1. Apparatus for generating a high frequency audio signal, comprising:
an analyzer for analyzing an input signal to determine a transient information, wherein a first portion of the input signal
has associated the transient information, and a second later portion of the input signal does not comprise the transient information;

a spectral converter for converting the input signal into an input spectral representation;
a spectral processor for processing the input spectral representation to generate a processed spectral representation comprising
values for higher frequencies than the input spectral representation; and

a time converter for converting the processed spectral representation to a time representation,
wherein the spectral converter or the time converter are controllable to perform a frequency domain oversampling for the first
portion of the input signal having associated the transient information and to not perform the frequency domain oversampling
for the second later portion of the input signal or to perform a frequency domain oversampling with a smaller oversampling
factor compared to the first portion of the input signal.

1. An audio signal decoder for providing an upmix signal representation on the basis of a downmix signal representation and
an object-related parametric information, and depending on a rendering information, the apparatus comprising:
an object parameter determinator configured to acquire inter-object-correlation values for a plurality of pairs of audio objects,
wherein the object parameter determinator is configured to evaluate a bitstream signaling parameter in order to decide whether
to evaluate individual inter-object-correlation bitstream parameter values, to acquire inter-object-correlation values for
a plurality of pairs of related audio objects, or to acquire inter-object-correlation values for a plurality of pairs of related
audio objects using a common inter-object-correlation bitstream parameter value; and

a signal processor configured to acquire the upmix signal representation on the basis of the downmix signal representation
and using the inter-object-correlation values for a plurality of pairs of related audio objects and the rendering information;

wherein the audio signal decoder is configured to combine an inter-object-correlation value IOCi,j associated with a pair of related audio objects with an object level difference value OLDi describing an object level of a first audio object of the pair of related audio objects and with an object level difference
value OLDj describing an object level of a second audio object of the pair of related audio objects, to acquire a covariance value associated
with the pair of related audio objects;

wherein the audio decoder is configured to acquire an element ei,j of a covariance matrix according to ei,j=?{square root over (OLDiOLDj)}IOCi,j,

wherein the object-related parametric information comprises the bitstream signaling parameter and the individual inter-object-correlation
bitstream parameter values or the common inter-object-correlation bitstream parameter value.

1. An apparatus for processing an audio signal to generate a bandwidth extended signal comprising a high frequency part and
a low frequency part using parametric data for the high frequency part, the parametric data relating to frequency bands of
the high frequency part, comprising:
a patch border calculator for calculating a patch border of a plurality of patch borders such that the patch border coincides
with a frequency band border of the frequency bands of the high frequency part; and

a patcher for generating a patched signal using the audio signal and the patch border, wherein the patch borders relate to
the high frequency part of the bandwidth extended signal;

wherein the patch border calculator is configured for:
calculating a frequency table defining the frequency bands of the high frequency part using the parametric data or further
configuration input data;

setting a target synthesis patch border different from the patch border using at least one transposition factor;
searching, in the frequency table, for a matching frequency band comprising a matching border coinciding with the target synthesis
patch border within a predetermined matching range, or searching for the frequency band comprising a frequency band border
being closest to the target synthesis patch border; and

selecting the matching frequency band as the patch border, wherein the matching frequency band comprises a matching border
coinciding with the target synthesis patch border within a predetermined matching range or comprises a frequency band border
being closest to the target synthesis patch border.

1. A method of operating an audio decoder to decode audio data that includes encoded blocks of N.n channels of audio data
to form decoded audio data that includes M.m channels of decoded audio, M?1, n being the number of low frequency effects channels
in the encoded audio data, and m being the number of low frequency effects channels in the decoded audio data, the method
comprising:
accepting the audio data that includes blocks of N.n channels of encoded audio data encoded by an encoding method, the encoding
method including transforming N.n channels of digital audio data, and forming and packing frequency-domain exponent and mantissa
data; and

decoding the accepted audio data, the decoding including:
unpacking and decoding the frequency-domain exponent and mantissa data;
determining transform coefficients from the unpacked and decoded frequency-domain exponent and mantissa data;
ascertaining whether M upon ascertaining that M and upon determining for a particular block to apply frequency-domain downmixing, downmixing in the frequency domain according
to downmixing data such that the frequency-domain data is data after downmixing;

inverse transforming the frequency-domain data and applying further processing to determine sampled audio data; and
if for the case M sampled audio data according to downmixing data.

1. A method for ranking candidate representative segments within media data, comprising:
creating one or more media fingerprints each of which comprises a plurality of hash bits generated from the media data;
extracting features from the media data;
detecting a plurality of scenes within the media data based at least in part on the one or more media fingerprints and a distance
analysis for the features extracted from the media data;

assigning a plurality of ranking scores to a plurality of candidate representative segments in the media data, each individual
candidate representative segment in the plurality of candidate representative segments comprises at least one scene of the
plurality of scenes in the media data, each individual ranking score in the plurality of ranking scores being assigned to
an individual candidate representative segment in the plurality of candidate representative segments;

selecting from the plurality of candidate representative segments, based on the plurality of ranking scores, a representative
segment;

wherein the method is performed by one or more computing devices.

1. An apparatus for generating a bandwidth extended audio signal from an input signal, comprising:
a patch generator for generating one or more patch signals from the input signal, wherein a patch signal comprises a patch
center frequency being different from a patch center frequency of a different patch or from a center frequency of the input
audio signal,

wherein the patch generator is configured for performing a time stretching of subband signals from an analysis filterbank,
and

wherein the patch generator comprises a phase adjuster for adjusting phases of the subband signals using a filterbank-channel
dependent phase correction, the filterbank-channel dependent phase correction comprising:

?C(k+½)

wherein k indicates a filterbank channel and C is a real number between 2 and 4.

1. A pickup system comprising:
a wind detector configured to receive first and second input signals, the wind detector including:
a plurality of analyzers each configured to analyze the first and second input signals; and
a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level
indication sinal indicative of wind activity; and

a wind suppressor including:
a ratio calculator configured to generate a ratio of sub-band powers of the first and second input signals; and
a mixer configured to select one of the first or second input signals and to apply to said selected input signal one of first
or second panning coefficients based on the wind level indication signal and on the ratio, the other of the first or second
input signals being unselected, wherein:

application of the first or second panning coefficients is a function of a ratio of the first and second input signals; and
one of the first or second panning coefficients ? is defined as
?=10?2*WindLevel*(Ratio-RatioTgt)/20Ratio?RatioTgt<0

where WindLevel is wind detector output signal provided to the wind suppressor, Ratio is a current ratio of the sub-band powers
(in dB) for the first and second input signals, and RatioTgt is a pre-selected ratio value for the sub-band powers (in dB)
of the first and second input signals.

1. A method for providing perceptually relevant loudness related values for loudness normalization to a media player, the
method comprising:
providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been
determined according to a first procedure; wherein the first procedure comprises processing the audio signal in accordance
to human loudness perception;

converting the first loudness related value into a second loudness related value using a model comprising a reversible relation;
wherein the second loudness related value is associated with a second procedure for determining loudness related values; wherein
the reversible relation is an approximation of the actual relationship between the first and second loudness related values,
and is given by either:

L2=A+BL1
wherein L2 is the second loudness related value measured in dB, L1 is the first loudness related value measured in dB, and either:

?17?A??15 and ?0.7?B??0.9;
or:
?19?A??18 and B=?1.0;
or:
L2=A+BL1+CL12
wherein L2 is the second loudness related value measured in dB, L1 is the first loudness related value measured in dB and A, B and C are real numbers;

storing the second loudness related value in metadata associated with the audio signal; and
providing the metadata to the media player to enable the media player to render the audio signal using the second loudness
related value.

1. A method for scene change detection in media data, comprising:
deriving a set of filtered values from the media data;
identifying a plurality of seed time points among time points at which the set of filtered values derived from the media data
reach extremum values;

determining one or more statistical patterns of media features in a plurality of time-wise intervals around the plurality
of seed time points of the media data using one or more types of features extractable from the media data, at least one of
the one or more types of features comprising a type of features that captures structural properties, tonality including harmony
and melody, timbre, rhythm, loudness, stereo mix, or a quantity of sound sources as related to the media data;

detecting, based on the one or more statistical patterns, a plurality of beginning scene change points and a plurality of
ending scene change points in the media data for the plurality of seed time points in the media data;

wherein the method is performed by one or more computing devices.

1. A method for generating 3D output audio comprising N+M full range channels, where N and M are positive integers and the
N+M full range channels are intended to be rendered by speakers including at least two speakers at different distances from
a listener, said method including the steps of:
(a) providing N channel input audio, comprising N full range channels;
(b) upmixing the input audio to generate the 3D output audio, and
(c) providing source depth data indicative of distance from the listener of at least one audio source,
wherein step (b) includes a step of upmixing the N channel input audio to generate the 3D output audio using the source depth
data,

wherein the N channel input audio is a soundtrack of a stereoscopic 3D video program comprising left and right eye frame images,
and step (c) includes generating the source depth data, including by identifying at least one visual image feature determined
by the 3D video program, and generating the source depth data to be indicative of determined depth of each said visual image
feature,

wherein generating the source depth data comprises measuring a disparity of the least one visual image feature of the left
and right eye frame images, using the disparity to create a visual depth map, and using the visual depth map to generate the
source depth data.

1. A method for mixing two input audio signals into a single, mixed audio signal while maintaining a perceived sound level
of the mixed audio signal, the method comprising:
receiving a main input audio signal;
receiving an associated input audio signal; wherein the associated input audio signal is coupled with the main input audio
signal;

receiving mixing metadata, which contains scaling information for scaling the main input audio signal and which specifies
how the main input audio signal and the associated input audio signal should be mixed, in order to generate a mixed audio
signal at the perceived sound level; wherein the scaling information from the mixing metadata comprises a metadata scale factor
for the main input audio signal, for scaling the main input audio signal relative to the associated input audio signal;

receiving a mixing balance input, which denotes an adjustable balance between the main input audio signal and the associated
input audio signal, wherein the mixing balance input comprises scaling information which allows a deviation from a weighting
of the main input audio signal and the associated input audio signal in the mixed audio signal as specified in the mixing
metadata;

identifying a dominant signal as either the main input audio signal or the associated input audio signal from the scaling
information provided by the mixing metadata and from the mixing balance input, wherein the respective other input audio signal
is then identified as a non-dominant signal; and wherein the dominant signal is identified by comparing the mixing balance
input with the metadata scale factor for the main input audio signal;

scaling the non-dominant signal in relation to the dominant signal; and
combining the scaled non-dominant signal with the dominant signal to yield the mixed audio signal.

1. An audio encoding system for producing, based on an audio signal, a gain profile to be distributed with said audio signal,
the gain profile comprising a time-variable voice activity gain and a time-variable and frequency-variable cleaning gain,
wherein the audio encoding system comprises:
a voice activity detector adapted to determine the voice activity gain by at least determining voice activity in the audio
signal; and

a noise estimator adapted to determine the cleaning gain by at least estimating noise in said audio signal,
wherein the cleaning gain is separable from the voice activity gain in the gain profile.

1. An audio signal encoder for providing a downmix signal representation and an object-related parametric information on the
basis of a plurality of object signals, the audio encoder comprising:
a downmixer configured to provide one or more downmix signals in dependence on downmix coefficients associated with the object
signals, such that the one or more downmix signals comprise a superposition of a plurality of object signals;

a side information provider configured to provide an inter-object-relationship side information describing level differences
and correlation characteristics of object signals and an individual-object side information describing one or more individual
properties of the individual object signals,

wherein the individual-object side information comprises an object signal tonality information which describes tonalities
of the individual object signals.

1. An apparatus for improving a left/right or mid/side stereo audio signal of an FM stereo radio receiver, the FM stereo radio
receiver configured to receive an FM radio signal comprising a mid signal and side signal, the apparatus comprising:
a parametric stereo parameter estimation stage, the parameter estimation stage configured to determine one or more parametric
stereo parameters based on the left/right or mid/side audio signal in a frequency-variant or frequency-invariant manner;

an upmix stage, the upmix stage configured to generate a stereo signal based on a first audio signal and the one or more parametric
stereo parameters, the first audio signal obtained from the left/right or mid/side audio signal; and

a quantizer unit operatively connecting the parametric stereo parameter estimation stage to the upmix stage, wherein the upmix
stage includes a standard HE AAC v2 decoder.

1. An apparatus for providing an upmix signal representation on the basis of a downmix signal representation and an object-related
parametric information, which are part of a bitstream representation of an audio content, and in dependence on a rendering
information, the apparatus comprising:
a distortion limiter configured to adjust upmix parameters using a distortion control scheme to avoid or limit audible distortions
which are caused by an inappropriate choice of rendering parameters,

wherein the distortion limiter is configured to acquire a distortion limitation control parameter which is part of the bitstream
representation of the audio content, and to adjust the distortion control scheme in dependence on the distortion limitation
control parameter;

wherein the distortion limiter is configured to evaluate a dynamic update flag within a configuration portion of the bitstream
representation of the audio content, and

wherein the distortion limiter is configured to evaluate the configuration portion of the bitstream representation of the
audio content, to acquire the distortion limitation control parameter, if the dynamic update flag is inactive, and to evaluate
a frame portion of the bitstream representation of the audio content, to repeatedly acquire updates of the distortion limitation
control parameter, if the dynamic update flag is active.

1. An apparatus for enhancing a source decoder, the source decoder generating a decoded signal by decoding an encoded signal
obtained by source encoding of an original signal, the original signal having a low band portion and a high band portion,
the encoded signal including the low band portion of the original signal and not including the high band portion of the original
signal, wherein the decoded signal is used for a high-frequency reconstruction to obtain a high-frequency reconstructed signal
including a reconstructed high band portion of the original signal, the apparatus comprising:
a high-frequency reconstructor for generating the reconstructed high band portion of the original signal from the decoded
signal;

a noise adder for generating the high-frequency reconstructed signal having a noise content similar to the noise content of
the original signal by adaptively adding noise to the reconstructed high band portion of the original signal; and

a complex low delay filter bank for synthesizing an output audio signal from a combination of the decoded signal and the high-frequency
reconstructed signal.

1. An apparatus comprising a decoding unit configured to decode a data stream, the data stream representing at least one coded
image and being stored on a non-transitory computer readable medium, wherein the data stream includes:
a plurality of context adaptive entropy encoded sub-streams for a subset of blocks of the at least one coded image, each of
the plurality of context adaptive entropy encoded sub-streams including a row of consecutive blocks of quantized coefficients
of transformed residual values;

wherein,
a first context adaptive entropy encoded sub-stream includes a first row of consecutive blocks within a first subset of blocks;
a second context adaptive entropy encoded sub-stream includes a second row of consecutive blocks within a second subset of
blocks;

the second row of consecutive blocks is immediately after the first row of consecutive blocks in a raster order of an image
decoded from the coded image; and

the second row of consecutive blocks is not immediately after the first row of consecutive blocks in the data stream.

1. A method, performed by a device, for providing audio transitions between audio signals during audio browsing, comprising
the steps of:
the device associating a first browsing direction (A1), selected by a user via a control means, for transitioning from a current audio signal (S0) to a first alternative audio signal (S1) with a first transition effect template;

the device associating a second browsing direction (A2), selected by a user via a control means, which is different from the first browsing direction, for transitioning from the
current audio signal to a second alternative audio signal (S2), with a second transition effect template, which is perceptually different from the first transition effect template, wherein
the effect is selected from the group comprising simulated motion, simulated Dopper effect, stereo regeneration, spectral
band replication, and reverberation; and

the device playing, in response to a browsing action in one of said browsing directions, a transition in which an exit segment
(S0-out1, S0-out2), extracted from the current audio signal, and an entry segment (S1-in, S2-in), extracted from the alternative audio signal, are mixed in accordance with the associated transition effect template;
and

the device subsequently playing the alternative audio signal from the end of the entry segment.

1. A computer-implemented method comprising:
receiving a stream representative of at least one coded image;
identifying, from the stream, a predetermined plurality of groups of blocks;
providing each group of blocks to a first decoding unit; and
processing, by the first decoding unit, a first block in a given group of blocks, wherein the processing of the first block
comprises:

determining that the first block is first in an order of blocks in the given group of blocks;
in response to determining that the first block is first in the order of blocks in the given group of blocks, retrieving a
first set of probability data from a buffer, wherein the first set of probability data comprises a first set of probabilities
of occurrence of symbols associated with a block that is situated immediately adjacent to the first block and that belongs
to another group of blocks that is different from the given group of blocks in the predetermined plurality of groups of blocks;

entropy decoding the first block based on the first set of probability data; and
processing, by the first decoding unit, a second block in the given group of blocks, wherein the processing of the second
block comprises:

determining that the second block is not first in the order of blocks in the given group of blocks;
in response to determining that the second block is not first in the order of blocks in the given group of blocks, retrieving
a second set of probability data from a memory unit, wherein the second set of probability data comprises a second set of
probabilities of occurrence of symbols associated with at least one other already decoded block belonging to the given group
of blocks in the predetermined plurality of groups of blocks, wherein the second set of probabilities of occurrence of symbols
are not associated with blocks that do not belong to the given subset of blocks; and

entropy decoding the second block based on the second set of probability data.

1. An audio signal enhancing device for upmixing a stereophonic input signal comprising two audio signals, the device comprising:
signal enhancement means for processing the two input signals to generate at least one enhanced signal, the signal enhancement
means comprising:

two parallel processing lines comprising two parallel processing branches each;
the first processing branch comprising adaptive filter means; and
the second processing branch comprising means for delaying a signal;
means for combining the output signal of the first processing branch of the first processing line with the output signal of
the second processing branch of the second processing line to generate a first enhanced signal;

means for combining the output signal of the first processing branch of the second processing line with the output signal
of the second processing branch of the first processing line to generate a second enhanced signal;

means for combining the first and second enhanced signals to generate a third enhanced signal;
means for determining a dominant image direction of the third enhanced signal to determine a centre channel weighting coefficient
that determines an output level of a centre channel to create a perception of sound location as in the direction of a single
front side loudspeaker; and

control means for controlling the signal enhancement means.

1. An audio encoder configured to encode at least one frame of an audio signal at a first target data rate in accordance with
an E-AC-3 codec system, thereby generating a bitstream indicative of encoded audio content including quantized mantissas,
said encoder comprising:
a transform subsystem coupled and configured to determine spectral coefficients indicative of audio content of the frame of
the audio signal;

a floating-point encoding subsystem configured to determine mantissas and encoded exponents based on the spectral coefficients;
a bit allocation and quantization subsystem configured to determine a first control parameter indicative of an allocation
of available bits for quantizing the mantissas in accordance with the E-AC-3 codec system, and to quantize the mantissas in
accordance with the first control parameter to determine the quantized mantissas;

a bitstream packing subsystem coupled and configured to generate the bitstream at the first target data rate such that said
bitstream is indicative of a second control parameter and encoded audio content of the frame, said encoded audio content including
the quantized mantissas; and

a transcoding simulation subsystem configured to simulate transcoding, said transcoding including decoding of the encoded
audio content of the frame to generate decoded data including de-quantized mantissas and re-encoding of the decoded data at
a second target data rate in accordance with an AC-3 codec system to generate a second bitstream indicative of re-encoded
audio content including re-quantized mantissas, wherein the transcoding simulation subsystem is configured to execute an iterative
bit allocation process to determine the second control parameter such that said second control parameter is indicative of
an allocation of available bits for quantizing the de-quantized mantissas to generate said re-quantized mantissas during generation
of the second bitstream in accordance with the AC-3 codec system at the second target data rate, wherein each bit allocation
iteration of the iterative bit allocation process assumes a candidate allocation of available bits determined by a different
candidate second control parameter of a set of candidate second control parameters, the set of candidate second control parameters
having been predetermined by statistical analysis of results of bit allocation processing of audio data in accordance with
the E-AC-3 codec system assuming the first target data rate, and results of bit allocation processing of the audio data in
accordance with the AC-3 codec system assuming the second target data rate.

1. A method for decoding a video bitstream, the method comprising:
a) parsing the video bitstream to identify a first slice header associated with a first portion of a picture in the video
bitstream and a second slice header associated with a second portion of the picture, comprising:

analyzing a flag of the first slice header to determine that the first slice header is a regular slice header when the flag
is equal to 0, and

analyzing a flag of the second slice header to determine that the second slice header is a partitioned slice header when the
flag is equal to 1,

wherein the second slice header is different than the first slice header and shares some slice attributes with the first slice
header, and

wherein the size of the second slice header is smaller than the size of the first slice header;
b) entropy decoding the video bitstream, the first slice header and the second slice header to produce a first decoded data
associated with the first portion of the picture and a second decoded data associated with the second portion of the picture;

c) reconstructing a first region associated with a picture in the video by using the first decoded data; and
d) reconstructing a second region associated with said picture in the video by using the second decoded data and the reconstructed
first region based on analyzing the flag of the second slice header to determine that the second slice header is a partitioned
slice header.

1. A decoder system for decoding a bit stream signal as an audio time signal, the decoder system including:
a decoding section for decoding the bit stream signal as a preliminary audio time signal, wherein the decoding section comprises
a code-excited linear prediction, CELP, decoding module and a transform-coded excitation, TCX, decoding module; and

an interharmonic noise attenuation post filter adapted to receive the preliminary audio time signal, and to supply the audio
time signal, wherein the post filter comprises a control section for selectively operating the post filter in one of the following
modes:

i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies
the filtered signal as the audio time signal; and

ii) a pass-through mode, wherein the post filter supplies the preliminary audio time signal as the audio time signal,
wherein the decoder system selectively operates in one of the following modes:
a) the TCX module is enabled and the post filter is operated in the pass-through mode;
b) the CELP module is enabled and, in response to a post-filtering signal, the post filter is operated in the filtering mode;
and

c) the CELP module is enabled and, in response to the post-filtering signal, the post filter is operated in the pass-through
mode.

