the adjusted transcoding
coefficients (352; such that a deviation of the adjusted transcoding coefficients
from the temporal mean is restricted.
11. The apparatus (100; 350; 450) according to claim 10, wherein the parameter
adjuster is configured to leave a transcoding coefficient (337; T), which is within a
tolerance interval determined in dependence on the temporal mean unchanged,
and
to selectively set a transcoding coefficient, which is larger than an upper boundary
value of the tolerance interval, to a value which is smaller than or equal to
the upper boundary value of the tolerance interval, and
to selectively set a transcoding coefficient, which is smaller than a lower boundary
value of the tolerance interval, to a value which is larger than or equal to
the lower boundary value.
12. The apparatus (100; 350; 450) according to claim 10 or claim 11, wherein the
parameter adjuster is configured to calculate the temporal mean using a
recursive low pass filtering of the sequence of transcoding coefficients (337; T).
13. The apparatus (100; 250; 350; 440; 450) according to one of claims 1 to 12,
wherein the parameter adjuster is configured to provide a given one of the one or
more adjusted parameters such that the given one of the adjusted parameters is
within a tolerance interval, boundaries of which are defined in dependence on the

average value (132; ) of the plurality of input parameter values and one
or more tolerance parameters and such that a
deviation between an input parameter and a corresponding adjusted parameter is
minimized or kept within a predetermined maximal allowable range.
14. The apparatus (100; 250; 350; 440; 450) according to claim 13, wherein the
parameter adjuster is configured to selectively set an input parameter, which is
found to be outside of the tolerance interval, boundaries of which are defined in
dependence on the average value (132; ) of the plurality of input
parameter values, to an upper boundary value (A or a
lower boundary value of the tolerance interval, in
order to obtain an adjusted version of the input parameter.
15. The apparatus (100; 250; 350; 440; 450) according to claim 13, wherein the
parameter adjuster is configured to iteratively select a respective one (R(imax); Xj*)
of the input parameters, which comprises a maximum deviation from the average
value (132; in a respective iteration, and to bring the selected one of the
input parameters closer to the average, in order to iteratively bring input
parameters, which are determined to be outside of a tolerance interval, boundaries
of which are defined in dependence on the average value, into the tolerance
interval.
16. The apparatus (100; 350; 450) according to claim 15, wherein the parameter
adjuster is configured to choose a modification step size used to bring the selected
one (R(imax); Xi*) of the input parameters closer to the average value to be a
predetermined fraction of a difference between the selected one of the input
parameters and the average value.
17. An apparatus (200; 300; 410) for providing an upmix signal representation (220;
430a-430M) on the basis of a downmix signal representation (210; 420) and a
parametric side information (212; 422), the apparatus comprising:
an apparatus (100; 250; 350; 440; 450) for providing one or more adjusted
parameters (120; 252; 352; on the basis of one or more received parameters
(110; 214; 337; R; T), according to one of claims 1 to 16;

a signal processor (230; 330) configured to obtain the upmix signal representation
on the basis of the downmix signal representation and the parametric side
information,
wherein the apparatus for providing one or more adjusted parameters is configured
to adjust one or more processing parameters (252; 352; R; T) of the signal
processor.
18. The apparatus (200; 300; 410) according to claim 17, wherein the signal processor
(230) is configured to provide the upmix signal representation (220; 430a-430M) in
dependence on adjusted rendering coefficients (252; describing contributions of
audio objects to one or more channels of the upmix signal representation; and
wherein the apparatus (100; 250; 440) for providing one or more adjusted
parameters is configured to receive a plurality of user-specified rendering
parameters (214; R) as input parameters and to provide, on the basis thereof, one or
more adjusted rendering parameters (252; for use by the signal processor.
19. The apparatus (300; 410) according to claim 17, wherein the apparatus (100; 350;
450) for providing the one or more adjusted parameters is configured to receive one
or more mix matrix elements (337; T) of a mix matrix as the one or more input
parameters, and to provide, on the basis thereof, one or more adjusted mix matrix
elements (352; of the mix matrix for use by the signal processor (330); and
wherein the signal processor (330) is configured to provide the upmix signal
representation (220; 430a-430M) in dependence on the adjusted mix matrix
elements (352; of the mix matrix, wherein the mix matrix describes a mapping
of one or more audio channel signals of the downmix signal representation onto one
or more audio channel signals of the upmix signal representation.
20. The apparatus (200; 300; 410) according to claim 17, wherein the signal processor
is configured to obtain an MPEG surround arbitrary-downmix-gain value, and
when the apparatus for providing one or more adjusted parameters is configured to
receive a plurality of arbitrary-downmix-gain values as input parameters and to
provide a plurality of adjusted arbitrary-downmix-gain values.

21. A method for providing one or more adjusted parameters for the provision of an
upmix signal representation on the basis of a downmix signal representation and a
parametric side information associated with the downmix signal representation, the
method comprising:
receiving one or more parameters; and
providing, on the basis thereof, one or more adjusted parameters, wherein the one
or more adjusted parameters are provided in dependence on an average value of a
plurality of parameter values, such that a distortion of the upmix signal
representation caused by the use of non-optimal parameters is reduced at least for
one or more parameters deviating from optimal parameters by more than a
predetermined deviation.
22. A computer program for performing the method according to claim 21, when the
computer program runs on a computer.

ABSTRACT

An apparatus for providing one or more adjusted parameters for a provision of an upmix
signal representation on the basis of a downmix signal representation and a parametric side
information associated with the downmix signal representation comprises a parameter
adjuster. The parameter adjuster is configured to receive one or more parameters and to
provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is
configured to provide the one or more adjusted parameters in dependence on an average
value of a plurality of parameter values, such that a distortion of the upmix signal
representation caused by the use of non-optimal parameters is reduced at least for
parameters deviating from optimal parameters by more than a predetermined deviation.