1. A method for extracting tempo information of an audio signal, the method comprising:
providing a compressed, spectral band replication (SBR) encoded bitstream of the audio signal, wherein the encoded bitstream
comprises spectral band replication data;

determining an amount of data comprised in one or more fill-element fields of the encoded bit-stream for a time-interval of
the audio signal;

determining a size of SBR payload data comprised in the encoded bit-stream for the time interval of the audio signal based
on the amount of data comprised in the one or more fill-element fields of the encoded bit-stream for the time-interval of
the audio signal;

repeating the determining steps for successive time intervals of the encoded bit-stream of the audio signal, thereby determining
a sequence of sizes of SBR payload data;

identifying a periodicity in the sequence of sizes of SBR payload data; and
extracting tempo information of the audio signal from the identified periodicity, wherein the method is implemented by an
audio signal processing device comprising one or more hardware elements.

1. Audio decoder for decoding an encoded original audio signal, the encoded original audio signal comprising a lowband signal
and one or more coded spectral lines, comprising:
a line decoder configured to decode the one or more coded spectral lines;
a decoder for decoding the encoded original audio signal to produce the lowband signal;
a high frequency reconstruction processor configured to perform a high frequency reconstruction processing by extrapolating
the lowband signal to obtain a highband signal, wherein the decoded one or more spectral lines correspond to one or more differences
between the original audio signal and the highband signal; and

a combiner configured to combine the decoded one or more spectral lines, the highband signal, and the lowband signal to obtain
a decoded audio signal;

wherein one or more of the line decoder, the high frequency reconstruction processor, and the combiner are implemented, at
least in part, by one or more hardware elements of the audio decoder.

1. A decoding system configured to reconstruct an n-channel audio signal on the basis of a bitstream, the decoding system
comprising:
a parametric-mode demultiplexer for receiving the bitstream and outputting, based thereon and in a parametric coding mode
of the system, an encoded core signal and multichannel coding parameters;

a core signal decoder for receiving the encoded core signal and outputting, based thereon, an m-channel core signal, where
1?m
a parametric synthesis stage for receiving the core signal and the multichannel coding parameters and outputting, based thereon,
the n-channel signal,

wherein the parametric-mode demultiplexer is further configured to output, based on the bitstream, pre-processing dynamic
range control, DRC, parameters quantifying an encoder-side dynamic range limiting of the core signal, and

wherein the decoding system is operable to cancel the encoder-side dynamic range limiting based on the pre-processing DRC
parameters.

1. Transcoding tool for transcoding an audio stream to be played at a playback device, the transcoding tool comprising:
a receiving section adapted to receive at least one bit stream comprising an audio stream and first metadata associated with
the audio stream and second metadata associated with the audio stream,

a processing section connected to the receiving section and adapted to select between the first and second metadata, on the
basis of a selection criterion, wherein the processing section is adapted to use at least one parameter as a selection criterion,
said at least one parameter is at least one of:

associated with a configuration of the playback device, and
associated with an ambient environment of the playback device, wherein the receiving section is further adapted to receive
said at least one parameter, and

wherein the processing section is further adapted to create a processed audio stream based on the audio stream and the selected
metadata, and

wherein the transcoding tool further comprises
a transmitting section connected to the processing section and adapted to transmit the created processed audio stream to the
playback device.

1. A device for encoding an input audio signal into a channel-independent representation comprising a multichannel output
audio signal for reproduction over a multiple loudspeaker system, the device comprising:
a receiver that receives the input audio signal comprising a plurality of individual channels, N;
an interface that defines a space D covering a target audience and for partitioning the space D into a plurality of portions
k independent from the plurality of channels N;

a processor that generates at least one spatial presence factor m for each combination of an input audio channel and portion
k, wherein each factor m quantifies a degree of presence of each input audio signal into each portion k of space D; and

a processor that maps the input audio signal to the output audio signal, for reproduction within the portions k, based on
the value assigned to each spatial presence factor m.

1. A system configured to generate an improved stereo signal of a received FM radio signal; wherein the received FM radio
signal is representable as a mid signal and a side signal; wherein the side signal is indicative of a difference between a
left signal and a right signal; the system comprising:
a power determination unit configured to determine a plurality of powers for a plurality of subbands of the mid signal, referred
to as subband mid powers, and a plurality of powers for a plurality of corresponding subbands of the side signal, referred
to as subband side powers;

a ratio determination unit configured to determine a plurality of subband mid-to-side ratios as the ratios of the plurality
of subband mid powers and the plurality of subband side powers;

a quality determination unit configured to determine a quality indicator of the received FM radio signal from the minimum
of the plurality of subband mid-to-side ratios across the plurality of subbands; wherein the system is configured to generate
the improved stereo signal in dependence of the determined quality indicator.

1. A method for coding at least one image, the method comprising:
segmenting the image into a plurality of blocks,
grouping the blocks into a predetermined number of subsets of blocks,
coding, using an entropy coding module, a current block of the subsets of blocks, wherein the coding comprises:
when the current block is a first block in an encoding order of a subset which is not the first subset of the image in the
encoding order:

determining probabilities of symbol occurrence for the current block, the probabilities being those which have been determined
by coding a predetermined block of at least one other subset, wherein the predetermined block is the second block in the encoding
order in the other subset,

initializing state variables of the entropy coding module, and
coding the current block; and
generating at least one data sub-stream for the at least one image.

1. A method for decoding a unified bitstream generated by an encoder, wherein the unified bitstream is indicative of first
encoded audio data that have been encoded in accordance with a first encoding protocol and additional encoded audio data that
have been encoded in accordance with a second encoding protocol, and the unified bitstream is decodable by a first decoder
configured to decode audio data that have been encoded in accordance with the first encoding protocol, and by a second decoder
configured to decode audio data that have been encoded in accordance with the second encoding protocol, wherein the first
encoded data is interleaved with the additional encoded data with a start of a first frame of the first encoded data being
provided before a start of a first frame of the additional encoded data, with an end of the first frame of the first encoded
data being provided after the start of the first frame of the additional encoded data, with the start of the first frame of
the additional encoded data being provided before a start of a second frame of the first encoded data, and with an end of
the first frame of the additional encoded data being provided after the start of the second frame of the first encoded data,
said method including the steps of:
(a) providing the unified bitstream to a decoder configured to decode audio data that have been encoded in accordance with
the first encoding protocol; and

(b) decoding the unified bitstream using the decoder configured to decode audio data that have been encoded in accordance
with the first encoding protocol, including by decoding the first encoded audio data and ignoring the additional encoded audio
data.

1. A three-dimensional (3D) image interpolation device that performs frame interpolation on 3D video, the 3D image interpolation
device comprising:
a range image interpolation unit configured to generate at least one interpolation range image to be interpolated between
a first range image and a second range image, the first range image indicating a depth of a first image included in the 3D
video, and the second range image indicating a depth of a second image included in the 3D video;

an image interpolation unit configured to generate at least one interpolation image to be interpolated between the first image
and the second image;

a range motion vector calculation unit configured to calculate, as a range motion vector, a motion vector between the first
range image and the second range image;

an image motion vector calculation unit configured to calculate, as an image motion vector, a motion vector between the first
image and the second image;

a vector similarity calculation unit configured to calculate a vector similarity that is a value indicating a degree of a
similarity between the image motion vector and the range motion vector; and

an interpolation parallax image generation unit configured to generate, based on the at least one interpolation image interpolated
according to the vector similarity, at least one pair of interpolation parallax images having parallax according to a depth
indicated by the at least one interpolation range image; and

an interpolation image number determination unit configured to determine an upper limit of the number of interpolations, so
that the number of the interpolations increases as the vector similarity calculated by the vector similarity calculation unit
increases,

wherein the interpolation parallax image generation unit is configured to generate the at least one pair of interpolation
parallax images which is equal to or less than the upper limit determined by the interpolation image number determination
unit.

1. A system configured to determine a parametric stereo parameter from a two-channel audio signal and configured to process
frames of the two-channel audio signal using the parametric stereo parameter to generate a noise reduced two-channel audio
signal; the system comprising:
a parametric stereo parameter estimation stage configured to determine a first parametric stereo parameter based on a first
frame of the received two-channel audio signal; and

a concealment detection stage configured to
determine an energy of a side signal within the first signal frame; wherein the side signal is obtainable from the two-channel
audio signal and wherein the energy is above a high threshold;

determine a transition period of a number of following successive signal frames during which the energy of the side signal
drops from a value above the high threshold to a value below a low threshold;

determine that the two-channel audio signal following the first signal frame is a forced mono signal if the number of successive
signal frames of the transition period is below a frame threshold; and

determine the parametric stereo parameter for the processing of frames of the two-channel audio signal succeeding the first
signal frame based on the first parametric stereo parameter, if it is determined that the two-channel audio signal following
the first signal frame is a forced mono signal,

wherein the system is carried out by one or more processors and/or one or memory devices.

1. An image processing apparatus comprising: an imaging device which captures an image; an optical system which causes said
imaging device to form an image of an object; and a distance determining unit configured to determine distance of the object
between said optical system and the object based on a size of a blur developed on the image, wherein said optical system has
a characteristic which simultaneously satisfies both of conditions that (i) variation in magnification is equal to or smaller
than the predetermined number of pixels in the case where a focused point is set farthest from and closest to said optical
system in a range of the distance of the object determined by said distance determining unit and (ii) variation in a Point
Spread Function due to an image height of said optical system is equal to or smaller than a predetermined degree so as not
to affect the determination of the distance of the object by said distance determining unit; and wherein the range of the
distance is determined based on a same set of coordinates of the image at different focus points.

1. A method for detecting frequency extension coding in the coding history of an audio signal, the method comprising
providing a plurality of subband signals in a corresponding plurality of subbands comprising low and high frequency subbands,
the plurality of subband signals generated using a filter bank comprising a plurality of filters; wherein the plurality of
subband signals corresponds to a time/frequency domain representation of the audio signal;

determining a degree of relationship between subband signals in the low frequency subbands and subband signals in the high
frequency subbands; wherein the degree of relationship is determined based on the plurality of subband signals;

wherein determining the degree of relationship comprises determining a set of cross-correlation, wherein the set of cross-correlation
values comprises a subset of elements of a K x K similarity matrix, wherein the K x K similarity matrix comprises cross-correlation
values corresponding to all pairs of subband signals from the plurality of subband signals;

wherein determining a cross-correlation value comprises determining an average over time of products of corresponding samples
of a first and a second subband signal at zero time lag; and
determining frequency extension coding history if the degree of relationship is greater than a relationship threshold.

1. An audio signal decoder for providing an upmix signal representation based on a downmix signal representation and an object-related
parametric information, and depending on a rendering information, the audio signal decoder comprising:
an object parameter determinator configured to acquire inter-object-correlation values for a plurality of pairs of audio objects,
wherein the object parameter determinator is configured to evaluate a bitstream signaling parameter in order to decide whether
to evaluate individual inter-object-correlation bitstream parameter values, to acquire inter-object-correlation values for
a plurality of pairs of related audio objects, or to acquire inter-object-correlation values for the plurality of pairs of
related audio objects using a common inter-object-correlation bitstream parameter value; and

a signal processor configured to acquire the upmix signal representation based on the downmix signal representation and using
the inter-object-correlation values for the plurality of pairs of related audio objects and the rendering information;

wherein the object-related parametric information comprises the bitstream signaling parameter and the individual inter-object-correlation
bitstream parameter values or the common inter-object-correlation bitstream parameter value;

wherein the object parameter determinator is configured to evaluate an object-relationship-information, describing whether
two audio objects are related to each other; and

wherein the object parameter determinator is configured to selectively acquire inter-object-correlation values for pairs of
audio objects, for which the object-relationship-information indicates a relationship, using the common inter-object-correlation
bitstream parameter value and to set inter-object-correlation values for pairs of audio objects, for which the object-relationship
information indicates no relationship, to a predefined value.

1. A method for decoding a coded video sequence comprising:
receiving a video syntax set that includes information applicable to said coded video sequence;
determining, based on a flag included in said video syntax, that a scalability dimension identifier for said coded video sequence
is implicitly signaled, wherein the flag is indicative of either implicit or explicit signaling of the scalability dimension
identifier,

wherein said scalability dimension identifier specifies a scalability dimension of a particular layer of said coded video
sequence, the scalability dimension being one of multiple types, including: a spatial type and a quality type;

deriving the scalability dimension identifier from a network abstraction layer (NAL) unit header in response to determining
that the scalability dimension identifier is implicitly signaled;

decoding an enhancement layer based on the scalability dimension; and
generating a decoded video sequence based on, in part, the enhancement layer.

1. An audio decoder for decoding an audio bitstream generated by an audio encoder, the audio decoder comprising:
a first decoding module adapted to operate in a first coding mode;
a second decoding module adapted to operate in a second coding mode, the second coding mode being different from the first
coding mode; and

a pitch filter included in either the first coding mode or the second coding mode, the pitch filter adapted to filter a preliminary
audio signal generated by the first decoding module or the second decoding module to obtain a filtered signal,

wherein the pitch filter is selectively enabled or disabled based on a value of a first parameter encoded in the audio bitstream,
the first parameter being distinct from a second parameter encoded in the audio bitstream, the second parameter specifying
a current coding mode of the audio decoder and the pitch filter further de-emphasizes spectral valleys.

1. A method for decoding a video bitstream, the method comprising:
a) parsing the video bitstream to identify a first slice header associated with a first portion of a picture in the video
bitstream and a second slice header associated with a second portion of the picture, comprising:

analyzing a flag of the first slice header to determine that the first slice header is a regular slice header when the flag
is equal to 0, and

analyzing a flag of the second slice header to determine that the second slice header is a partitioned slice header when the
flag is equal to 1,

wherein the second slice header is different than the first slice header and shares some slice attributes with the first slice
header, and

wherein the size of the second slice header is smaller than the size of the first slice header;
b) entropy decoding the video bitstream, the first slice header and the second slice header to produce a first decoded data
associated with the first portion of the picture and a second decoded data associated with the second portion of the picture;

c) reconstructing a first region associated with a picture in the video by using the first decoded data; and
d) reconstructing a second region associated with said picture in the video by using the second decoded data and the reconstructed
first region based on analyzing the flag of the second slice header to determine that the second slice header is a partitioned
slice header.

1. An audio signal processing device for processing an audio signal, the audio signal processing device comprising:
a down sampled low delay decimated analysis filter bank comprising M analysis filters, wherein M is greater than 1 and wherein
the M analysis filters are modulated versions of an interpolated asymmetric prototype filter pi(n), wherein the interpolated asymmetric prototype filter pi(n) is determined from an asymmetric prototype filter p0(n) having a length N according to the formula:


 and
a down sampled low delay decimated synthesis filter bank comprising M synthesis filters, wherein the M synthesis filters are
modulated versions of the interpolated asymmetric prototype filter pi(n);

wherein the audio signal processing device is configured to:
obtain a plurality of subband signals by filtering the audio signal with the M analysis filters;
process the plurality of subband signals to generate a plurality of processed subband signals; and
obtain a processed audio signal by filtering the plurality of processed subband signals with the M synthesis filters
wherein one or more of the downsampled low delay decimated analysis filter bank, the down sampled low delay decimated synthesis
filter bank, obtaining the plurality of subband signals, processing the plurality of subband signals, and obtaining the processed
audio signal is implemented, at least in part, by one or more hardware elements of the audio signal processing device.

1. A computer-implemented method for entropy decoding, the method comprising:
receiving, by a decoder, a data stream representative of a coded image;
identifying, in the data stream, a plurality of rows of consecutive blocks of quantized coefficients of transformed residual
values for the coded image;

initializing one or more state variables for entropy decoding a current block in a current row of the plurality of rows;
entropy decoding the current block based on the one or more state variables for entropy decoding the current block;
wherein when the current block is a first block in the current row in a decoding order for decoding the coded image and the
current row is not the first row of the plurality of rows in the decoding order, the one or more state variables for decoding
the current block are initialized based on one or more state variables of a predetermined entropy decoded block; and

wherein the predetermined entropy decoded block is a second block in the decoding order in a row of consecutive blocks other
than the current row.

1. A system to generate a high frequency component of an audio signal, comprising:
an analysis filter bank having a frequency resolution of ?f and an analysis window of duration DA, the analysis filter bank providing a plurality of analysis subband signals from the audio signal;

a nonlinear processing unit determining a synthesis subband signal based on at least some of the plurality of analysis subband
signals, wherein the at least some of the plurality of analysis subband signals are phase shifted by a transposition order
T; and

a synthesis filter bank having a frequency resolution of Q·?f, the synthesis filter bank generating the high frequency component
from the synthesis subband signal,
wherein the value of the product of the frequency resolution ?f and the duration DA of the analysis filter bank is determined based on the frequency resolution Q·?f of the synthesis filter bank.

1. A computer-implemented method for decoding an image, the method comprising:
obtaining a set of coefficients representing a residual block of the image, the set of coefficients including a plurality
of non-zero coefficients;

determining whether a count of modifiable coefficients in the set of coefficients is greater than a predetermined number,
wherein modifiable coefficients in the set comprise a first non-zero coefficient according to a reverse scan order of the
residual block, a last non-zero coefficient according to the reverse scan order, and the coefficients between the first and
the last non-zero coefficients in the set of coefficients according to the reverse scan order;

if the count of modifiable coefficients in the set of coefficients is less than or equal to the predetermined number, determining
that a sign designation of the last non-zero coefficient according to the reverse scan order is not hidden; and

if the count of modifiable coefficients in the set of coefficients is greater than the predetermined number:
determining that the sign designation of the last non-zero coefficient is hidden, and
determining the sign designation for the last non-zero coefficient, comprising:
computing a sum of non-zero coefficients in the set of coefficients;
computing, using the sum of the non-zero coefficients, parity data; and
designating a sign for the last non-zero coefficient based on the parity data;
wherein the predetermined number is 4.

1. A decoding system for reconstructing an n-channel audio signal, wherein the decoding system is adapted to receive a bit
stream encoding an input signal segmented into time frames and representing the audio signal, in a given time frame, according
to a coding regime selected from the group comprising:
b) discrete coding using n discretely encoded channels; and
c) parametric coding of a first type using an m-channel core signal and at least one mixing parameter, wherein n>m?1,
the decoding system being operable to derive the audio signal either on the basis of said n discretely encoded channels or
by spatial synthesis,

the decoding system comprising:
an audio decoder adapted to extract a frequency-domain representation of the input signal from the bitstream and to transform
it into a time-domain representation of the input signal;

a downmix stage operable to output an m-channel downmix signal based on the time-domain representation of the input signal
in accordance with a downmix specification; and

a spatial synthesis stage operable to output an n-channel representation of the audio signal based on said downmix signal
and said at least one mixing parameter,

wherein the audio decoder is further adapted to reformat the frequency-domain representation of the input signal into n-channel
format by appending n?m neutral channels prior to transforming it into said time-domain representation, wherein the audio
decoder is adapted to perform said reformatting for at least an initial portion of each first type parametrically coded time
frame directly succeeding a discretely coded time frame and for at least a final portion of each first type parametrically
coded time frame directly preceding a discretely coded time frame.

1. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a
plurality of low frequency subband signals, the system comprising one or more processors adapted to:
receive the plurality of low frequency subband signals;
receive a set of target energies, each target energy covering a different target interval within the high frequency interval
and being indicative of the desired energy of one or more high frequency subband signals lying within the target interval;

generate the plurality of high frequency subband signals from the plurality of low frequency subband signals and from a plurality
of spectral gain coefficients associated with the plurality of low frequency subband signals, respectively, by applying the
plurality of spectral gain coefficients to the plurality of low frequency subband signals; and

adjust the energy of the plurality of high frequency subband signals using the set of target energies.

1. A method for decoding a sign-data-hiding enabled partition of an image, comprising:
receiving a set of context-based adaptive binary arithmetic coding (CABAC) encoded coefficients from an encoder;
decoding the set of CABAC encoded coefficients to generate a set of coefficients representing a residual block for the sign-data-hiding
enabled partition, the set of coefficients including a first non-zero coefficient that is without a sign designation;

applying a function to the set of coefficients to generate sign data, wherein applying the function to the set of coefficients
comprises:

computing a sum of non-zero coefficients in the set of coefficients; and
computing remainder data based on a division between the sum and a specific number; and
designating a sign for the first non-zero coefficient based on the sign data, wherein designating the sign for the first non-zero
coefficient comprises:

designating the sign for each one of the non-zero coefficients that are without sign designations based on the remainder data
and

inverse transforming the residual block for the sign-data-hiding enabled partition.

17. A method for encoding a stereo signal to a bitstream signal, the method comprising:
generating a downmix signal and a residual signal based on the stereo signal;
determining one or more parametric stereo parameters;
perceptual encoding downstream of generating the downmix signal and the residual signal, wherein
encoding based on a sum of the downmix signal and the residual signal and based on a difference of the downmix signal and
the residual signal or

encoding based on the downmix signal and based on the residual signal
is selectable in a frequency-variant or frequency-invariant manner;
wherein the method is performed by one or more microprocessor-based components.

1. A non-transitory computer-readable medium for storing data representing a sign-data-hiding enabled block of an image, comprising:
a bitstream written in the non-transitory computer-readable medium, the bitstream comprising a set of context-based adaptive
binary arithmetic coding (CABAC) encoded coefficients to be decoded by a decoder to generate a set of coefficients representing
a residual block for the sign-data-hiding enabled block, the set of coefficients including a particular non-zero coefficient
that is without a sign designation,

wherein remainder data, based on an operation representing a division between a sum of non-zero coefficients in the set of
coefficients and a specific number, is used to designate a sign for the particular non-zero coefficient.

1. Method of decoding an encoded power spectral envelope of a stereo signal or a multichannel signal, comprising:
receiving, by a receiver, the encoded power spectral envelope of the stereo signal or the multichannel signal having a first
channel and a second channel, the first channel and the second channel having a set of frequency bands, the encoded power
spectral envelope being represented by a balance parameter for each frequency band and a level parameter representing a total
power of the first channel and the second channel for each frequency band;

converting, by a converter, the balance parameters and the level parameters into power values of the first channel and the
second channel; and

calculating, by a calculator, a decoded stereo signal or a decoded multichannel signal using the power values of the first
channel and using the power values of the second channel,

wherein at least one of the receiver, the converter, and the calculator comprises a hardware implementation.

1. A system configured to generate a high frequency component of a signal from a low frequency component of the signal, the
system comprising:
an analysis filter bank configured to provide a set of analysis subband signals from the low frequency component of the signal;
wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank
has a frequency resolution of ?f;

a nonlinear processing unit configured to determine a set of synthesis subband signals from the set of analysis subband signals;
wherein the nonlinear processing unit is configured to determine an nth synthesis subband signal of the set of synthesis subband signals from a kth analysis subband signal and a (k+1)th analysis subband signal of the set of analysis subband signals; and

a synthesis filter bank configured to generate the high frequency component of the signal based on the set of synthesis subband
signals; wherein the synthesis filter bank has a frequency resolution of F?f; with F being a resolution factor, with F?1;
wherein the analysis filter bank and the synthesis filter bank are evenly stacked such that a center frequency of an analysis
subband is given by k?f and a center frequency of a synthesis subband is given by nF?f;

wherein the analysis filter bank, the nonlinear processing unit, or the synthesis filter bank are implemented at least in
part in hardware; wherein the analysis filter bank has a number LA of analysis subbands, with LA >1, where k is an analysis subband index with k =0. . . LA?1; and the synthesis filter bank has a number Ls of synthesis subbands, with Ls>0, where n is a synthesis subband index with n =0 Ls,?1.

1. An apparatus configured to reduce noise of a received multi-channel FM radio signal; wherein the received multi-channel
FM radio signal is representable as a received mid signal and a received side signal; wherein the received side signal is
indicative of a difference between a left signal and a right signal of the received multi-channel FM radio signal; the apparatus
comprising
a parameter determination unit configured to determine one or more parameters indicative of a correlation and/or decorrelation
between the received mid signal and the received side signal; wherein the parameter determination unit is configured to determine
a decorrelation parameter b indicative of a decorrelation between the received mid signal and the received side signal; and

a noise reduction unit configured to generate a noise-reduced side signal from the received mid signal using the one or more
parameters; wherein the noise reduction unit is configured to generate the noise-reduced side signal also from a decorrelated
version of the received mid signal using the decorrelation parameter b; wherein the received side signal is not in a signal
path for the generation of the noise-reduced side signal.

16. A decoding method for upmixing an input stereo signal by complex prediction stereo coding into an output stereo signal,
wherein:
said input stereo signal comprises first frequency-domain representations of a downmix channel and a residual channel and
a complex prediction coefficient; and

each of said first frequency-domain representations comprises first spectral components representing spectral content of the
corresponding signal expressed in a first subspace of a multidimensional space,

the method being performed by an upmix stage and including the steps of:
computing a second frequency-domain representation of the downmix channel based on the first frequency-domain representation
thereof, the second frequency-domain representation comprising second spectral components representing spectral content of
the signal expressed in a second subspace of the multidimensional space that includes a portion of the multidimensional space
not included in the first subspace;

computing the side channel on the basis of the first and second frequency-domain representations of the downmix signal, the
first frequency-domain representation of the residual signal and the complex prediction coefficient;

and further comprising either the step, to be performed prior to the step of upmixing, of applying temporal noise shaping,
TNS, to said first frequency-domain representation of the downmix signal and/or said first frequency-domain representation
of the residual signal;

or the step, to be performed after the step of upmixing, of applying TNS to at least one channel of said stereo signal.

1. An audio decoder for decoding an encoded audio bitstream, the audio decoder comprising:
a demultiplexer for extracting a frequency domain representation of a lowband audio signal having frequency content below
a predetermined frequency, envelope data, and additional information from the encoded audio bitstream;

a core decoder for receiving the frequency domain representation of the lowband audio signal and decoding the frequency domain
representation of the lowband audio signal to produce a time domain lowband audio signal;

an envelope decoder for receiving the envelope data and decoding the envelope data to produce an estimated spectral envelope;
an analysis filterbank for filtering the time domain lowband audio signal to produce a subband domain representation of the
lowband audio signal;

a high frequency reconstructor for regenerating a subband domain representation of a highband audio signal from the subband
domain representation of the lowband audio signal;

a manipulator for adding a spectral line that is a sinusoidal component specified by the additional information to the subband
domain representation of the highband audio signal;

an envelope adjuster for adjusting a spectral envelope of the subband domain representation of the highband audio signal based,
at least in part, on the estimated spectral envelope; and

a synthesis filterbank for combining the subband domain representation of the lowband audio signal and the subband domain
representation of the highband audio signal to produce a wideband time domain audio signal, and output the produced wideband
time domain audio signal;

wherein the high frequency reconstructor includes a transposer for transposing several consecutive analysis filter bank channels
below the predetermined frequency to certain consecutive synthesis filter bank channels above the predetermined frequency,

wherein the analysis filterbank and the synthesis filterbank are complex quadrature mirror filter (QMF) banks,
wherein the predetermined frequency includes a variable cross-over frequency,
wherein the core decoder operates at half the sampling rate of the high frequency reconstructor,
wherein the additional information includes a location of the spectral line,
wherein the location represents a filterbank channel,
wherein the spectral line is added to a middle of a scalefactor band associated with the location,
wherein the envelope adjuster compensates for the spectral line added by the manipulator based, at least in part, on the estimated
spectral envelope,

wherein the additional information further includes noise floor data and the manipulator uses the noise floor data for determining
a level of the spectral line, and

wherein one or more of the demultiplexer, the core decoder, the envelope decoder, the analysis filterbank, the high frequency
reconstructor, the manipulator, the envelope adjuster, and the synthesis filterbank are implemented, at least in part, by
one or more hardware elements of the audio decoder.

1. A decoding system that reconstructs an n-channel audio signal, wherein the decoding system receives a bitstream encoding
an input signal segmented into time frames and representing the audio signal, in a given time frame, according to a coding
regime selected from the group comprising:
a) parametric coding of a first type using at least one mixing parameter; and
b) discrete coding using n discretely encoded channels,
the decoding system deriving the audio signal either on the basis of said n discretely encoded channels or by spatial synthesis,
the decoding system comprising:
a downmixer that outputs an m-channel downmix signal based on the input signal in accordance with a downmix specification,
wherein n>m?1; and

a spatial synthesizer that outputs an n-channel representation of the audio signal based on said downmix signal and said at
least one mixing parameter,

wherein the downmixer is active in at least the first time frame in each episode of discretely coded time frames and in at
least the first time frame after each episode of discretely coded time frames, and

wherein the downmixer is deactivated during at least a time frame subsequent to the first time frame in an episode of discretely
coded time frames.

19. An audio signal decoder, comprising:
a mechanism for receiving an encoded audio signal representation including
an encoded spectrum representation, and
an encoded time warp information,
wherein the encoded spectrum representation includes a codeword describing one or more spectral values or at least a portion
of a number representation of one or more spectral values in dependence on a context state;

a context-based spectral value decoder configured to decode the codeword, to acquire decoded spectral values;
a context state determinator linked to the context-based spectral value decoder, wherein the context state determinator is
configured to determine a current context state in dependence on one or more previously decoded spectral values of the received
encoded audio signal representation; and

a time warping frequency-domain-to-time-domain converter linked to the context-based spectral value decoder, wherein the time
warping frequency-domain-to-time-domain converter is configured to output an output signal that comprises a time-warped time-domain
representation of a given audio frame of the received encoded audio signal representation on the basis of a set of decoded
spectral values associated with the given audio frame and provided by the context-based spectral value decoder and in dependence
on the time warp information;

wherein the context-state determinator is configured to adapt the determination of the context state to a change of a fundamental
frequency between subsequent audio frames; and

wherein the audio signal decoder is implemented using a hardware apparatus, or using a computer, or using a combination of
a hardware apparatus and a computer.

1. A decoder system for providing a stereo signal by complex prediction stereo coding, the decoder system comprising:
an upmix stage adapted to generate the stereo signal based on first frequency-domain representations of a downmix signal and
a residual signal, each of the first frequency-domain representations comprising first spectral components representing spectral
content of the corresponding signal expressed in a first subspace of a multidimensional space, the upmix stage comprising:

a module for computing a second frequency-domain representation of the downmix signal based on the first frequency-domain
representation thereof, the second frequency-domain representation comprising second spectral components representing spectral
content of the signal expressed in a second subspace of the multidimensional space that includes a portion of the multidimensional
space not included in the first subspace;

a weighted summer for computing a side signal on the basis of the first and second frequency-domain representations of the
downmix signal, the first frequency-domain representation of the residual signal and a complex prediction coefficient encoded
in a bit stream signal received by the decoder system; and

a sum-and-difference stage for computing the stereo signal on the basis of the first frequency-domain representation of the
downmix signal and the side signal,

wherein the upmix stage is further operable in a pass-through mode, in which said downmix and residual signals are supplied
to the sum-and-difference stage directly.

1. A method, comprising:receiving an audio signal that comprises audio content and one or more sets of differential gains wherein differential encoding is applied over discrete time indices;
identifying a specific set of differential gains, among the one or more sets of differential gains, for a gain profile in a specific playback environment;
generating a set of default gains based on the specific set of differential gains; and
based at least in part on the specific set of differential gains, controlling the dynamic range on one more portions of the audio content extracted from the audio signal;
wherein the method is performed by one or more computing devices.

1. An audio encoder configured to encode a frame of an audio signal according to a first audio codec system, thereby yielding
a first bitstream at a first target data-rate; wherein the audio encoder comprises a processor configured to perform as:
a transform unit configured to determine a set of spectral coefficients based on the frame of the audio signal;
a floating-point encoding unit configured to
determine a set of scale factors and a set of scaled values, based on the set of spectral coefficients; and
encode the set of scale factors to yield a set of encoded scale factors;
a bit allocation and quantization unit configured to
determine a total number of available bits for quantizing the set of scaled values, based on the first target data-rate and
based on the number of bits used for the set of encoded scale factors;

determine a first control parameter indicative of an allocation of the total number of available bits for quantizing the scaled
values of the set of scaled values; and

quantize the set of scaled values in accordance to the first control parameter to yield a set of quantized scaled values;
a transcoding simulation unit configured to derive a second control parameter for enabling a transcoder to convert the first
bitstream into a second bitstream at a second target data-rate; wherein the second bitstream accords to a second audio codec
system different from the first audio codec system; wherein the transcoding simulation unit is configured to derive the second
control parameter from the first control parameter; and

a bitstream packing unit configured to generate the first bitstream comprising the set of quantized scaled values, the set
of encoded scale factors, the first control parameter and the second control parameter wherein the transcoding simulation
unit is configured to derive the second control parameter from the first control parameter alone.

1. An audio processing system comprising a multiband filter for providing a partially complex frequency-domain representation
of a signal, the multiband filter comprising:
a synthesis stage receiving a first subband range of a first frequency-domain representation of a signal, the first frequency-domain
representation being segmented into time blocks and comprising first spectral components representing spectral content of
the signal in the first subband range expressed in a first subspace of a multidimensional space, and outputting, based on
the first subband range, an intermediate time-domain representation of the signal;

an analysis stage receiving the intermediate time-domain representation of the signal and outputting, based thereon, a second
frequency-domain representation of the signal, the second frequency-domain representation being segmented into time blocks
and comprising second spectral components representing spectral content of the signal in the first subband range expressed
in a second subspace of the multidimensional space that includes a portion of the multidimensional space not included in the
first subspace; and

a processor receiving the first and second subband ranges of the first frequency-domain representation of the signal and the
second frequency-domain representation of the signal and combining these to output a partially complex frequency-domain representation
of the signal,

wherein:
the synthesis stage is operable to release an approximate value of the intermediate time-domain representation in a time block
located d1?1 time blocks ahead of its output block, which approximate value is computed on the basis of any available time blocks of
the first frequency-domain representation; and

said approximate value contributes, in the analysis stage, to a time block of the second frequency-domain representation of
the signal.

1. A method for encoding initial audio signals and related spatial information into a reproduction layout-independent format,
the initial audio signals arising from any source of a plurality of sources, the method comprising:
defining a threshold directionality value to assign to one of a first group and a second group of one or more sources of the
plurality of sources requiring localization;

assigning a directionality coefficient to each source of the one or more sources;
grouping sources with a directionality coefficient above the threshold value to the first group, wherein the first group of
sources generate a first set of tracks of audio signals that require narrow localization and encoding the first group only
as a set of mono audio tracks with associated metadata describing the direction of origin of the signal of each track with
respect to a recording position, and its initial playback time;

encoding individual audio tracks of the first group with the associated metadata to facilitate playback through a minimal
number of loudspeakers about an intended location of each respective source of the first group;

grouping sources with a directionality coefficient equal to or below the threshold value to the second group, wherein the
second group sources generate a second set of tracks of audio signals that do not require narrow localization and encoding
the second group as at least one set of Ambisonics tracks of a given order and mixture of orders; and

encoding in the metadata, spread parameters associated to each source of the first group, wherein a value between 0 and 1
describes an angular width of a recorded sound image of the first group.

1. A method for encoding time marking information within audio data, wherein the audio data is a bitstream, the method comprising:
encoding time marking information as audio metadata within the audio data, thereby forming a joint bitstream, wherein the
time marking information indicates a plurality of sections of an audio object in the audio data; wherein the time marking
information is encoded in multiple positions of the audio data; wherein the multiple positions occur at a particular occurrence
rate in the audio data bitstream; wherein time marking information in a given position of the multiple positions is specified
in relation to the occurrence of the given position in the audio data bitstream; and wherein the time marking information
is encoded in a metadata container of the joint bitstream; thereby enabling a corresponding decoder to start playback of the
audio object at a beginning of a section of the audio object indicated by the time marking information.

1. An audio processing apparatus configured to accept an audio bitstream, the audio processing apparatus comprising:
an audio decoder adapted to receive the bitstream and to output quantized spectral coefficients;
a first processor that includes:
a dequantizer adapted to receive the quantized spectral coefficients and to output a first frequency-domain representation
of an intermediate signal; and

an inverse transformer for receiving the first frequency-domain representation of the intermediate signal and synthesizing,
based thereon, a time-domain representation of the intermediate signal;

a second processor that includes:
an analysis filterbank for receiving the time-domain representation of the intermediate signal and outputting a second frequency-domain
representation of the intermediate signal;

an adjuster for receiving said second frequency-domain representation of the intermediate signal and outputting a frequency-domain
representation of a processed audio signal; and

a synthesis filterbank for receiving the frequency-domain representation of the processed audio signal and outputting a time-domain
representation of the processed audio signal; and

a sample rate converter for receiving said time-domain representation of the processed audio signal and outputting a reconstructed
audio signal sampled at a target sampling frequency,

wherein the respective internal sampling rates of the time-domain representation of the intermediate audio signal and of the
time-domain representation of the processed audio signal are equal, and wherein said at least one processing component includes:

a parametric upmixer for receiving a downmix signal with M channels and outputting, based thereon, a signal with N channels,
wherein the parametric upmixer is operable at least in a mode where 1?M and

a first delay configured to incur a delay, when the parametric upmixer is in the mode where 1?M=N, to compensate for the delay
associated with the mode where 1?M mode of the parametric upmixer.

1. A method for processing an input audio signal, comprising:
separating the input audio signal into a transient component characterized by fast fluctuations in the input signal envelope
and a continuous component characterized by slow fluctuations in the input signal envelope;

processing the continuous component in a decorrelation circuit to generate a decorrelated continuous signal, wherein the decorrelated
continuous signal is scaled with a time-varying scaling function, dependent on the envelope of the input audio signal and
the output of the decorrelation circuit; and

combining the decorrelated continuous signal with the transient component to construct an output signal.

1. An interharmonic noise attenuation post filter adapted to receive an input signal, which comprises a preliminary audio
signal decoded according to one of a plurality of decoding modes and to supply an output audio signal, the interharmonic noise
attenuation post filter comprising:
a control section for selectively operating the post filter in one of the following modes:
i) a filtering mode, wherein the post filter filters the preliminary audio signal to obtain a filtered signal and supplies
the filtered signal as an output audio signal; or

ii) a pass-through mode, wherein the post filter supplies the unfiltered preliminary audio signal as an output audio signal,
said control section being adapted to receive a post-filtering signal that is generated in response to a value of a data field
associated with a time frame of the preliminary audio signal, and to disable the post filter while the post filter is operating
in the pass-through mode in response to the post-filtering signal.

1. A pitch filter for filtering a preliminary audio signal generated from an audio bitstream in an audio decoder, the pitch
filter having an operating mode selected from one of either:
(i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal,
and

(ii) an inactive mode where the pitch filter is disabled, allowing voiced content in the preliminary audio signal to pass
through the audio decoder unfiltered;

wherein the audio decoder has a coding mode selected from one of at least two distinct coding modes, and the pitch filter
is capable of being selectively operated in either the active mode or the inactive mode based on control information while
the audio decoder is operating in the coding mode.

1. A method for generating an object based audio program, said method including the steps of:determining at least one bed of speaker channels indicative of audio content of a first subset of a set of audio signals indicative of captured audio content;
determining a set of object channels indicative of audio content of a second subset of the set of audio signals;
generating object related metadata indicative of the object channels;
generating the object based audio program, such that said object based audio program includes each said bed of speaker channels, the object channels, and the object related metadata, and is renderable to provide sound perceived as a mix of first audio content indicated by one said bed of speaker channels and second audio content indicated by a selected subset of the object channels, such that the second audio content is perceived as emitting from source locations determined by the selected subset of the object channels; and
transmitting or storing the object based audio program.

1. A method for decoding video, the method comprising:
receiving quantized coefficients representative of a block of pixels in a video image;
descaling the quantized coefficients to generate descaled coefficients;
clipping the descaled coefficients to a predetermined bit depth to generate clipped coefficients;
inverse transforming in a first direction the clipped coefficients to generate first direction inverse transformed coefficients;
clipping the first direction inverse transformed coefficients to the predetermined bit depth to generate clipped inverse transformed
coefficients; and

inverse transforming the clipped inverse transformed coefficients in a second direction to determine a decoded residue.

1. A method, comprising:
receiving, in an input interface to an encoder component of an audio rendering system, audio data comprising audio objects,
the audio objects comprising audio object signals and associated metadata, the associated metadata including at least audio
object size data;

determining, by a large object detection component based on the audio object size data, a large audio object having an audio
object size that is greater than a threshold size, wherein the large audio object is spatially diffuse and requires a plurality
of speakers to reproduce the large audio object; and

performing, in a decorrelator component coupled to the input interface, a decorrelation process on audio signals of the large
audio object to produce decorrelated large audio object audio signals that are dependent on a defined location of the large
audio object and other information, wherein the decorrelated large audio object signals are mutually independent of one another,
and the decorrelation process comprises adjusting a level of each of the audio signals by adjusting a respective audio gain
for each of the audio signals to generate the decorrelated large audio object audio signals corresponding to a speaker feed
to each speaker of the plurality of speakers, and further wherein the plurality of speakers covers a large spatial area.

1. An apparatus comprising:
a non-transitory computer-readable medium storing a bitstream of image data, the bitstream comprising:
data representing a plurality of context adaptive entropy encoded sub-streams for a subset of blocks of a coded image,
where each of the plurality of context adaptive entropy encoded sub-streams comprise a row of consecutive blocks of coefficients;
wherein,
a first context adaptive entropy encoded sub-stream comprises a first row of consecutive blocks within a first subset of blocks,
and

a second context adaptive entropy encoded sub-stream comprises a second row of consecutive blocks within a second subset of
blocks; and

wherein,
the second row of consecutive blocks is immediately after the first row of consecutive blocks in a raster order of an image
decoded from the coded image, and

the second row of consecutive blocks is not immediately after the first row of consecutive blocks in the bitstream.

1. An audio encoding system, comprising:
a spatial analyzer configured to receive a plurality of audio signals, and to output, based thereon, decomposition parameters;
an adaptive rotation stage configured to receive said plurality of audio signals and to output a plurality of rotated audio
signals obtained by an adaptive energy-compacting orthogonal transformation, wherein quantitative properties of the transformation
are determined by the decomposition parameters, and wherein the plurality of rotated audio signals and the decomposition parameters
are discretely decodable into a first sound field representation; and

an analysis stage configured to output, based on said plurality of audio signals, a time-variable gain profile comprising
at least one frequency-variable component for attenuating non-voice content when applied to at least one of the plurality
of rotated audio signals, at least one of de-rotated versions of the plurality of rotated audio signal, or another sound field
representation of at least one of the plurality of rotated audio signals, wherein the analysis stage is further adapted to
output, based on said plurality of audio signals, spatial parameters adapted for use in spatial synthesis of a first rotated
audio signal,

wherein the audio encoding system is operable to suspend output of a set of signals selected from the group comprising:
said decomposition parameters and all of said plurality of rotated audio signals, except said first rotated audio signal;
and

said spatial parameters, said decomposition parameters and all of said plurality of rotated audio signals, except said first
rotated audio signal.

1. A method for decoding a coded video sequence comprising:
receiving a video syntax set that includes information applicable to said coded video sequence;
determining, based on a flag included in said video syntax set, whether a scalability dimension identifier is implicitly signaled;
wherein said scalability dimension identifier specifies a scalability dimension of a particular layer of said coded video
sequence, the scalability dimension being one of multiple types, including a spatial type and a quality type;

in response to determining that the scalability dimension identifier is implicitly signaled, deriving the scalability dimension
identifier from a network abstraction layer (NAL) unit header;

decoding a base layer of a picture in the video sequence;
decoding an enhancement layer based on the scalability dimension; and
generating a picture to be displayed based on the base layer and the enhancement layer.

10. A method for generating real-valued output audio samples, the method comprising:
storing complex-valued input subband samples in a memory;
generating the real-valued output audio samples in response to the complex-valued input subband samples, real-valued intermediate
samples, and window coefficients using a complex-valued low delay synthesis filter bank; and

storing the real-valued output audio samples in the memory,
wherein the complex-valued low delay synthesis filter bank:
shifts a first subset of the real-valued intermediate samples in the memory;
multiplies the complex-valued input subband samples by a complex-valued exponential modulation matrix to generate complex-valued
intermediate samples;

stores the real part of the complex-valued intermediate samples in the memory as a second subset of the real-valued intermediate
samples;

extracts a third subset of the real-valued intermediate samples from the memory;
multiplies the third subset of the real-valued intermediate samples by the window coefficients to generate windowed samples;
combines the windowed samples to generate the real-valued output audio samples; and
stores the real-valued output audio samples in the memory.

1. A method for encoding an audio signal comprising a first and a directly following second audio track for seamless and individual
playback of the first and second audio tracks; wherein the first and second audio tracks comprise a first and second plurality
of audio frames, respectively; the method comprising
jointly encoding the audio signal using a frame based audio encoder, thereby yielding a continuous sequence of encoded frames;
extracting a first plurality of encoded frames from the continuous sequence of encoded frames; wherein the first plurality
of encoded frames corresponds to the first plurality of audio frames;

extracting a second plurality of encoded frames from the continuous sequence of encoded frames; wherein the second plurality
of encoded frames corresponds to the second plurality of audio frames; wherein the second plurality of encoded frames directly
follows the first plurality of encoded frames in the continuous sequence of encoded frames;

appending one or more rear extension frames to an end of the first plurality of encoded frames; wherein the one or more rear
extension frames correspond to one or more frames from a beginning of the second plurality of encoded frames, thereby yielding
a first encoded audio file; and

appending one or more front extension frames to the beginning of the second plurality of encoded frames; wherein the one or
more front extension frames correspond to one or more frames from the end of the first plurality of encoded frames, thereby
yielding a second encoded audio file.

10. A method for encoding a plurality of audio signals as data suitable for parametric reconstruction, comprising:
receiving a time/frequency tile of said plurality of audio signals;
computing a downmix signal by forming linear combinations of the audio signals according to a downmixing rule, wherein the
downmix signal comprises fewer channels than the number of audio signals to be reconstructed;

determining dry upmix coefficients in order to define a linear mapping of the downmix signal approximating the audio signals
to be encoded in the time/frequency tile;

determining wet upmix coefficients based on a covariance of the audio signals as received and a covariance of the audio signals
as approximated by the linear mapping of the downmix signal; and

outputting the downmix signal together with the wet and dry upmix coefficients, which coefficients on their own enable decoder-side
computation according to a predefined rule of a further set of coefficients defining a pre-decorrelation linear mapping as
part of parametric reconstruction of the audio signals,

wherein the wet upmix coefficients are determined by:
setting a target covariance to supplement the covariance of the audio signals as approximated by the linear mapping of the
downmix signal; and

decomposing the target covariance as a product of a matrix and its own transpose, wherein the elements of said matrix, after
column-wise rescaling, correspond to the wet upmix coefficients.

1. A system configured to determine a master scale factor band table of a highband signal of an audio signal, wherein the
master scale factor band table is indicative of a frequency resolution of a spectral envelope of the highband signal; wherein
the system is configured to:
receive a set of parameters transmitted from an audio encoder along with an audio bitstream being indicative of a lowband
signal of the audio signal, the set of parameters including a selection parameter and one or more index parameters;

store a plurality of pre-determined scale factor band tables in a memory of the system independently from the audio encoder;
wherein at least one scale factor band of the pre-determined scale factor band tables comprises a plurality of frequency bands;

determine the master scale factor band table by selecting a particular one of the pre-determined scale factor band tables
based on the selection parameter of the received set of parameters and by selecting some or all of the scale factor bands
of the selected pre-determined scale factor band table using the one or more index parameters of the received set of parameters,
the one or more index parameters representing indexes into the selected pre-determined scale factor band table; and

reconstruct the highband signal from the lowband signal using the master scale factor band table.

1. A method, comprising:
receiving a bitstream including audio data and metadata associated with the audio data;
analyzing the metadata to determine whether said metadata is or includes profile metadata indicative of a target profile,
where the profile metadata is useful to perform at least one of loudness control, loudness normalization, or dynamic range
control on the audio data in accordance with the target profile, and where the target profile determines a target loudness
and/or at least one target dynamic range characteristic of a rendered version of the audio data for playback by an audio playback
device of a group of audio playback devices;

responsive to determining that the metadata is or includes the profile metadata, using the profile metadata and the audio
data to render audio including by performing said at least one of loudness control, loudness normalization, or dynamic range
control on the audio data accordance with the target profile; and

responsive to determining that the metadata is not and does not include the profile metadata, analyzing one or more characteristics
of the group, and generating the profile metadata based on the one or more characteristics.

1. A non-transitory computer-readable medium for storing data representing a sign-data-hiding enabled block of an image, comprising:
a bitstream written in the non-transitory computer-readable medium, the bitstream comprising:
a set of entropy encoded coefficients representing a set of coefficients of a residual block of the sign-data-hiding enabled
block, the set of coefficients including a particular non-zero coefficient that is without a sign designation; and

an information item representing a prediction mode of the sign-data-hiding enabled block,
wherein remainder data, which is based on an operation representing a division between a sum of non-zero coefficients in the
set of coefficients and a specific number, is used to designate a sign for the particular non-zero coefficient,

wherein the residual block of the sign-data-hiding enabled block is a difference between an original block and a prediction
block generated based on the prediction mode, and

wherein a count of non-zero coefficients in the set is greater than a threshold number.

1. A method, comprising:
receiving, by a logic system of an apparatus, audio reproduction data comprising one or more audio objects, the audio objects
comprising audio signals and associated metadata, the metadata including at least audio object position data and audio object
size data;

determining, by the logic system, for an audio object from the one or more audio objects, a plurality of virtual sources at
virtual source locations that are within an audio object area or volume defined by the audio object position data and the
audio object size data, each of the virtual source locations corresponding to a static location within a reproduction environment;

determining, by the logic system, a virtual source gain value corresponding to each of the virtual sources, the virtual source
gain value having been output from a panning process based on the virtual source location and the location of one or more
reproduction speakers of the reproduction environment, wherein determining a virtual source gain value involves retrieving
a previously-computed virtual source gain value from a memory; and

computing, by the logic system, a set of audio object gain values for each of a plurality of output channels based, at least
in part, on the virtual source gain values, wherein each output channel corresponds to at least one reproduction speaker of
the reproduction environment and wherein computing the set of audio object gain values involves interpolating between the
virtual source gain values based on an audio object position and the virtual source locations that are within the audio object
area or volume, the audio object position being based on received audio object position data.

1. A method performed in an audio decoder for reconstructing N audio channels from M audio channels, the method comprising:
receiving an encoded audio bitstream, the encoded audio bitstream including a downmixed audio signal and surround data, the
downmixed audio signal having M audio channels and the surround data including a set of spatial parameters, the set of spatial
parameters including at least one inter-channel intensity difference parameter and at least one inter-channel coherence parameter;

decoding, in a surround data decoder, the surround data to produce decoded surround data;
decoding, in a core decoder, the downmixed audio signal having M audio channels to obtain a decoded frequency domain representation
of the M audio channels, wherein the decoded frequency domain representation of the M audio channels includes a plurality
of frequency bands, and each frequency band includes one or more spectral components;

reconstructing, in a surround decoder, a frequency domain representation of the N audio channels from the decoded frequency
domain representation of the M audio channels, down-mixing information used to generate the downmixed audio signal and the
decoded surround data;

synthesizing, with one or more synthesis filterbanks, the frequency domain representation of the N audio channels to create
a time domain representation of the N audio channels; and

outputting the time domain representation of the N audio channels;
wherein M is one or more, M is less than N, and the audio decoder is implemented at least in part with hardware.

1. An audio decoding processor for decoding an encoded audio bitstream, the audio decoding processor capable of being operated
in one of at least two different decoding modes and comprising:
a demultiplexer for obtaining audio data and control information from the encoded audio bitstream;
a first audio decoder configured to operate in a first decoding mode using a first decoding technique;
a second audio decoder configured operate in a second decoding mode using a second decoding technique, the second decoding
technique being different from the first decoding technique;

a pitch predictor integrated into the second audio decoder, the pitch predictor including a long-term prediction filter and
a short-term prediction filter;

a selector for selecting one of the at least two different decoding modes based on first control information obtained from
the encoded audio bitstream;

a pitch enhancement filter that is selectively operated in either an active mode or an inactive mode based on second control
information obtained from the encoded audio bitstream that is independent of the first control information for selecting the
decoding mode, wherein the active mode causes the pitch enhancement filter to filter a preliminary audio signal generated
by the first audio decoder or the second audio decoder and the inactive mode causes the pitch enhancement filter to not filter
the preliminary audio signal; and

an output interface for outputting a decoded audio signal, the decoded audio signal being processed at least in part by the
first audio decoder or the second audio decoder.

1. An audio decoder for decoding an audio bitstream generated by an audio encoder, the audio decoder comprising:
a first decoding module adapted to operate in a first coding mode;
a second decoding module adapted to operate in a second coding mode, the second coding mode being different from the first
coding mode; and

a pitch filter included in either the first coding mode or the second coding mode, the pitch filter adapted to filter a preliminary
audio signal generated by the first decoding module or the second decoding module to obtain a filtered signal,

wherein the pitch filter is selectively enabled or disabled based on a value of a first parameter encoded in the audio bitstream,
the first parameter being distinct from a second parameter encoded in the audio bitstream, the second parameter specifying
a current coding mode of the audio decoder.

1. An image decoding method comprising:
receiving a bitstream representative of at least one coded image;
identifying, from the bitstream, a predetermined plurality of groups of blocks representative of the at least one coded image;
processing a first group of blocks, wherein the processing of the first group comprises:
entropy decoding a first row of consecutive blocks within the first group of blocks;
processing a second group of blocks, wherein the processing of the second group comprises:
entropy decoding a second row of consecutive blocks within the second group of blocks;
wherein the second row of consecutive blocks is immediately after the first row of consecutive blocks in a raster order of
an image decoded from the at least one coded image; and

wherein the second row of consecutive blocks is not immediately after the first row of consecutive blocks in the bitstream.

1. An audio encoding method, including the steps of:
(a) performing tonality detection on frequency domain audio data to generate compensation control data indicative of whether
each low frequency band of a set of at least some low frequency bands of the audio data has prominent tonal content; wherein
the frequency domain audio data comprises an exponent for said each low frequency band of the set, and wherein performing
tonality detection includes a step of determining, for said each low frequency band of the set, a measure of difference between
exponents and corresponding tented exponents of the audio data; wherein the tented exponents are determined by determining
differences between consecutive exponents and by modifying one of the exponents being subtracted so that the differences lie
within a range of 2, 1, 0, ?1 and ?2; and

(b) performing low frequency compensation to generate a corrected masking value for the audio data in each said low frequency
band having prominent tonal content as indicated by the compensation control data, and generating a masking value for the
audio data in each other low frequency band in the set without performing low frequency compensation;

wherein one or more of performing tonality detection and performing low frequency compensation is implemented, at least in
part, by one or more hardware devices.

1. A system for generating an output audio signal from an input audio signal using a transposition factor T, comprising:
an analysis window unit applying an analysis window of length La, thereby extracting a frame of the input audio signal;

an analysis transformation unit of order M, transforming the samples into M complex coefficients;
a nonlinear processing unit, altering the phase of the complex coefficients by using the transposition factor T;
a synthesis transformation unit of order M, transforming the altered coefficients into M altered samples; and
a synthesis window unit applying a synthesis window of length Ls to the M altered samples, thereby generating a frame of the output audio signal;

wherein the order M is a function of the transposition factor T; and
wherein the difference between the order M and the average length of the analysis window and the synthesis window is proportional
to (T?1).

1. A method for encoding an audio signal, the method comprising:
determining a spectral representation of the audio signal, the determining a spectral representation comprising determining
modified discrete cosine transform, MDCT, coefficients;

encoding the audio signal using the determined spectral representation;
determining a pseudo spectrum from the MDCT coefficients, wherein determining the pseudo spectrum comprises, for a particular
MDCT coefficient Xm in a particular frequency bin m, determining a corresponding coefficient Ym of the pseudo spectrum as


 wherein Xm?1 and Xm+1 are MDCT coefficients in frequency bins m?1 and m+1, respectively, adjacent to the particular frequency bin m;

classifying parts of the audio signal to be speech parts or non-speech parts based at least in part on the determined pseudo
spectrum; and

determining a loudness measure for the audio signal based on the speech parts.

1. A system configured to determine a parametric stereo parameter from a received two-channel audio signal, the system comprising:
a noise estimation stage configured to determine an impact factor characteristic for the noise of a side signal obtained from
the two-channel audio signal, based on the side signal;

a parametric stereo parameter estimation stage configured to determine the parametric stereo parameter; wherein the determining
is based on the two-channel audio signal and the impact factor; and wherein the determining involves compensating an error
of the parametric stereo parameter resulting from the noise of the side signal, using the impact factor; and

one or more processors configured to perform the noise estimation stage or the parametric stereo parameter estimation stage.

1. Apparatus for processing a time discrete input audio signal, comprising:
a synthesis filterbank that receives, as an input, a plurality of time discrete first subband signals representing the time
discrete input audio signal and having been generated by an analysis filterbank, and that synthesizes an audio intermediate
signal from the input audio signal, wherein a number of filterbank channels of the synthesis filterbank is smaller than a
number of channels of the analysis filterbank; and

a further analysis filterbank that receives, as an input, the audio intermediate signal and that generates a plurality of
time discrete second subband signals from the audio intermediate signal, wherein the further analysis filterbank comprises
a number of channels being different from the number of channels of the synthesis filterbank, and wherein a sampling rate
of a time discrete subband signal of the plurality of time discrete second subband signals is different from a sampling rate
of a time discrete first subband signal of the plurality of time discrete first subband signals,

wherein the synthesis filterbank is configured for only processing a sub-group of all first subband signals of the plurality
of first subband signals representing the full bandwidth input audio signal, and wherein the synthesis filterbank is configured
for generating the audio intermediate signal as a band segment of the full bandwidth input audio signal modulated to the base
band, and

wherein at least one of the synthesis filterbank and the analysis filterbank comprises a hardware implementation.

1. An apparatus for generating real-valued output audio samples, the apparatus comprising:
memory that stores complex-valued input subband samples, real-valued demodulated samples, and the real-valued output audio
samples;

a phase shifter that shifts a phase of the complex-valued input subband samples by an amount equal to a previously added phase
shift; and

a complex-valued synthesis filter bank that generates the real-valued output audio samples in response to the complex-valued
input subband samples, the real-valued demodulated samples, and prototype filter coefficients, wherein the complex-valued
synthesis filter bank:

shifts a first subset of the real-valued demodulated samples in the memory;
multiplies the complex-valued input subband samples by a complex-valued exponential modulation matrix to generate complex-valued
demodulated samples;

stores the real part of the complex-valued demodulated samples in the memory as a second subset of the real-valued demodulated
samples;

extracts a third subset of the real-valued demodulated samples from the memory;
multiplies the third subset of the real-valued demodulated samples by the prototype filter coefficients to generate windowed
samples;

combines the windowed samples to generate the real-valued output audio samples; and
stores the real-valued output audio samples in the memory.

1. Apparatus for performing gain adjustment on a plurality of audio sub band signals generated by filtering an audio signal
using a filter bank, the filter bank having sub band filters, adjacent sub band filters of the filterbank having transition
bands overlapping in an overlapping range, comprising:
a calculator configured for calculating a first gain adjustment value and a second gain adjustment value for grouped adjacent
audio sub band signals comprising an audio sub band signal and an adjacent audio sub band signal, wherein the calculator is
operative

to determine a first energy measure indicating a signal energy of the audio sub band signal and a second energy measure indicating
a signal energy of the adjacent audio sub band signal,

to determine an indication of a reference energy for the grouped adjacent audio sub band signals as a linear combination of
a first reference energy value for the audio sub band signal and a second reference energy value for the adjacent audio sub
band signal, and

to determine an energy estimate for an energy in the grouped adjacent audio sub band signals as a linear combination of the
first energy measure for the audio sub band signal and the second energy measure for the adjacent audio sub band signal, and

to calculate the first gain adjustment value and the second gain adjustment value for the grouped adjacent audio sub band
signals based on the linear combination of the first reference energy value for the audio sub band signal and the second reference
energy value for the adjacent audio sub band signal and based on the linear combination of the first energy measure for the
audio sub band signal and the second energy measure for the adjacent audio sub band signal; and

a gain adjuster configured for applying the first gain adjustment value to the audio sub band signal of the grouped adjacent
audio sub band signals and for applying the second gain adjustment value to the adjacent audio sub band signal of the grouped
adjacent audio sub band signals,

wherein at least one of the calculator and the gain adjuster comprises a hardware implementation.

1. An adaptive audio encoding device, comprising:
a spatial analyzer configured to receive a plurality of audio signals and to determine, based on the plurality of audio signals,
frame-wise decomposition parameters;

an adaptive rotation stage configured to receive said plurality of audio signals and to output at least a first, second, and
third rotated audio signal obtained by an energy-compacting orthogonal transformation, wherein quantitative properties of
the transformation are determined by the decomposition parameters;

a spectral envelope analyzer configured to receive a frequency-domain representation of the rotated audio signals, which contains
transform coefficients, and to output, based thereon, a spectral envelope; and

a multichannel encoder configured to receive the frequency-domain representation of the rotated audio signals and to output
transform coefficients of the first rotated audio signal only for frequency subbands in a first subband collection, transform
coefficients of the second rotated audio signal only for frequency subbands in a second subband collection, and transform
coefficients of the third rotated audio signal only for frequency subbands in a third subband collection, wherein any subbands
not included in any subband collection are to be synthesized at decoding,

wherein the multichannel encoder determines the first subband collection, the second subband collection, and the third subband
collection by means of a rate allocation process based on a joint comparison of a noise profile for the rotated audio signals
and the spectral envelopes of the rotated audio signals,

wherein at least one of the spatial analyser, the adaptive rotation stage, the spectral envelope analyser, and the multichannel
encoder, are implemented, at least in part, by one or more hardware elements of the adaptive audio encoding device.

1. An audio processing device including a subband processing unit configured to determine a synthesis subband signal from
an analysis subband signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples at
different times, each having a phase and a magnitude; wherein the analysis subband signal is associated with a frequency band
of an input audio signal; wherein the subband processing unit comprises
a block extractor configured to repeatedly
derive a frame of L input samples from the plurality of complex valued analysis samples; the frame length L being greater
than one; and

apply a block hop size of P samples to the plurality of complex valued analysis samples, prior to deriving a next frame of
L input samples;
thereby generating a suite of frames of L input samples;
a nonlinear frame processing unit configured to determine a frame of processed samples from a frame of input samples, by determining
for each processed sample of the frame:

the phase of the processed sample by offsetting the phase of the corresponding input sample; and
the magnitude of the processed sample based on the magnitude of the corresponding input sample and the magnitude of a predetermined
input sample; and

an overlap and add unit configured to determine the synthesis subband signal by overlapping and adding the samples of a suite
of frames of processed samples; wherein the synthesis subband signal is associated with a frequency band of a signal which
is time stretched and/or frequency transposed with respect to the input audio signal, wherein one or more of the block extractor,
the nonlinear frame processing unit, and the overlap and add unit is implemented, at least in part, by one or more hardware
devices.

13. An apparatus for generating low latency virtual bass, comprising:
a first component receiving an input audio signal and performing harmonic transposition on low frequency components of the
input audio signal to generate transposed data indicative of harmonics of the input audio signal; and

a second component generating a virtual bass signal in response to the transposed data and combining the virtual bass signal
with a delayed version of the input audio signal to generate an enhanced audio signal, wherein the harmonic transposition
employs combined transposition using a base transposition order B higher than 2 such that the harmonics include a second order
harmonic and at least one higher order harmonic of each of the low frequency components, and such that all of the harmonics
are generated in response to frequency-domain values determined by a common time-to-frequency domain transform stage using
an asymmetric analysis window, and a subsequent inverse transform determined by a common frequency-to-time domain transform
stage using an asymmetric synthesis window.

1. A decoder system for providing a stereo signal by complex prediction stereo coding, the decoder system comprising:
an upmix stage adapted to generate the stereo signal based on first frequency domain representations of a downmix signal and
a residual signal, each of the first frequency-domain representations comprising first spectral components representing spectral
content of the corresponding signal expressed in a first subspace of a multidimensional space, the upmix stage comprising:

a module for computing a second frequency-domain representation of the downmix signal based on the first frequency-domain
representation thereof, the second frequency-domain representation comprising second spectral components representing spectral
content of the signal expressed in a second subspace of the multidimensional space that includes a portion of the multidimensional
space not included in the first subspace;

a weighted summer for computing a side signal on the basis of the first and second frequency-domain representations of the
downmix signal, the first frequency-domain representation of the residual signal and a complex prediction coefficient encoded
in the bit stream signal; and

a sum-and-difference stage for computing the stereo signal on the basis of the first frequency-domain representation of the
downmix signal and the side signal,

wherein the upmix stage is adapted to apply independent bandwidth limits for the downmix signal and the residual signal.

1. A method for obtaining an envelope adjusted and frequency-translated signal, comprising:
filtering a lowband signal using an analysis filterbank to obtain complex-valued subband signals within a source range, wherein
each complex-valued subband signal is represented by a real-valued component and an imaginary-valued component;

patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i within
the source range to a complex-valued subband signal with index j within a reconstruction range, wherein the source range comprises
frequencies lower than frequencies in the reconstruction range;

patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i+1 within
the source range to a complex-valued subband signal with index j+1 within the reconstruction range;

applying an envelope adjustment to the patched complex-valued subband signals within the reconstruction range; and
filtering the patched and envelope adjusted complex-valued subband signals within the reconstruction range using a synthesis
filterbank to obtain the envelope adjusted and frequency-translated signal.

1. A computer-implemented method for generating a bitstream indicative of an object based audio program, wherein the object based audio program comprises a plurality of substreams; wherein the bitstream comprises a sequence of containers for a corresponding sequence of audio program frames of the object based audio program; wherein a first container of the sequence of containers comprises a plurality of substream entities for the plurality of substreams, respectively; wherein a substream entity comprises data relating to a frame of a corresponding substream; wherein the first container further comprises a presentation section; wherein the computer-implemented method comprises:determining a set of object channels indicative of audio content of at least some of a set of audio signals; wherein the set of object channels comprises a sequence of sets of object channel frames;
providing a set of object related metadata for the set of object channels; wherein the set of object related metadata comprises a sequence of sets of object related metadata frames; wherein a first audio program frame of the object based audio program comprises a first set of object channel frames and a corresponding first set of object related metadata frames; wherein an object channel is to be presented by a combination of speakers of a presentation environment, wherein the object related metadata of an object channel is indicative of a position within the presentation environment from which the object channel is to be rendered;
inserting the first set of object channel frames and the first set of object related metadata frames into a respective set of object channel substream entities of the plurality of substream entities of the first container; and
inserting presentation data into the presentation section; wherein the presentation data is indicative of at least one presentation; wherein a presentation comprises a set of substream entities from the plurality of substream entities which are to be presented simultaneously.

1. An audio decoder for decoding an encoded audio bitstream, the audio decoder comprising:
a demultiplexer for extracting a frequency domain representation of a lowband audio signal having frequency content below
a predetermined frequency, envelope data, and additional information from the encoded audio bitstream;

a core decoder for receiving the frequency domain representation of the lowband audio signal and decoding the frequency domain
representation of the lowband audio signal to produce a time domain lowband audio signal;

an envelope decoder for receiving the envelope data and decoding the envelope data to produce an estimated spectral envelope;
an analysis filterbank for filtering the time domain lowband audio signal to produce a subband domain representation of the
lowband audio signal;

a high frequency reconstructor for regenerating a subband domain representation of a highband audio signal from the subband
domain representation of the lowband audio signal;

a manipulator for adding a spectral line that is a sinusoidal component specified by the additional information to the subband
domain representation of the highband audio signal;

an envelope adjuster for adjusting a spectral envelope of the subband domain representation of the highband audio signal based,
at least in part, on the estimated spectral envelope; and

a synthesis filterbank for combining the subband domain representation of the lowband audio signal and the subband domain
representation of the highband audio signal to produce a wideband time domain audio signal, and output the produced wideband
time domain audio singal;

wherein the high frequency reconstructor includes a transposer for transposing several consecutive analysis filter bank channels
below the predetermined frequency to certain consecutive synthesis filter bank channels above the predetermined frequency,

wherein the analysis filterbank and the synthesis filterbank are complex quadrature mirror filter (QMF) banks,
wherein the predetermined frequency includes a variable cross-over frequency,
wherein the core decoder operates at half the sampling rate of the high frequency reconstructor,
wherein the additional information includes a location of the spectral line,
wherein the location represents a filterbank channel,
wherein the spectral line is added to a middle of a scalefactor band associated with the location,
wherein the envelope adjuster compensates for the spectral line added by the manipulator based, at least in part, on the estimated
spectral envelope, and

wherein one or more of the demultiplexer, the core decoder, the envelope decoder, the analysis filterbank, the high frequency
reconstructor, the manipulator, the envelope adjuster, and the synthesis filterbank are implemented, at least in part, by
one or more hardware elements of the audio decoder.

1. A signal processing device for filtering and performing high frequency reconstruction of an audio signal, the signal processing
device comprising:
an analysis filter bank that receives real valued time domain input audio samples and generates complex-valued subband samples;
a phase shifter that shifts a phase of the complex-valued subband samples by an amount;
a high frequency reconstructor or parametric stereo synthesizer that generates modified complex-valued subband samples;
a phase shifter that unshifts a phase of the modified complex-valued subband samples by the amount; and
a synthesis filter bank that receives the modified complex-valued subband samples and generates time domain output audio samples,
wherein the analysis filter bank comprises analysis filters (hk(n)) and the synthesis filter bank comprises synthesis filters (fk(n)) that are complex exponential modulated versions of a prototype filter (p0(n)) according to:


where M is a number of channels, the prototype filter (p0(n)) has a length N, and the analysis filter bank and synthesis filter bank have a system delay of D samples,

wherein the signal processing device is implemented, at least in part, by one or more hardware elements.

1. A method for decoding a bitstream associated with transform coefficients comprising the steps of:
(a) obtaining a bitstream;
(b) decoding binary data associated with a target transform coefficient from the obtained bitstream by using an arithmetic
decoding;

(c) deriving a Rice parameter for said target transform coefficient;
(d) receiving an indicator from the bitstream, wherein the indicator is a flag that is different from the Rice parameter:
(e) determining, based on the indicator, whether a first code or a second code is to be used in combination with a Rice code
corresponding to the Rice parameter; and

(f) converting said binary data to a parameter associated with magnitude of said target transform coefficient based on a combination
of the Rice code and the determined first or second code,

wherein the first code is a k-th order exponential Golomb code.

1. An apparatus for generating complex-valued output samples, the apparatus comprising:
memory that stores real-valued input audio subband samples, a windowed sample vector, and complex-valued output audio samples;
and

a complex-valued low delay analysis filter bank that generates the complex-valued output audio samples in response to the
real-valued input audio subband samples, the windowed sample vector, and prototype filter coefficients, wherein the complex-valued
low delay analysis filter bank:

shifts a first subset of the real-valued input audio subband samples in the memory;
multiplies the first subset of the real-valued input audio subband samples by the prototype filter coefficients to generate
windowed samples;

combines the windowed samples to generate the windowed sample vector;
multiplies the windowed sample vector by a complex-valued exponential modulation matrix to generate complex-valued output
audio subband samples; and

stores the complex-valued output audio subband samples in the memory.

1. A method for dynamic range control (DRC) of input audio signals, the method comprising:
receiving, by a decoding system, a bitstream having an input audio signal and encoder-generated DRC metadata, the encoder-generated
DRC metadata containing a plurality of sets of DRC gains, the plurality of sets of DRC gains including a first set of DRC
gains that have been applied by an encoding system to the input audio signal of the bitstream and a second set of DRC gains
that are yet to be applied by the decoding system to the input audio signal;

determining, based on one or more of user input or properties of playback equipment, one of the first set of DRC gains or
the second set of DRC gains as a specific set of DRC gains that should be applied to the input audio signal; and

applying the specific set of DRC gains as at least a part of overall gains applied to the input audio signal.

1. A method, comprising:
receiving audio data comprising N audio objects, the audio objects including audio signals and associated metadata, the metadata
including at least audio object position data; and

performing an audio object clustering process that produces M clusters from the N audio objects, M being a number less than
N, wherein the clustering process comprises:

selecting M representative audio objects;
determining a cluster centroid position for each of the M clusters according to audio object position data of each of the
M representative audio objects, each cluster centroid position being a single position that is representative of positions
of all audio objects associated with a cluster; and

determining a gain contribution of the audio signal for each of the N audio objects to at least one of the M clusters, wherein
determining the gain contribution involves:

determining a center of loudness position that is a function of cluster centroid positions and gains assigned to each cluster;
and

determining a minimum value of a cost function, the cost function including three terms, a first term representing a difference
between the center of loudness position and an audio object position, a second term representing a distance between the object
position and a cluster centroid position and a third term setting a scale for determined gain contributions allowing the cost
function to discriminate between determined gain contributions and select a single set of gain contributions from multiple
sets of gain contributions, wherein the number of clusters is minimized for which the single set of gain contributions is
selected, wherein determining the center of loudness position involves:

determining products of each cluster centroid position and a gain assigned to each cluster centroid position;
calculating a sum of the products;
determining a sum of the gains for all cluster centroid positions; and
dividing the sum of the products by the sum of the gains.

1. A method, performed by an audio signal processing device, for estimating a first sample of a first subband signal in a
first subband of an audio signal; wherein the first subband signal of the audio signal is determined using an analysis filterbank
comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the
audio signal, respectively, the method comprising
determining a model parameter of a signal model;
determining a prediction coefficient to be applied to a previous sample of a first decoded subband signal derived from the
first subband signal, based on the signal model, based on the model parameter and based on the analysis filterbank; wherein

a time slot of the previous sample is prior to a time slot of the first sample; and
determining an estimate of the first sample by applying the prediction coefficient to the previous sample;
wherein
determining the prediction coefficient comprises determining the prediction coefficient using a look-up table or an analytical
function;

the look-up table or the analytical function provide the prediction coefficient as a function of a parameter derived from
the model parameter;

the look-up table or the analytical function are pre-determined based on the signal model and based on the analysis filterbank;
and

the audio signal processing device comprises one of more processors.

1. An encoder configured to generate control data from an audio signal, wherein the audio encoder:
analyses the spectral shape of the audio signal and determines a degree of spectral envelope discontinuities introduced when
re-generating a high frequency component of the audio signal from a plurality of low frequency subband signals of the audio
signal; wherein determining the degree of spectral envelope discontinuities comprises determining a ratio information by studying
lowest frequencies of the plurality of low frequency subband signals and highest frequencies of the plurality of low frequency
subband signals to assess a spectral variation of the plurality of low frequency subband signals; and

generates control data for controlling the re-generation of the high frequency component based on the degree of discontinuities.

1. A method for decoding a sign-data-hiding enabled data signal representative of at least one image split up into partitions
which has been previously coded, the decoding method comprising:
decoding the data of the current partition to obtain a set of coefficients, wherein a particular non-zero coefficient of the
set of coefficients is without a sign designation;

identifying a first non-zero coefficient within the set of coefficients;
identifying a last non-zero coefficient within the set of coefficients;
determining a count of coefficients in the set of coefficients from the first non-zero coefficient to the last non-zero coefficient
including the first non-zero coefficient and the last non-zero coefficient; and

in response to determining that the count is greater than 4:
calculating a sum of amplitude information of non-zero coefficients in the set of coefficients;
calculating, using the sum of the amplitude information of the non-zero coefficients, parity data;
designating, on the basis of the parity data, a sign for the particular non-zero coefficient; and
dequantizing the set of coefficients including the particular non-zero coefficient designated with the sign.

1. A decoding method for decoding two audio signals, comprising the steps of:
receiving a first signal and a second signal corresponding to a time frame of the two audio signals, wherein the first signal
comprises a first waveform-coded signal comprising spectral data corresponding to frequencies up to a first cross-over frequency
and a downmix signal comprising waveform-coded spectral data corresponding to frequencies between a first cross-over frequency
and a second cross-over frequency, and wherein the second signal comprises a second waveform-coded signal comprising spectral
data corresponding to frequencies up to the first cross-over frequency, wherein the first and the second waveform-coded signal
as received are waveform-coded in a left-right form, or a downmix-complementary form wherein, in case of a downmix-complementary
form, the complementary signal depends on a weighting parameter ? which is signal adaptive and which is received in addition
to the received first and second signals;

transforming the first and the second waveform-coded signals into a sum-and-difference form such that the first signal is
a combination of a waveform-coded sum-signal comprising spectral data corresponding to frequencies up to the first cross-over
frequency and said downmix signal comprising spectral data corresponding to frequencies between the first cross-over frequency
and the second cross-over frequency, and the second signal comprises a waveform-coded difference-signal comprising spectral
data corresponding to frequencies up to the first cross-over frequency;

receiving high frequency reconstruction parameters;
extending said downmix signal to a frequency range above the second cross-over frequency by performing high frequency reconstruction
using the high frequency reconstruction parameters,

receiving upmix parameters,
mixing the first and the second signal so as to generate a left and a right channel of a stereo signal, wherein for frequencies
below the first cross-over frequency the mixing comprises performing an inverse sum-and-difference transformation of the first
and the second signal, and for frequencies above the first cross-over frequency the mixing comprises performing parametric
upmixing of said downmix signal by using the upmix parameters.

1. A method for determining relative polarities of a set of N speakers in a playback environment using a set of M microphones
in the playback environment, where M is a positive integer and N is an integer greater than one, said method including steps
of:
(a) measuring impulse responses, including an impulse response for each speaker-microphone pair;
(b) clustering the speakers into a set of groups, each group in the set including at least two of the speakers which are similar
to each other in at least one respect; and

(c) for each said group, determining cross-correlations of pairs of the impulse responses of speakers in the group and determining
relative polarity of the speakers in said group from the cross-correlations.

1. A method, comprising:
generating audio content coded for a reference speaker configuration;
downmixing the audio content coded for the reference speaker configuration to downmix audio content coded for a specific speaker
configuration;

performing one or more gain adjustments on individual portions of the downmix audio content coded for the specific speaker
configuration, wherein the one or more gain adjustments use different gain adjustment parameter values for at least two different
portions of the individual portions of the downmixed audio content;

performing loudness measurements on the individual portions of the downmix audio content; and
generating an audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness
metadata created based at least in part on the loudness measurements on the individual portions of the downmix audio content;
wherein the method is performed by one or more computing devices and the one or more gain adjustments comprise at least one
gain adjustment relating to one or more of dialogue normalization, dynamic range compression, or fixed attenuation to protect
against downmix overload.

1. An apparatus comprising:
a non-transitory computer-readable medium having stored thereon instructions that when executed by one or more processors
cause the one or more processors to perform operations to generate a plurality of reconstruction slices of coded image data
representative of a video frame;

wherein each reconstruction slice comprises a plurality of entropy slices;
wherein each entropy slice comprises a plurality of macroblocks;
wherein no macroblock in a reconstruction slice is in two different entropy slices;
wherein a first entropy slice in a first reconstruction slice is entropy coded independent of the remaining entropy slices
in the first reconstruction slice;

wherein the first entropy slice represents a first portion of the video frame;
wherein a second entropy slice in the first reconstruction slice represents a second portion of the video frame and is encoded
using the first portion of the video frame;

wherein each entropy slice comprises a respective header;
wherein each header includes a respective flag;
wherein when the flag in the header is 0 then the entropy slice is the first entropy slice of the reconstruction slice;
wherein when the flag in the header is 1 then the entropy slice is not the first entropy slice of the reconstruction slice;
and

wherein the header of the second entropy slice is smaller than the header of the first entropy slice.

1. A method for decoding a motion vector predictor of a current block in a picture of a sequence of pictures, the method comprising:
identifying a first adjacent block which is adjacent to the current block in the picture;
identifying a second adjacent block which is adjacent to the current block in the picture;
when a motion vector of the first adjacent block is not equal to a motion vector of the second adjacent block, generating
a motion vector predictor candidate set comprising the motion vectors of both the first and the second adjacent blocks;

when the motion vector of the first adjacent block is equal to the motion vector of the second adjacent block, generating
a motion vector predictor candidate set comprising only the motion vector of the first adjacent block or the motion vector
of the second adjacent block;

receiving a flag from a bitstream indicating whether a temporally-located motion vector can be used as a motion vector predictor;
when the flag indicates that a temporally-located motion vector can be used as a motion vector predictor, including a motion
vector of a block in another picture in the motion vector predictor candidate set;

when the flag indicates that a temporally-located motion vector cannot be used as a motion vector predictor, excluding the
motion vector of the block in another picture from the motion vector predictor candidate set; and

selecting a motion vector from the motion vector predictor candidate set as the motion vector predictor of the current block.

1. An audio signal processing device, the apparatus comprising:
a low delay decimated analysis filter bank comprising M analysis filters, wherein M is greater than 1 and wherein the M analysis
filters are modulated versions of an asymmetric prototype filter p0(n) having a length N; and

a low delay decimated synthesis filter bank comprising M synthesis filters, wherein the M synthesis filters are modulated
versions of the asymmetric prototype filter p0(n);

wherein the audio signal processing device is configured to:
obtain a plurality of subband signals by filtering the audio signal with the M analysis filters, wherein the filtering comprises
applying, to arrays of 128 samples derived from the audio signal, a complex-exponential modulation matrix of the form:


process the plurality of subband signals to generate a plurality of processed subband signals; and
obtain a processed audio signal by filtering the plurality of processed subband signals with the M synthesis filters, wherein
the filtering comprises applying, to arrays of M samples derived from the plurality of processed subband signals, a complex-exponential
modulation matrix of the form:


wherein one or more of obtaining a plurality of subband signals, processing the plurality of subband signals, and obtaining
a processed audio signal is implemented, at least in part, by one or more hardware devices within the audio signal processing
device.

1. A system configured to generate a time stretched and/or frequency transposed signal from an input signal, the system comprising:
an analysis filter bank configured to derive a number Y?1 of analysis subband signals from the input signal, wherein each
analysis subband signal comprises a plurality of complex-valued analysis samples, each having a phase and a magnitude;

a subband processing unit configured to generate a synthesis subband signal from the Y analysis subband signals using a subband
transposition factor Q and a subband stretch factor S, at least one of Q and S being greater than one, wherein the subband
processing unit comprises:

a block extractor configured to:
i) form Y frames of L input samples, each frame being extracted from said plurality of complex-valued analysis samples in
an analysis subband signal and the frame length being L>1; and

ii) apply a block hop size of h samples to said plurality of analysis samples, prior to forming a subsequent frame of L input
samples, thereby generating a sequence of frames of input samples;

a nonlinear frame processing unit configured to generate, on the basis of Y corresponding frames of input samples formed by
the block extractor, a frame of processed samples by determining a phase and magnitude for each processed sample of the frame,
wherein, for at least one processed sample:

i) the phase of the processed sample is based on the respective phases of the corresponding input sample in each of the Y
frames of input samples; and

ii) the magnitude of the processed sample is determined as a mean value of the magnitude of the corresponding input sample
in a first frame of the Y frames of input samples and the magnitude of the corresponding input sample in a second frame of
the Y frames of input samples;

and
an overlap and add unit configured to determine the synthesis subband signal by overlapping and adding the samples of a sequence
of frames of processed samples;

and
a synthesis filter bank configured to generate the time stretched and/or frequency transposed signal from the synthesis subband
signal, wherein the system is operable at least for Y=2.

1. A multi-channel decoder for generating an energy-corrected binaural signal from a downmix signal derived from an original multi-channel signal using parameters including an upmix rule information useable for upmixing the downmix signal with an upmix rule, the upmix rule resulting in an energy-error, comprising:a gain factor calculator configured for calculating at least one gain factor for reducing or eliminating the energy-error obtainable by the upmixing of the downmix signal using the upmix rule, based on the upmix rule information and filter characteristics of head related transfer function based filters corresponding to upmix channels, wherein the gain factor calculator is operative to calculate the gain factor based on

wherein gn is the gain factor for the first channel, when n is set to 1, wherein g2 is the gain factor of a second channel, when n is set to 2, wherein EnB is a weighted addition energy calculated by weighting energies of channel impulse responses using weighting parameters, and wherein ?EnB is an estimate for the energy error introduced by the upmix rule, wherein ?, ?, and ? are upmix rule dependent parameters, and wherein ? is a number greater than or equal to zero; and
a processor for generating head related transfer function parameters, wherein the filter characteristics of the head related transfer function are determined based on the head related transfer function parameters; and
a filter processor configured for filtering the downmix signal using the at least one gain factor, the filter characteristics of the head related transfer function based filters and the upmix rule information to obtain the energy-corrected binaural signal.

1. A computer-implemented method for entropy decoding, the method comprising:receiving, by a decoder, a data stream representative of a coded image;
identifying, in the data stream, a plurality of rows of consecutive blocks of quantized coefficients of transformed residual values for the coded image;
initializing one or more state variables for entropy decoding a current block in a current row of the plurality of rows; and
entropy decoding the current block based on the one or more state variables for entropy decoding the current block,
wherein when the current block is a first block in the current row in a decoding order for decoding the coded image and the current row is not the first row of the plurality of rows in the decoding order, the one or more state variables for decoding the current block are initialized based on one or more state variables of a predetermined entropy decoded block,
wherein the predetermined entropy decoded block is a second block in the decoding order in a row of consecutive blocks other than the current row,
wherein entropy decoding a block comprises partitioning an interval into a plurality of sub-intervals where the sub-intervals represent probabilities of occurrence of symbols, and
wherein when the current block is a last block in the current row in decoding order, the interval is reinitialized.

1. A method, comprising:determining positions of a plurality of speakers;
receiving an object-based audio item comprising at least one sound content portion associating with a gain value and position metadata indicating a content position in a virtual sound plane at which the sound content portion is intended to play;
determining, based on the content position and the positions of the plurality of speakers, a gain adjustment value for the sound content portion;
wherein the gain adjustment value for the sound content portion is set to be scaled with a maximum gain adjustment value; wherein the maximum gain adjustment value is determined based on availability of speakers in different spatial regions;
adjusting the gain value for the sound content portion according to the gain adjustment value;
causing the source content portion with the adjusted gain value to be played by the plurality of speakers.

1. A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels, the method comprising:receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter;
decoding the encoded audio signal having M audio channels to obtain a decoded representation of the M audio channels;
decorrelating at least a portion of the decoded representation with an all-pass filter to obtain M decorrelated signals, the all-pass filter including a plurality of filter links, wherein a transfer function H(z) in a Z-domain of at least some of the plurality of filter links is at least partially derivable from or based on:

where q is a complex valued phase rotation factor, m is a delay length and a is a filter coefficient;
reconstructing N audio channels from the M decorrelated signals and the decoded representation of the M audio channels to obtain N audio signals that collectively having N audio channels, wherein N is two or more, M is one or more, and M is less than N; and
synthesizing the N audio signals with one or more synthesis filterbanks to convert the N audio signals from a frequency domain to a time domain;
wherein the set of spatial parameters is defined on a per frame basis and the audio decoder is implemented at least in part with hardware.

1. A method for encoding metadata relating to N audio objects of an audio scene, with N >1; whereinthe metadata comprises a first set of metadata and a second set of metadata;
the first set of metadata is associated with M downmix signals;
the M downmix signals are generated by downmixing the N audio objects; and
M is smaller than N;
the first set of metadata comprises one or more data elements indicative of a property of a downmix signal from the M downmix signals;
a property of a downmix signal describes how the downmix signal is to be rendered by a channel-based renderer;
the second set of metadata comprises one or more data elements which are indicative of a property of one or more audio objects from the N audio objects;
a property of an audio object describes how the audio object is to be rendered by an object-based renderer; and
the method comprises
identifying a redundant data element which is common to the first and second sets of metadata; and
encoding the redundant data element of the first set of metadata by referring to a redundant data element external to the first set of metadata.

1. A method performed in an audio decoder for reconstructing an original audio signal having a lowband portion and a highband
portion, the method comprising:
receiving an encoded audio signal, the encoded audio signal including spectral coefficients of the lowband portion and not
the highband portion;

extracting reconstruction parameters from the encoded audio signal, the reconstruction parameters including a cross over frequency,
spectral envelope information, and location information, wherein the spectral envelope information includes a spectral envelope
value for each frequency band of the highband portion and the location information specifies a particular frequency band of
the highband portion;

decoding the encoded audio signal with a core audio decoder to obtain a decoded lowband portion, the core audio decoder operating
at a first sampling frequency;

regenerating the highband portion based at least in part on the cross over frequency and the decoded lowband portion to obtain
a regenerated highband portion, wherein the regenerating operates at a second sampling frequency that is twice the first sampling
frequency;

creating a synthetic sinusoid with a level based at least in part on the spectral envelope value for the particular frequency
band and a noise floor value for the particular frequency band, the synthetic sinusoid representing a tonal component;

adding the synthetic sinusoid to the regenerated highband portion in the particular frequency band specified by the location
information, wherein the location information specifies a frequency band where a difference is detected between a highband
of the original audio signal and the regenerated highband portion, and

combining the lowband portion and the regenerated highband portion to obtain a full bandwidth audio signal; and
outputting the full bandwidth audio signal,
wherein the audio decoder is implemented at least in part with hardware.

1. A method for decoding video comprising:
receiving quantized coefficients representative of a block of video representative of a plurality of pixels;
descaling the quantized coefficients based on a quantization parameter, a coefficient index, and a transform size to generate
descaled coefficients;

applying an adjustment mechanism to the descaled coefficients to modify the descaled coefficients to generate modified descaled
coefficients, wherein the adjustment mechanism is a variable based on the transform size;

clipping the modified descaled coefficients to a predetermined bit depth to generate clipped coefficients;
storing the clipped coefficients in a memory; and
inverse transforming the clipped coefficients to output a decoded residue, wherein the inverse transforming comprises:
one-dimension inverse transforming in a first direction the clipped coefficients to generate first direction inverse transformed
coefficients;

prior to one-dimension inverse transforming the first direction inverse transformed coefficients in a second direction, shifting
the first direction inverse transformed coefficients right to generate first shifted inverse transformed coefficients having
a bit depth less than the first direction inverse transformed coefficients;

clipping the first shifted inverse transformed coefficients to the predetermined bit depth to generate second clipped coefficients;
and

one-dimension inverse transforming the second clipped coefficients in the second direction to determine the decoded residue.

8. A system for generating an object based audio program indicative of audio content including first non-ambient content,
second non-ambient content different than the first non-ambient content, and third content different than the first non-ambient
content and the second non-ambient content, said system including:
a first subsystem configured to determine:
a set of object channels consisting of N object channels, where a first subset of the set of object channels is indicative
of the first non-ambient content, the first subset consists of M object channels of the set of object channels, each of N
and M is an integer greater than zero, and M is equal to or less than N,

a bed of speaker channels indicative of a default mix of audio content, where an object based speaker channel subset consisting
of M of the speaker channels of the bed is indicative of the second non-ambient content or a mix of the second non-ambient
content and at least some other audio content of the audio content of the default mix, and

a set of M replacement speaker channels, where each replacement speaker channel in the set of M replacement speaker channels
is indicative of none or some, but not all, of the content of a corresponding speaker channel of the object based speaker
channel subset, wherein the first subsystem is also configured to generate metadata indicative of at least one selectable
predetermined alternative mix of content of at least one of the object channels and content of predetermined ones of the speaker
channels of the bed and/or the replacement speaker channels, where the metadata includes rendering parameters for each said
alternative mix, and at least one said alternative mix is a replacement mix indicative of at least some of the audio content
of the bed and the first non-ambient content, but not the second non-ambient content; and

an encoding subsystem coupled to the first subsystem and configured to generate the object based audio program, such that
said object based audio program includes the bed of speaker channels, the set of M replacement speaker channels, the set of
object channels, and the metadata, and such that the bed of speaker channels is renderable without use of the metadata to
provide sound perceivable as the default mix, and the replacement mix is renderable, in response to at least some of the metadata,
to provide sound perceivable as a mix including said at least some of the audio content of the bed and the first non-ambient
content but not the second non-ambient content.

1. An audio signal encoder for providing a bitstream representation on the basis of a plurality of audio object signals, the
audio signal encoder comprising:
a downmixer configured to provide a downmix signal on the basis of the audio object signals and in dependence on downmix parameters
describing contributions of the audio object signals to one or more channels of the downmix signal; and

a parameter provider configured to provide a common inter-object-correlation bitstream parameter value associated with a plurality
of pairs of related audio object signals, and to also provide a bitstream signaling parameter indicating that the common inter-object-correlation
bitstream parameter value is provided instead of a plurality of individual inter-object-correlation bitstream parameter values;

wherein the parameter provider is configured to also provide an object relationship information describing whether two audio
objects are related to each other; and

a bitstream formatter configured to provide a bitstream comprising a representation of the downmix signal, a representation
of the common inter-object-correlation bitstream parameter value and the bitstream signaling parameter.

1. An audio decoder for decoding an encoded audio bitstream, the audio decoder comprising:
a demultiplexer for extracting a frequency domain representation of a lowband audio signal having frequency content below
a predetermined frequency, envelope data, and additional information from the encoded audio bitstream;

a core decoder for receiving the frequency domain representation of the lowband audio signal and decoding the frequency domain
representation of the lowband audio signal to produce a time domain lowband audio signal;

an envelope decoder for receiving the envelope data and decoding the envelope data to produce an estimated spectral envelope;
an analysis filterbank for filtering the time domain lowband audio signal to produce a subband domain representation of the
lowband audio signal;

a high frequency reconstructor for regenerating a subband domain representation of a highband audio signal from the subband
domain representation of the lowband audio signal;

a manipulator for adding a spectral line that is a sinusoidal component specified by the additional information to the subband
domain representation of the highband audio signal;

an envelope adjuster for adjusting a spectral envelope of the subband domain representation of the highband audio signal based,
at least in part, on the estimated spectral envelope; and

a synthesis filterbank for combining the subband domain representation of the lowband audio signal and the subband domain
representation of the highband audio signal to produce a wideband time domain audio signal, the produced wideband time domain
audio signal is output as an analog wideband signal;

wherein the high frequency reconstructor includes a transposer for transposing several consecutive analysis filter bank channels
below the predetermined frequency to certain consecutive synthesis filter bank channels above the predetermined frequency,

wherein the analysis filterbank and the synthesis filterbank are complex quadrature mirror filter (QMF) banks,
wherein the predetermined frequency includes a variable cross-over frequency,
wherein the core decoder operates at half the sampling rate of the high frequency reconstructor,
wherein the additional information includes a location of the spectral line,
wherein the location represents a filterbank channel,
wherein the spectral line is added to a middle of a scalefactor band associated with the location, and
wherein one or more of the demultiplexer, the core decoder, the envelope decoder, the analysis filterbank, the high frequency
reconstructor, the manipulator, the envelope adjuster, and the synthesis filterbank are implemented, at least in part, by
one or more hardware elements of the audio decoder.

1. A method for reconstructing a wideband audio signal, the method comprising:
decomposing a lowband audio signal into a plurality of complex subband signals with an L-channel analysis filterbank, each
of the plurality of complex subband signals representing a frequency channel of the analysis filterbank;

generating a highband audio signal by patching a number of consecutive complex subband signals, wherein the generating includes:
frequency translating a complex subband signal in a source area channel of the lowband audio signal having an index i to a
reconstruction range channel having an index j of the highband audio signal, and

frequency translating a complex subband signal in a source area channel of the lowband audio signal having an index i+1 to
a reconstruction range channel having an index j+1 of the highband audio signal;

adjusting a spectral envelope of the highband audio signal to a desired level;
combining the lowband audio signal and the highband audio signal with a Q·L-channel synthesis filterbank to generate the wideband
audio signal,

wherein the lowband audio signal has frequency components below a crossover region and the highband audio signal has frequency
components above the crossover region,

wherein Q is chosen so that Q·L is an integer value, and
wherein the frequency translating of the lowband audio signals having an index i and i+1 represent a patch, and the generating
uses multiple patches.

1. A signal processing device for filtering and processing an audio signal, the signal processing device comprising:
an analysis filter bank that receives real valued time domain input audio samples and generates complex-valued subband samples;
a high frequency reconstructor or a parametric stereo processor that generates modified complex-valued subband samples; and
a synthesis filter bank that receives the modified complex-valued subband samples and generates time domain output audio samples,
wherein the analysis filter bank comprises analysis filters (hk(n)) and the synthesis filter bank comprises synthesis filters (fk(n)) that are complex exponential modulated versions of a prototype filter (p0(n)) according to:


where M is a number of channels, the prototype filter (p0(n)) has a length N, and the analysis filter bank and synthesis filter bank have a system delay of D samples,

wherein the signal processing device is implemented, at least in part, by one or more hardware elements.

1. Apparatus for generating a corrected multi-channel representation of an original multi-channel signal having at least three
original channels, the apparatus comprising:
a receiver for receiving a parameter representation having a parameter set, allowing a multi-channel reconstruction, the parameter
representation including a level parameter in addition to the parameter set, the level parameter being a level difference
between a master down-mix and a parameter down-mix, on which the parameter set is based;

a level corrector for applying a level correction of at least one down-mix channel using the level parameter by weighting
the at least one down-mix channel using the level parameter; and

an upmixer for up-mixing using parameters in the parameter set to obtain the corrected multi-channel representation.

1. An audio processing system for performing spatial synthesis,
the system comprising an upmix stage for receiving a decoded m-channel downmix signal and for outputting, based thereon, an
n-channel upmix signal, wherein 2?m
a downmix modifying processor for receiving the m-channel downmix signal and for outputting a modified m-channel downmix signal,
the downmix modifying processor adapted to cross mix and process the downmix signal in a non-linear fashion; and
a first mixing matrix for receiving the downmix signal and the modified downmix signal, the first mixing matrix adapted to
perform a n-channel linear combination of the m-channel downmix signal and modified downmix signal only and for outputting
the n-channel upmix signal, wherein:
the first mixing matrix is adapted to receive one or more mixing parameters for controlling at least one gain in the linear
combination performed by the first mixing matrix:

and where the mixing parameters are in quantized format; and wherein
the n-channel upmix signal comprises a set of channels that are obtained as linear combinations of both the downmix signal
and the modified downmix signal; and wherein

in the linear combination performed by the first mixing matrix, all gains applied in order to obtain said set of channels
are polynomials of one or more of the mixing parameters, wherein the order of each polynomial is less than or equal to 2.

1. An encoding device, comprising one or more hardware elements, configured to generate an encoded bitstream comprising a
content frame and an associated metadata frame; wherein the content frame is indicative of a signal encoded according to a
first codec system; wherein the encoding device is configured to
generate a block of metadata;
insert the block of metadata into the metadata frame;
select a secure key from a plurality of pre-determined secure keys; wherein a first key of the plurality of pre-determined
secure keys provides a first level of trust, and a second key of the plurality of pre-determined secure keys provides a second
level of trust that is different from the first level; wherein the first key is a highly secure key known to only to a first
set of entities; wherein the second key is a moderately secure key known to a second set of entities which includes the first
set of entities;

generate a cryptographic value based at least on the content frame, on the associated metadata frame and on the selected secure
key; and

insert the generated cryptographic value into the metadata frame.

2. A method for encoding a sign-data-hiding enabled data signal representative of at least one image split up into partitions,
the method comprising:
setting a partition of an image comprising a set of coefficients;
identifying a first non-zero coefficient within the set of coefficients;
identifying a last non-zero coefficient within the set of coefficients;
determining a count of coefficients from the first non-zero coefficient to the last non-zero coefficient including the first
and last non-zero coefficients; and

in response to determining that the count is greater than 4:
calculating, using the sum of the amplitude information of the non-zero coefficients, parity data;
calculating, using the sum of the amplitude information of the non-zero coefficients, parity data;
selecting a particular non-zero coefficient to be encoded into the bitstream without a sign bit;
when the parity is even and the sign of the particular non-zero coefficient is negative, modifying a coefficient between the
first and the last non-zero coefficient; and

transmitting the modified set of coefficients.

1. A method for gain adjustments of audio signals based on encoder-generated loudness information, the method comprising:
receiving, by an audio decoder operating in a specific playback environment different from a reference channel configuration,
an audio signal for the reference channel configuration, the audio signal including audio sample data and encoder-generated
loudness metadata, the encoder-generated loudness metadata comprising a plurality of portions of loudness metadata for a plurality
of playback environments, the plurality of portions of loudness metadata comprising one or more respective portions of loudness
metadata for each playback environment in the plurality of playback environments;

selecting one or more portions of specific loudness metadata, based on the specific playback environment, from among the plurality
of portions of loudness metadata for the plurality of playback environments, the one or more portions of specific loudness
metadata relating to the specific playback environment;

determining loudness adjustment gains from the one or more portions of specific loudness metadata for the specific playback
environment;

applying the loudness adjustment gains as a part of overall gains applied to the audio sample data to generate output audio
data.

1. A method for reconstructing a time/frequency tile of an audio scene with at least one audio object (Sn, n=NB+1, . . . , N), which is associated with positional metadata (xn, n=NB+1, . . . , N), and at least one bed channel (Sn, n=1, . . . , NB), the method comprising:
receiving a bitstream;
from the bitstream, extracting a downmix signal (Y) comprising M downmix channels, each of which comprises a linear combination
of one or more of the audio object(s) and the bed channel(s) (Ym=?n=1N dm,nSn, m=1, . . . , M) in accordance with downmix coefficients (dm,n, m=1, . . . , M, n=1, . . . , N),

wherein each of the NB?M bed channels is associated with a corresponding downmix channel;

from the bitstream, further extracting the positional metadata of the audio objects or the downmix coefficients; and
reconstructing a bed channel as the corresponding downmix channel after suppressing content representing at least one audio
object from the corresponding downmix channel, wherein the suppression is made either on the basis of a positional locator
(zm, m=1, . . . , M), with which the corresponding downmix channel is associated, and the extracted positional metadata of the
audio objects, or on the basis of the downmix coefficients;

wherein the bed channels are reconstructed by suppressing content representing so many audio objects that a signal energy
of a remaining content representing audio objects is below a predefined threshold.

15. A system configured to determine a first banded tonality value for a first frequency subband of an audio signal; wherein
the first banded tonality value is used for approximating a high frequency component of the audio signal based on a low frequency
component of the audio signal; wherein the system comprises:
a microprocessor; and
a memory,
wherein the microprocessor is configured to determine a set of transform coefficients in a corresponding set of frequency
bins based on a block of samples of the audio signal;

wherein the microprocessor is configured to determine a set of bin tonality values for the set of frequency bins using the
set of transform coefficients, respectively; and

wherein the microprocessor is configured to combine a first subset of two or more of the set of bin tonality values for two
or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby
yielding the first banded tonality value for the first frequency subband;
wherein
the microprocessor is further configured to determine a sequence of sets of transform coefficients based on a corresponding
sequence of blocks of the audio signal;

for a particular frequency bin, the sequence of sets of transform coefficients comprises a sequence of particular transform
coefficients;

determining the bin tonality value for the particular frequency bin comprises:
determining a sequence of phases based on the sequence of particular transform coefficients; and
determining a phase acceleration based on the sequence of phases; and
the bin tonality value for the particular frequency bin is a function of the phase acceleration.

1. A signal processing device for filtering and performing high frequency reconstruction of an audio signal, the signal processing
device comprising:
an analysis filter bank that receives real valued time domain input audio samples and generates complex valued subband samples;
a high frequency reconstructor that generates modified complex valued subband samples through a high frequency reconstruction
process; and

a synthesis filter bank that receives the modified complex valued subband samples and generates time domain output audio samples,
wherein the analysis filter bank comprises analysis filters (hk(n)) that are complex exponential modulated versions of a prototype filter (p0(n)) according to:
0?n where A is an arbitrary phase shift constant, the analysis filter bank has M channels, the prototype filter (p0(n)) has a length N, and the analysis filter bank and synthesis filter bank have a system delay of D samples,

wherein one or more of the analysis filter bank, the high frequency reconstructor, and the synthesis filter bank is implemented,
at least in part, by one or more hardware elements of the signal processing device.

1. An image decoding method comprising:
receiving a bitstream comprising one or more coding parameters used in decoding a target picture of the video sequence
deriving, based on the bitstream, a reference picture set to be applied to the target picture;
generating a reference picture list based on reference picture list (RPL) modification information decoded from a slice header
and the reference picture set;

determining a value of a first flag included in the one or more coding parameters, the first flag indicating whether information
regarding list sorting is present within the slice header; and

omitting decoding of a part of information included in the RPL modification information,
wherein the RPL modification information includes at least a reference picture list sorting presence or absence flag and a
reference list sorting order, and

wherein decoding of the reference picture list sorting presence or absence flag and the reference list sorting order is omitted
based on the number of current picture referable pictures and the value of the first flag.

1. A method, performed by an audio signal processing device, for resampling a sequence of metadata instances representing
time-varying rendering metadata in an object-based audio system, wherein each metadata instance:
specifies a desired rendering state;
is associated with a time stamp, the time stamp indicating a point in time to begin a transition from a current rendering
state to the desired rendering state; and

includes one or more parameters indicative of the time stamp and an interpolation duration parameter indicating the required
time to reach the desired rendering state;

the method comprising:
receiving or generating the sequence of metadata instances;
generating one or more additional metadata instances; and
inserting the one or more additional metadata instances between a first and a second metadata instance of the sequence of
metadata instances to generate the resampled metadata sequence;

wherein the one or more additional metadata instances are substantially similar to the first metadata instance and/or the
second metadata instance, with the exception of the interpolation duration parameter, which is different than the interpolation
duration parameters of the first metadata instance and/or the second metadata instance; and

wherein the desired rendering state is determined by converting the metadata instance into coefficients specifying gain factors
for playback of audio content through audio drivers in a playback system.

1. A teleconferencing method in which nodes perform audio coding to generate spatially layered encoded audio, the nodes include
endpoints, and at least some of the spatially layered encoded audio is transmitted from one of the nodes to at least another
one of the nodes, wherein the nodes include a first node which is configured to generate spatially layered encoded audio in
response to soundfield audio data, said encoded audio including any of a number of different subsets of a set of layers, said
set of layers including at least one monophonic layer and at least one soundfield layer, said method including steps of:
(a) in the first node, determining a first subset of the set of layers by performing at least one of perceptually-driven layer
selection or endpoint-driven layer selection, said first subset including at least one of said monophonic layer or said soundfield
layer, wherein said endpoint-driven layer selection includes at least one independent decision by at least one of the endpoints
based on at least one analyzed characteristic of said at least one of the endpoints or of audio content captured by said at
least one of the endpoints, and wherein said perceptually-driven layer selection is not based on any downstream capability
consideration; and

(b) in said first node, generating first spatially layered encoded audio, wherein the first spatially layered encoded audio
includes the first subset of the set of layers determined in step (a), and wherein the first spatially layered encoded audio
does not include any layer of said set of layers which is not included in said first subset of the set of layers determined
in step (a).

1. A pitch filter for filtering a preliminary audio signal generated from an audio bitstream, the pitch filter having an operating
mode selected from one of either:
(i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal,
and

(ii) an inactive mode where the pitch filter is disabled;
wherein the preliminary audio signal is generated in an audio decoder operating in a coding mode selected from at least two
distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive
mode based on control information while the audio decoder is operating in the coding mode, and

wherein the control information is a parameter one bit in length, and a first value of the parameter indicates that the pitch
filter should be operated in the active mode and a second value of the parameter indicates that the pitch filter should be
operated in the inactive mode.

1. A method, performed by an audio signal processing device, for estimating a tempo metric related to an audio signal based
on an encoded bit-stream representing the audio signal, wherein the bit-stream includes a plurality of audio blocks, the method
comprising:
receiving the bit-stream;
analyzing the bit-stream to detect transitions in block sizes of said audio blocks in the bit-stream;
determining at least one periodicity related to a re-occurrence of said detected transitions; and
determining an estimated tempo metric based on the determined periodicity;
wherein one or more of receiving the bit-stream, detecting transitions, determining at least one periodicity, and determining
an estimated tempo metric are implemented, at least in part, by one or more hardware elements of the audio signal processing
device.

1. A method for encoding audio objects as a data stream, comprising:
receiving N audio objects associated with time-variable spatial positions, wherein N>1;
calculating M downmix signals, wherein M?N, by forming combinations of the N audio objects;
calculating time-variable side information including parameters which allow reconstruction of a set of audio objects formed
on the basis of the N audio objects from the M downmix signals, wherein the audio objects in said set of audio objects are
associated with time-variable spatial positions; and

including the M downmix signals and the side information in a data stream for transmittal to a decoder,
wherein the method further comprises including, in the data stream:
a plurality of side information instances specifying respective desired reconstruction settings for reconstructing said set
of audio objects formed on the basis of the N audio objects; and

for each side information instance, transition data including two independently assignable portions which in combination define
a point in time to begin a transition from a current reconstruction setting to the desired reconstruction setting specified
by the side information instance, and a point in time to complete the transition.

5. A method for Higher Order Ambisonics (HOA) decoding comprising:
receiving information regarding direction values (?l) of loudspeakers and a decoder Ambisonics order (N1);

determining ket vectors (|Y(?l)) of spherical harmonics for loudspeakers located at directions corresponding to the direction values (?l) and a decoder mode matrix (?o×L) based on the direction values (?l) of loudspeakers and the decoder Ambisonics order (Nl);

determining two corresponding decoder unitary matrices (Ul†, Vl) and a decoder diagonal matrix (?l) containing singular values and a final rank (rfind) of the decoder mode matrix (?o×L) based on a Singular Value Decomposition of the decoder mode matrix (?o×L);

determining a final mode matrix rank (rfin) based on the final encoder mode matrix rank (rfine) and the final decoder mode matrix rank (rfind);

determining an adjoint pseudo inverse (?+)† of the encoder mode matrix (?o×s), resulting in an Ambisonics ket vector (|a?s), based on the encoder unitary matrices (Us, Vs†), the encoder diagonal matrix (?s) and the final mode matrix rank (rfin);

determining an adapted Ambisonics ket vector (|a?l) based on a reduction of a number of components of the Ambisonics ket vector (|a?s) according to the final mode matrix rank (rfin);

determining an adjoint decoder mode matrix (?)†, resulting in a ket vector (|y(?l)) of output signals for all loudspeakers, based on the adapted Ambisonics ket vector (|a?l), the decoder unitary matrices (Ul†, Vl), the decoder diagonal matrix (?l) and the final mode matrix rank.

1. A signal processing device for filtering and processing an audio signal, the signal processing device comprising:
an analysis filter bank that receives real valued time domain input audio samples and generates complex-valued subband samples;
a phase shifter that shifts a phase of the complex-valued subband samples by an amount;
a high frequency reconstructor or a parametric stereo processor that generates modified complex-valued subband samples;
a phase shifter that unshifts a phase of the modified complex-valued subband samples by the amount; and
a synthesis filter bank that receives the modified complex-valued subband samples and generates time domain output audio samples,
wherein the analysis filter bank comprises analysis filters (hk(n)) and the synthesis filter bank comprises synthesis filters (fk(n)) that are complex exponential modulated versions of a prototype filter (p0(n)) according to:


where M is a number of channels, the prototype filter (p0(n)) has a length N, and the analysis filter bank and synthesis filter bank have a system delay of D samples,

wherein one or more of the analysis filter bank, phase shifter, the high frequency reconstructor, parametric stereo processor,
and the synthesis filter bank is implemented, at least in part, by one or more hardware elements of the signal processing
device, and

wherein the complex valued subband samples are oversampled by a factor of two.

1. A method for decoding an encoded audio bitstream, the method comprising:
receiving the encoded audio bitstream, the encoded audio bitstream containing a lowband audio signal and envelope data;
extracting and decoding the lowband audio signal from the encoded audio bitstream to generate a decoded lowband audio signal;
extracting and decoding the envelope data from the encoded audio bitstream to generate decoded spectral envelope data;
filtering the decoded lowband signal with an analysis filterbank to produce lowband subband signals, wherein the analysis
filterbank is maximally decimated;

generating a highband signal by copying a number of lowband subband signals from consecutive lowband channels to consecutive
highband channels to form a patch, wherein the generating is performed more than once so as to produce more than one patch;

adjusting a spectral envelope of the highband signal using the decoded spectral envelope data;
filtering the lowband subband signals and the highband signal with a synthesis filterbank to produce a digital wideband output
audio signal, wherein a number of channels of the synthesis filterbank is different than a number of channels of the analysis
filterbank,

wherein the generating further comprises frequency translating a complex subband signal in a source area channel having an
index i to a complex subband signal in a reconstruction range channel having an index j and frequency translating a complex
subband signal in a source area channel having an index i+1 to a complex subband signal in a reconstruction range channel
having an index j+1.

1. A method for reconstructing a wideband audio signal, the method comprising:
decomposing a lowband audio signal into a plurality of complex subband signals with an analysis filterbank, each of the plurality
of complex subband signals representing a frequency channel of the analysis filterbank;

generating a highband audio signal by patching a number of consecutive complex subband signals, wherein the generating includes:
frequency translating a complex subband signal in a source area channel of the lowband audio signal having an index i to a
reconstruction range channel having an index j of the highband audio signal, and

frequency translating a complex subband signal in a source area channel of the lowband audio signal having an index i+1 to
a reconstruction range channel having an index j+1 of the highband audio signal;

adjusting a spectral envelope of the highband audio signal to a desired level;
combining the lowband audio signal and the highband audio signal with a synthesis filterbank to generate the wideband audio
signal,

wherein the lowband audio signal has frequency components below a crossover frequency and the highband audio signal has frequency
components above the crossover frequency, and

wherein the analysis filterbank and the synthesis filterbank each have L channels, and L is an integer value.

1. A method of decoding a stream representative of at least one coded image, the method comprising:
receiving the stream representative of at least one coded image;
identifying, from the stream, a predetermined plurality of groups of blocks;
processing a first block in a given group of blocks, wherein the processing of the first block comprises:
determining that the first block is first in an order of blocks in the given group of blocks;
in response to determining that the first block is first in the order of blocks in the given group of blocks, entropy decoding
the first block based on a first set of probability data, wherein the first set of probability data comprises a first set
of probabilities of occurrence of symbols associated with a block that is situated immediately adjacent to the first block
and that belongs to another group of blocks that is different from the given group of blocks in the predetermined plurality
of groups of blocks, wherein two blocks are situated immediately adjacent to one another when the two blocks share a spatial
boundary in an image;

dequantizing the first block; and
inverse transforming the first block; and
processing a second block in the given group of blocks, wherein the processing of the second block comprises:
determining that the second block is not first in the order of blocks in the given group of blocks;
in response to determining that the second block is not first in the order of blocks in the given group of blocks, entropy
decoding the second block based on a second set of probability data, wherein the second set of probability data comprises
a second set of probabilities of occurrence of symbols associated with at least one other already decoded block belonging
to the given group of blocks in the predetermined plurality of groups of blocks, wherein the second set of probabilities of
occurrence of symbols are not associated with blocks that do not belong to the given group of blocks;

dequantizing the second block; and
inverse transforming the second block.

1. A method of coding at least one image, the method comprising:
cutting the at least one image into a plurality of blocks;
grouping the plurality of blocks into a predetermined number of groups of blocks; and
coding each of the groups of blocks, comprising:
for a first block of a given group of blocks:
transforming the first block;
determining that the first block is first in an order of blocks in the given group of blocks; and
in response to determining that the first block is first in the order of blocks in the given group of blocks, entropy coding
the first block based on a first set of probability data, wherein the first set of probability data comprises a first set
of probabilities of occurrence of symbols associated with a block that is situated immediately adjacent to the first block
and that belongs to another group of blocks that is different from the given group of blocks in the predetermined plurality
of groups of blocks;

for a second block of the given group of blocks:
transforming the second block;
determining that the second block is not first in the order of blocks in the given group of blocks; and
in response to determining that the second block is not first in the order of blocks in the given group of blocks, entropy
coding the second block based on a second set of probability data, wherein the second set of probability data comprises a
second set of probabilities of occurrence of symbols associated with at least one other already coded block belonging to the
given group of blocks in the predetermined plurality of groups of blocks, wherein the second set of probabilities of occurrence
of symbols are not associated with blocks that do not belong to the given subset of blocks.

1. A spatial adaptation system comprising:
a frame power module configured to determine a determined frame power based on at least a converted front microphone signal;
a posterior signal to noise ratio module configured to determine a determined posterior signal to noise ratio that represents
a signal to noise ratio of a noise source based on said converted front microphone signal, wherein the determined posterior
signal to noise ratio is a temporal feature;

an inference and weight module configured to receive a plurality of inputs based on two or more input signals captured by
at least two microphones, said plurality of inputs including the determined frame power and the determined posterior signal
to noise ratio, said inference and weight module configured to determine one or more noise target weights based on at least
said determined posterior signal to noise ratio;

a noise magnitude ratio update module coupled with said inference and weight module, said noise magnitude ratio update module
configured to receive said one or more noise target weights from said inference and weight module and configured to determine
an updated noise target value based on said one or more noise target weights from said inference and weight module, said updated
noise target value used to adapt a power level of at least one of said two or more input signals captured by said at least
two microphones; and

a spatial feature module coupled with said inference and weight module, said spatial feature module to determine one or more
spatial features based on said two or more input signals,

wherein said inference and weight module determines one or more noise target weights based on said one or more spatial features
determined by said spatial feature module.

1. A non-transitory digital storage medium having stored thereon a bitstream comprising a configuration block and a sequence
of frames respectively representing consecutive time periods of an audio content, wherein the sequence of frames is a composition
of N sequences of frame elements with each frame element being of a respective one of a plurality of element types so that
each frame comprises one frame element out of the N sequences of frame elements, respectively, and for each sequence of frame
elements, the frame elements are of equal element type relative to each other,
wherein the configuration block comprises, for at least one of the sequences of frame elements, a default payload length information
on a default payload length, and

wherein each frame element of the at least one of the sequences of frame elements, comprises a length information comprising,
for at least a subset of the frame elements of the at least one of the sequences of frame elements, a default payload length
flag followed, if the default payload length flag is not set, by a payload length value,

wherein any frame element of the at least one of the sequences of frame elements, the default payload length flag of which
is set, comprises the default payload length, and any frame element of the at least one of the sequences of frame elements,
the default payload length flag of which is not set, comprises a payload length corresponding to the payload length value.

1. A decoding system (100) for reconstructing an n-channel audio signal, wherein the decoding system is adapted to receive an input signal segmented
into time frames and representing the audio signal, in a given time frame, according to a coding regime selected from the
group comprising:
parametric coding (P; P(I), P(P)), in which the input signal comprises m channels and at least one mixing parameter (?), where
n>m?1; and

discrete coding (D; Discr.), in which the input signal comprises the n channels discretely encoded,
the decoding system being operable to derive the audio signal at least
in a parametric mode (PM; 1301) of the decoding system, by spatial synthesis guided by said at least one mixing parameter from a current frame, and,

in a discrete mode (DM; 1303) of the decoding system, on the basis of said n discretely encoded channels,

the decoding system comprising a controller (170) for controlling the mode of the decoding system on the basis of at least a current mode of the decoding system and a current
received frame of the input signal, such as by performing a mode transition into a mode corresponding to a coding regime of
the current received frame,

wherein the controller is configured to respond to receipt of a defective frame (Def.), when in the parametric mode, by entering
a keep mode (KM; 1302), in which the decoding system derives the audio signal by spatial synthesis taking as input m channels from the defective
frame and being guided by at least one mixing parameter from a previous frame, wherein

the decoding system (100) is adapted to receive the input signal representing the audio signal according to parametric coding by either an independent
frame (P(I)) or a predicted frame (P(P)), wherein said at least one mixing parameter in a current predicted frame is decodable
only after decoding a preceding independent frame,

wherein the controller is further configured to respond to receipt of a predicted frame, when in the keep mode, by remaining
in the keep mode.

1. A pitch filter for filtering a preliminary audio signal generated from an audio bitstream, the pitch filter having an operating
mode selected from one of either:
(i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal,
and

(ii) an inactive mode where the pitch filter is disabled;
wherein the preliminary audio signal is generated in an audio decoder operating in a coding mode selected from at least two
distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive
mode based on control information while the audio decoder is operating in the coding mode, wherein the audio bitstream is
segmented into frames of audio content and the control information includes a frame type parameter with one or more first
values of the frame type parameter indicating that the pitch filter should be operated in the active mode and a second value
of the parameter indicating that the pitch filter should be operated in the inactive mode.

1. An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal
representation comprising a sampling frequency information, an encoded time warp information and an encoded spectrum representation,
the audio signal decoder comprising:
a time warp calculator configured to map the encoded time warp information onto a decoded time warp information,
wherein the time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information
onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information;
and

a warp decoder configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation
and in dependence on the decoded time warp information;

wherein the audio signal decoder is implemented using a hardware apparatus, or using a computer, or using a combination of
a hardware apparatus and a computer.

1. A method for low bit rate parametric encoding and transport of haptic-tactile signals, the method comprising:at a computing device comprising one or more processors and memory storing one or more programs executed by the one or more processors to perform the method, performing operations comprising:
for at least one frame of a plurality of frames of a source haptic-tactile signal, representing the source haptic-tactile signal in the frame as a set of parameters according to a functional representation, wherein the source haptic-tactile signal represents haptic or tactile interaction with a physical environment; and
including the set of parameters in a bit stream;
wherein the functional representation is based on one of a set of orthogonal functionals, or polynomial approximation.

1. A method, comprising:receiving mixed audio content, in a reference audio channel representation, that are distributed over a plurality of audio channels of the reference audio channel representation, the mixed audio content having a mix of speech content and non-speech audio content;
transforming one or more portions of the mixed audio content that are distributed over two or more non-Mid/Side (non-M/S) channels in the plurality of audio channels of the reference audio channel representation into one or more portions of the transformed mixed audio content in an M/S audio channel representation that are distributed over one or more channels of the M/S audio channel representation, wherein the M/S audio channel representation comprises at least a mid-channel signal and a side-channel signal, wherein the mid-channel signal represents a weighted or non-weighted sum of two channels of the reference audio channel representation, and wherein the side-channel signal represents a weighted or non-weighted difference of two channels of the reference audio channel representation;
determining metadata for speech enhancement of the one or more portions of the transformed mixed audio content in the M/S audio channel representation, wherein a first type of speech enhancement is waveform-encoded speech enhancement of a reduced quality version of the mid-channel signal in the M/S audio channel representation, and a second type of speech enhancement is parametric-encoded speech enhancement of a reconstructed version of the mid-channel signal in the M/S audio channel representation, the metadata including a mid-channel prediction parameter to reconstruct the mid-channel signal, a first gain parameter for waveform-encoded speech enhancement of the mid-channel signal, and a second gain parameter for parametric-encoded speech enhancement of the reconstructed mid-channel signal; and
generating an audio signal that comprises the mixed audio content and the metadata for speech enhancement of the one or more portions of the transformed mixed audio content in the M/S audio channel representation;
wherein the method is performed by one or more computing devices.

1. A method, performed by an audio and/or video signal processing device, for processing a bitstream; wherein the bitstream comprises a sequence of encoded frames; wherein the frames of the sequence of encoded frames comprise a varying number of bits; wherein the frames of the sequence of encoded frames correspond to excerpts of an audio and/or video signal, wherein the excerpts have a constant temporal length; wherein at least two frames of the sequence of encoded frames comprise a wait_frames parameter; wherein the wait_frames parameter of a frame is indicative of a number of frames and/or a time interval that the corresponding frame is to be delayed in a buffer of the audio and/or video signal processing device prior to processing of the frame by the audio and/or video signal processing device; the method comprisingdetermining a total number of bits STot for N frames of a subsequence of encoded frames from the bitstream;
determining a corrected number N? of frames based on the number N of frames and based on a difference between the wait_frames parameters of a first frame and a second frame of the subsequence; wherein the first frame corresponds to the frame at a beginning of the subsequence; and wherein the second frame corresponds to the frame at an end of the subsequence;
determining a lower bitrate bound brmin and an upper bitrate bound brmax of a bitrate br of the bitstream based on the total number of bits STot, based on the corrected number N? of frames and based on a frame rate fframe of the bitstream;
determining an estimate brest of a bitrate br of the bitstream from the lower bitrate bound and an upper bitrate bound;
storing frames of the sequence of encoded frames in the buffer of the audio and/or video signal processing device;
processing the frames of the sequence of encoded frames, wherein processing the frames of the sequence of encoded frames comprises inserting the encoded frames into a multiplexed bitstream or decoding the sequence of encoded frames; and
outputting, from the audio and/or video signal processing device, the multiplexed bitstream or the decoded frames;
wherein one or more of determining a total number of bits, determining a number N? of frames, determining a lower bitrate bound brmin and an upper bitrate bound brmax, determining an estimate of the bitrate, storing frames of the sequence, processing the frames of the sequence, and outputting the multiplexed bitstream or the decoded frames is implemented, at least in part, by one or more hardware elements of the audio and/or video signal processing device.

1. A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels, the method comprising:receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter;
decoding the encoded audio signal having M audio channels to obtain a decoded representation of the M audio channels;
decorrelating at least a portion of the decoded representation with an all-pass filter to obtain M decorrelated signals, the all-pass filter including a plurality of filter links, wherein a transfer function H(z) in a Z-domain of at least some of the plurality of filter links is at least partially derivable from or based on:

where q is a complex valued phase rotation factor, m is a delay length and a is a filter coefficient;
reconstructing N audio channels from the M decorrelated signals and the decoded representation of the M audio channels to obtain N audio signals that collectively having N audio channels, wherein N is two or more, M is one or more, and M is less than N; and
synthesizing the N audio signals with one or more synthesis filterbanks to convert the N audio signals from a frequency domain to a time domain;
wherein the inter-channel intensity difference parameter is delta coded over frequency and the audio decoder is implemented at least in part with hardware.

1. A method for decoding video, the method comprising:(a) receiving quantized coefficients representative of a block of video representative of a plurality of pixels;
(b) descaling the quantized coefficients by multiplying the quantized coefficients with numbers dependent on a coefficient index and a transform size of the block to generate descaled coefficients;
(c) applying an adjustment to the descaled coefficients to generate adjusted descaled coefficients, wherein the adjustment is a variable based on the transform size;
(d) clipping the adjusted descaled coefficients to a predetermined bit depth to generate clipped coefficients;
(e) one-dimensional inverse transforming the clipped coefficients in a first direction to generate first direction inverse transformed coefficients;
(f) shifting the first direction inverse transformed coefficients to generate shifted coefficients;
(g) clipping the shifted coefficients to the predetermined bit depth to generate second clipped coefficients; and
(h) one-dimensional inverse transforming the second clipped coefficients in a second direction to determine a decoded residue,
wherein the applying the adjustment is performed after the descaling, and
wherein the shifting is performed after the one-dimensional inverse transforming in the first direction and prior to the one-dimensional inverse transforming in the second direction.

1. An audio decoding method comprising:receiving a two-channel downmix signal and upmix parameters for parametric reconstruction of an M-channel audio signal having a predefined channel configuration based on the downmix signal, where M?4;
receiving signaling indicating a selected one of at least two coding formats of the M-channel audio signal having a predefined channel configuration, wherein the indicated selected coding format switches between the at least two coding formats, and wherein the coding formats correspond to respective different partitions of the channels of the predefined channel configuration of the M-channel audio signal into respective first and second groups of one or more channels, wherein, in the indicated coding format, a first channel of the downmix signal corresponds to a linear combination of the first group of one or more channels of the predefined channel configuration of the M-channel audio signal and a second channel of the downmix signal corresponds to a linear combination of the second group of one or more channels of the predefined channel configuration of the M-channel audio signal;
determining a set of pre-decorrelation coefficients based on the indicated coding format;
computing a decorrelation input signal as a linear mapping of the downmix signal, wherein the set of pre-decorrelation coefficients is applied to the downmix signal, wherein the pre-decorrelation coefficients are determined such that a first channel of the predefined channel configuration of the M-channel audio signal contributes, via the downmix signal, to a first fixed channel of the decorrelation input signal in at least two of the coding formats;
generating a decorrelated signal based on the decorrelation input signal;
determining sets of wet and dry upmix coefficients based on the received upmix parameters and the indicated coding format;
computing a dry upmix signal as a linear mapping of the downmix signal, wherein the set of dry upmix coefficients is applied to the downmix signal;
computing a wet upmix signal as a linear mapping of the decorrelated signal, wherein the set of wet upmix coefficients is applied to the decorrelated signal; and
combining the dry and wet upmix signals to obtain a multidimensional reconstructed signal corresponding to the M-channel audio signal to be reconstructed.

1. An apparatus comprising:a decoder for decoding a motion vector predictor of a current block in a picture of a sequence of pictures, wherein the decoder is configured to:
identify a first adjacent block that is adjacent to the current block in the picture;
identify a second adjacent block that is adjacent to the current block in the picture;
when a motion vector of the first adjacent block is not equal to a motion vector of the second adjacent block, generate a motion vector predictor candidate set comprising the motion vectors of both the first and the second adjacent blocks;
when the motion vector of the first adjacent block is equal to the motion vector of the second adjacent block, generate a motion vector predictor candidate set comprising only the motion vector of the first adjacent block or the motion vector of the second adjacent block;
receive a flag from a bitstream indicating whether a temporally-located motion vector can be used as a motion vector predictor;
when the flag indicates that a temporally-located motion vector can be used as a motion vector predictor, include a motion vector of a block in another picture in the motion vector predictor candidate set;
when the flag indicates that a temporally-located motion vector cannot be used as a motion vector predictor, exclude the motion vector of the block in another picture from the motion vector predictor candidate set; and
select a motion vector from the motion vector predictor candidate set as the motion vector predictor of the current block,
wherein a motion vector for the current block is derived based on the selected motion vector predictor and a motion vector differential.

1. An audio processing device including a subband processing unit configured to determine a synthesis subband signal from
a first and a second analysis subband signal; wherein the first and the second analysis subband signal each comprise a plurality
of complex valued analysis samples at different times, referred to as the first and second analysis samples, respectively,
each analysis sample having a phase and a magnitude; wherein the first and second analysis subband signals are associated
with respective frequency bands of an input audio signal; wherein the subband processing unit comprises
a first block extractor configured to repeatedly
derive a frame of L first input samples from the plurality of first analysis samples; the frame length L being greater than
one; and

apply a block hop size of p samples to the plurality of first analysis samples, prior to deriving a next frame of L first
input samples;

thereby generating a suite of frames of L first input samples;
a second block extractor configured to derive a suite of second input samples by applying the block hop size p to the plurality
of second analysis samples; wherein each second input sample corresponds to a frame of first input samples;

a nonlinear frame processing unit configured to determine a frame of processed samples from a frame of first input samples
and from the corresponding second input sample, by determining for each processed sample of the frame:

the phase of the processed sample by offsetting the phase of the corresponding first input sample; and
the magnitude of the processed sample based on the magnitude of the corresponding first input sample and the magnitude of
the corresponding second input sample; and

an overlap and add unit configured to determine the synthesis subband signal by overlapping and adding the samples of a suite
of frames of processed samples; wherein the synthesis subband signal is associated with a frequency band of a signal which
is time stretched and/or frequency transposed with respect to the input audio signal, wherein one or more of the first block
extractor, the second block extractor, the nonlinear frame processing unit, and the overlap and add unit is implemented, at
least in part, by one or more hardware elements of the audio processing device.

1. An audio encoder configured to encode a frame of a soundfield signal comprising a plurality of audio signals, the audio
encoder comprising—a transform determination unit configured to determine an energy-compacting orthogonal transform based
on the frame of the soundfield signal; —a transform unit configured to apply the energy-compacting orthogonal transform to
a frame derived from the frame of the soundfield signal, and to provide a frame of a rotated soundfield signal comprising
a plurality of rotated audio signals;
a waveform encoding unit configured to encode a first rotated audio signal, but not a second rotated audio signal, of the
plurality of rotated audio signals; and

a parametric encoding unit configured to determine and encode a set of spatial parameters for determining the second rotated
audio signal of the plurality of rotated audio signals based on the first rotated audio signal, wherein the set of spatial
parameters enables a corresponding decoder to estimate at least one of a correlated component or a decorrelated component
of the second rotated audio signal based on the first rotated audio signal.

1. A method for reconstructing a time/frequency tile of N audio objects, comprising the steps of:
receiving M downmix signals;
receiving a reconstruction matrix enabling reconstruction of an approximation of the N audio objects from the M downmix signals;
applying the reconstruction matrix to the M downmix signals in order to generate N approximated audio objects;
subjecting at least a subset of the N approximated audio objects to a decorrelation process in order to generate at least
one decorrelated audio object, whereby each of the at least one decorrelated audio object corresponds to one of the N approximated
audio objects;

for each of the N approximated audio objects not having a corresponding decorrelated audio object, reconstructing a time/frequency
tile of the audio object by the approximated audio object; and

for each of the N approximated audio objects having a corresponding decorrelated audio object, reconstructing the time/frequency
tile of the audio object by:

receiving a single weighting parameter from which a first weighting factor and a second weighting factor are derivable,
weighting the approximated audio object by the first weighting factor,
weighting the decorrelated audio object corresponding to the approximated audio object by the second weighting factor, and
combining, by performing a summation, the weighted approximated audio object with the corresponding weighted decorrelated
audio object for reconstructing the time/frequency tile of the approximated audio object, whereby an energy level of the reconstructed
time/frequency tile equals an energy level of a corresponding time/frequency tile of the approximated audio object.

1. A system configured to generate a time stretched and/or frequency transposed signal from an input signal, the system comprising:
an analysis filter bank configured to derive a number Y?1 of analysis subband signals from the input signal, wherein each
analysis subband signal comprises a plurality of complex-valued analysis samples, each having a phase and a magnitude;

a subband processing unit configured to generate a synthesis subband signal from the Y analysis subband signals using a subband
transposition factor Q and a subband stretch factor S, at least one of Q and S being greater than one, wherein the subband
processing unit comprises:

a block extractor configured to:
i) form Y frames of L input samples, each frame being extracted from said plurality of complex-valued analysis samples in
an analysis subband signal and the frame length being L>1; and

ii) apply a block hop size of h samples to said plurality of analysis samples, prior to forming a subsequent frame of L input
samples, thereby generating a sequence of frames of input samples;

a nonlinear frame processing unit configured to generate, on the basis of Y corresponding frames of input samples formed by
the block extractor, a frame of processed samples by determining a phase and magnitude for each processed sample of the frame,
wherein, for at least one processed sample:

i) the phase of the processed sample is based on a linear combination, with non-negative integer coefficients, of respective
phases of the corresponding input sample in a first and second frame of the Y frames of input samples; and

ii) the magnitude of the processed sample is based on the magnitude of the corresponding input sample in each of the Y frames
of input samples; and

an overlap and add unit configured to determine the synthesis subband signal by overlapping and adding the samples of a sequence
of frames of processed samples; and

a synthesis filter bank configured to generate the time stretched and/or frequency transposed signal from the synthesis subband
signal, wherein the system is operable at least for Y=2.

1. A method for encoding audio objects as a data stream, comprising:
receiving N audio objects, wherein N>1;
calculating M downmix signals, wherein M?N, by forming combinations of the N audio objects;
calculating time-variable side information including parameters which allow reconstruction of a set of audio objects formed
on the basis of the N audio objects from the M downmix signals; and

including the M downmix signals and the side information in a data stream for transmittal to a decoder, wherein the data stream
corresponds to a plurality of time frames,

wherein the method further comprises including, in the data stream:
a plurality of side information instances specifying respective desired reconstruction settings for reconstructing said set
of audio objects formed on the basis of the N audio objects; and

for each side information instance, transition data including two independently assignable portions which in combination define
a point in time to begin a transition from a current reconstruction setting to the desired reconstruction setting specified
by the side information instance, and a point in time to complete the transition, and wherein for each specific side information
instance of the plurality of side information instances:

the point in time defined by the transition data of the specific side information instance for beginning a transition corresponds
to a first of the plurality of time frames, wherein the point in time defined by the transition data of the specific side
information instance for completing a transition corresponds to a second of the plurality of time frames,

the second time frame is either the same as the first time frame or subsequent to the first time frame.

1. An audio encoding device that generates a bitstream indicative of a downmix signal and spatial metadata for generating
a multi-channel upmix signal from the downmix signal; wherein the audio encoding device:
generates the downmix signal from a multi-channel input signal; wherein the downmix signal comprises m channels and wherein
the multi-channel input signal comprises n channels; n, m being integers with m
determines the spatial metadata from the multi-channel input signal; and
determines one or more control settings for the parameter processing unit based on one or more external settings; wherein
the one or more external settings comprise a target data-rate for the bitstream and one or more of: a sampling rate of the
multi-channel input signal, the number m of channels of the downmix signal, the number n of channels of the multi-channel
input signal, and an update period indicative of a time period required by a corresponding decoding system to synchronize
to the bitstream; and wherein the one or more control settings comprise a maximum data-rate for the spatial metadata and one
or more of: a temporal resolution setting indicative of a number of sets of spatial parameters per frame of spatial metadata
to be determined, a frequency resolution setting indicative of a number of frequency bands for which spatial parameters are
to be determined, a quantizer setting indicative of a type of quantizer to be used for quantizing the spatial metadata, and
an indication whether a current frame of the multi-channel input signal is to be encoded as an independent frame.

1. A signal processing device for filtering and processing an audio signal, the signal processing device comprising:
an analysis filter bank that receives real valued time domain input audio samples and generates complex valued subband samples;
a high frequency reconstructor or parametric stereo processor that generates modified complex valued subband samples; and
a synthesis filter bank that receives the modified complex valued subband samples and generates time domain output audio samples,
wherein the analysis filter bank comprises analysis filters (hk(n)) that are complex exponential modulated versions of a prototype filter (p0(n)) according to:


where A is a phase shift, the analysis filter bank has M channels, the prototype filter (p0(n)) has a length N, and the analysis filter bank and synthesis filter bank have a system delay of D samples,

wherein one or more of the analysis filter bank, the high frequency reconstructor, the parametric stereo processor, and the
synthesis filter bank is implemented, at least in part, by one or more hardware elements of the signal processing device.

1. A method for reconstructing a wideband audio signal, the method comprising:
decomposing a lowband audio signal into a plurality of complex subband signals with an L-channel analysis filterbank, each
of the plurality of complex subband signals representing a frequency channel of the analysis filterbank;

generating a highband audio signal by patching a number of consecutive complex subband signals, wherein the generating includes:
frequency translating a complex subband signal in a source area channel of the lowband audio signal having an index i to a
reconstruction range channel having an index j of the highband audio signal, and

frequency translating a complex subband signal in a source area channel of the lowband audio signal having an index i+1 to
a reconstruction range channel having an index j+1 of the highband audio signal;

adjusting a spectral envelope of the highband audio signal to a desired level;
combining the lowband audio signal and the highband audio signal with a Q·L-channel synthesis filterbank to generate the wideband
audio signal,

wherein the lowband audio signal has frequency components below a crossover region and the highband audio signal has frequency
components above the crossover region, and

wherein Q is chosen so that Q·L is an integer value.

1. A method for encoding a sequence of pictures, the method comprising:
receiving the sequence of pictures;
generating a base layer of coded pictures based on the sequence pictures;
generating at least one enhancement layer of coded pictures based on the sequence of pictures;
generating a video syntax set that includes information applicable to coded pictures including a flag indicating whether a
scalability dimension identifier is signaled implicitly or explicitly, and wherein the scalability dimension identifier specifies
a scalability dimension of a particular layer of coded pictures, the scalability dimension being one of multiple types, including:
a spatial type and a quality type;

generating one or more network abstraction layer (NAL) units representing the base layer and the enhancement layer; andin case the flag indicates that the scalability dimension identifier is implicitly signaled, signaling the scalability dimension
identifier implicitly in a header of a NAL unit representing the enhancement layer,
wherein the header of a NAL unit includes a layer identifier indicating whether the NAL unit belongs to the base layer or
the enhancement layer.

1. A method for a decoder for decoding a plurality of input audio signals for playback on a speaker configuration with N channels,
the plurality of input audio signals representing encoded multichannel audio content corresponding to K?N channels, comprising:
from the encoded multichannel audio content corresponding to K channels, extracting M input audio signals, wherein 1 wherein if N=M, the method further comprises the step of:
discarding any remaining signals in the encoded multichannel audio content;
decoding, in a first decoding module, the M input audio signals into M mid signals which are suitable for playback on a speaker
configuration with M channels;

wherein if N>M, the method further comprises the steps of:
from the encoded multichannel audio content corresponding to K channels, extracting N-M additional input audio signals, wherein
each of the additional input audio signals corresponds to one of the M mid signals and is either a side signal or a complementary
signal which together with the mid signal to which it corresponds and a weighting parameter a allows reconstruction of a side
signal; and for each of the N channels in excess of M channels

decoding, in a stereo decoding module, the additional input audio signal and the mid signal to which it corresponds so as
to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels
of the speaker configuration;
whereby N audio signals are generated.

1. Method for decoding stereo loudspeaker signals l(t) from a three-dimensional spatial higher-order Ambisonics audio signal
a(t), with t designating time, from azimuth angle values ?L and ?R of left and right loudspeakers, and from S sampling points on a circle, said method including the steps:
receiving said audio signal a(t),
calculating by at least one processor, from azimuth angle values ? of left and right loudspeakers and from the number S of
virtual sampling points on a circle, a matrix G containing desired panning function values for all virtual sampling points,

wherein
and the gL(?) and gR(?) elements are the panning functions and gL(?S) and gR(?S) are the values at the S different sampling points corresponding respectively to values ?1, ?2 . . . ?S of said azimuth angle value ?,
determining by said at least one processor the order N of said Ambisonics audio signal a(t);
calculating by said at least one processor from said number S and from said order N a mode matrix ? and the corresponding
pseudo-inverse ?+ of said mode matrix ?, wherein

?=[y*(?1), y*(?2), . . . , y*(?S)] and y*(?)=[Y?N*(?), . . . , Y0*(?), . . . , YN*(?)]T is the complex conjugation of the circular harmonics vector

y(?)=[Y?N(?), . . . , Y0(?), . . . , YN(?)]T of said Ambisonics audio signal a(t) and Ym(?) are the circular harmonic functions, with m being an integer comprises between ?N and N;

calculating by said from at least one processor from said matrices G and ?+ a decoding matrix D=G ?+;

calculating by said at least one processor the loudspeaker signals l(t)=Da(t), wherein a 3D-to-2D conversion of a(t) is carried
out for this calculating,

outputting said loudspeaker signals l(t).

1. A signal processing device for filtering and performing high frequency reconstruction of an audio signal, the signal processing
device comprising:
an analysis filter bank that receives real valued time domain input audio samples and generates complex-valued subband samples;
a phase shifter that shifts a phase of the complex-valued subband samples by an amount;
a high frequency reconstructor that generates modified complex-valued subband samples through a high frequency reconstruction
process;

a phase shifter that unshifts a phase of the modified complex-valued subband samples by the amount; and
a synthesis filter bank that receives the modified complex-valued subband samples and generates time domain output audio samples,
wherein the analysis filter bank comprises analysis filters (hk(n)) and the synthesis filter bank comprises synthesis filters (fk(n)) that are complex exponential modulated versions of a prototype filter (p0(n)) according to:


where M is a number of channels, the prototype filter (p0(n)) has a length N, and the analysis filter bank and synthesis filter bank have a system delay of D samples,

wherein one or more of the analysis filter bank, the high frequency reconstructor, and the synthesis filter bank is implemented,
at least in part, by one or more hardware elements of the signal processing device, and

wherein the complex valued subband samples are oversampled by a factor of two.

1. A decoding method in an audio processing system comprising a decoder, the method comprising:
receiving, by the decoder, a first waveform-coded signal having a spectral content up to a first cross-over frequency,
receiving, by the decoder, a second waveform-coded signal having a spectral content corresponding to a subset of the frequency
range above the first cross-over frequency, wherein the spectral content of the second waveform-coded signal includes a frequency
interval extending down to the first cross-over frequency and having a time-variable upper bound,

receiving, by the decoder, high frequency reconstruction parameters,
performing, by the decoder, high frequency reconstruction using the first waveform-coded signal and the high frequency reconstruction
parameters so as to generate a frequency extended signal having a spectral content above the first cross-over frequency, and

generating an interleaved signal by interleaving, by the decoder, the frequency extended signal with the second waveform-coded
signal.

1. A decoding method in a multi-channel audio processing system for reconstructing M encoded channels, wherein M>2, comprising
the steps of:
receiving N waveform-coded downmix signals comprising spectral coefficients corresponding to frequencies between a first and
a second cross-over frequency, wherein 1
receiving M waveform-coded signals comprising spectral coefficients corresponding to frequencies up to the first cross-over
frequency, each of the M waveform-coded signals corresponding to a respective one of the M encoded channels;

downmixing the M waveform-coded signals into N downmix signals comprising spectral coefficients corresponding to frequencies
up to the first cross-over frequency;

combining each of the N waveform-coded downmix signals comprising spectral coefficients corresponding to frequencies between
a first and a second cross-over frequency with a corresponding one of the N downmix signals comprising spectral coefficients
corresponding to frequencies up to the first cross-over frequency into N combined downmix signals;

extending each of the N combined downmix signals to a frequency range above the second cross-over frequency by performing
high frequency reconstruction, whereby each extended downmix signal comprises spectral coefficients corresponding to a range
extending below the first cross-over frequency and above the second cross-over frequency;

performing a parametric upmix of the N frequency extended combined downmix signals into M upmix signals comprising spectral
coefficients corresponding to frequencies above the first cross-over frequency, each of the M upmix signals corresponding
to one of the M encoded channels; and

combining the M upmix signals comprising spectral coefficients corresponding to frequencies above the first cross-over frequency
with the M waveform-coded signals comprising spectral coefficients corresponding to frequencies up to the first cross-over
frequency.

1. An audio processing system configured to accept an audio bitstream, the audio processing system comprising:
a decoder adapted to receive the bitstream and to output quantized spectral coefficients;
a front-end component, which includes:
a dequantization stage adapted to receive the quantized spectral coefficients and to output a first frequency-domain representation
of an intermediate signal; and

an inverse transform stage for receiving the first frequency-domain representation of the intermediate signal and synthesizing,
based thereon, a time-domain representation of the intermediate signal;

a processing stage, which includes:
an analysis filterbank for receiving the time-domain representation of the intermediate signal and outputting a second frequency-domain
representation of the intermediate signal;

at least one processing component for receiving said second frequency-domain representation of the intermediate signal and
outputting a frequency-domain representation of a processed audio signal; and

a synthesis filterbank for receiving the frequency-domain representation of the processed audio signal and outputting a time-domain
representation of the processed audio signal; and

a sample rate converter for receiving said time-domain representation of the processed audio signal and outputting a reconstructed
audio signal sampled at a target sampling frequency,

wherein the respective internal sampling rates of the time-domain representation of the intermediate audio signal and of the
time-domain representation of the processed audio signal are equal, and wherein said at least one processing component includes:

a parametric upmix stage for receiving a downmix signal with M channels and outputting, based thereon, a signal with N channels,
wherein the parametric upmix stage is operable at least in a mode where 1?M and

a first delay stage configured to incur a delay, when the parametric upmix stage is in the mode where 1?M=N, to compensate
for the delay associated with the mode where 1?M of a current operating mode of the parametric upmix stage.

1. A non-transitory computer-readable medium for storing data representing a sign-data-hiding enabled block of an image, comprising:a bitstream written in the non-transitory computer-readable medium, the bitstream comprising:
a set of context-based adaptive binary arithmetic coding (CABAC) encoded coefficients representing a set of coefficients of a residual block of the sign-data-hiding enabled block, the set of coefficients including a particular non-zero coefficient that is without a sign designation; and
an information item representing a prediction mode of the sign-data-hiding enabled block,
wherein remainder data, which is based on an operation representing a division between a sum of non-zero coefficients in the set of coefficients and a specific number, is used to designate a sign for the particular non-zero coefficient, and
wherein the residual block of the sign-data-hiding enabled block corresponds to a difference between an original block and a prediction block generated by using the prediction mode.

1. A decoder configured to decode an encoded bitstream, the decoder comprising:
a demultiplexer for demultiplexing the encoded bitstream for obtaining a lowband core decoder signal, level parameters, and
balance parameters;

a lowband core decoder for producing a lowband output signal, the lowband output signal having a lowband mono signal or a
lowband stereo signal;

a high-frequency reconstruction device for generating a synthetic highband using the lowband output signal, the level parameters,
and the balance parameters and for combining the synthetic highband and the lowband output signal to form a combined signal,
and

an output interface for outputting the combined signal,
wherein the level parameters represent a total power in a frequency band of a signal having two channels,
wherein the total power represents a sum of an energy of a left channel and an energy of a right channel for a given time
segment and frequency band,

wherein the balance parameters represent a quotient of an energy of the left channel and an energy of the right channel,
wherein the balance parameters are delta coded in frequency.

1. Apparatus for performing gain adjustment on a plurality of audio sub band signals generated by filtering an audio signal
using a filter bank, the filter bank having sub band filters, adjacent sub band filters of the filterbank having transition
bands overlapping in an overlapping range, comprising:
an analyzer for analysing the plurality of audio sub band signals generated by filtering the signal using the filter bank
to determine, whether a subband signal of a sub band filter and a sub band signal of an adjacent sub band filter have aliasing
generating signal components in the overlapping range between the sub band filter and the adjacent sub band filter to obtain
grouped audio sub band signals;

a calculator configured for calculating a first gain adjustment value and a second gain adjustment value for the grouped adjacent
audio sub band signals comprising an audio sub band signal and an adjacent audio sub band signal, wherein the calculator is
operative

to determine a first energy measure indicating a signal energy of the audio sub band signal and a second energy measure indicating
a signal energy of the adjacent audio sub band signal,

to determine an indication of a reference energy for the grouped adjacent audio sub band signals as a linear combination of
a first reference energy value for the audio sub band signal and a second reference energy value for the adjacent audio sub
band signal, and

to determine an energy estimate for an energy in the grouped adjacent audio sub band signals as a linear combination of the
first energy measure for the audio sub band signal and the second energy measure for the adjacent audio sub band signal, and

to calculate the first gain adjustment value and the second gain adjustment value for the grouped adjacent audio sub band
signals based on the linear combination of the first reference energy value for the audio sub band signal and the second reference
energy value for the adjacent audio sub band signal and based on the linear combination of the first energy measure for the
audio sub band signal and the second energy measure for the adjacent audio sub band signal; and

a gain adjuster configured for applying the first gain adjustment value to the audio sub band signal of the grouped adjacent
audio sub band signals and for applying the second gain adjustment value to the adjacent audio sub band signal of the grouped
adjacent audio sub band signals,

wherein one or more of the analyzer, the calculator, and the gain adjuster is implemented, at least in part, by one or more
hardware elements of the apparatus.

1. An apparatus for producing a full bandwidth audio signal having a low band portion and a high band portion, the apparatus
comprising:
an audio decoder that decodes an encoded audio signal to produce a time domain decoded audio signal, the time domain decoded
audio signal including only the lowband portion;

a cosine modulated, real-valued analysis filterbank that receives the time domain decoded audio signal and produces a plurality
of real-valued subband signals;

a high frequency reconstructor that regenerates at least some of the highband portion by copying one or more of the plurality
of real-valued subband signals up to the highband portion;

an aliasing detector that identifies subband signals where aliasing created by spectral envelope adjustment of an audio signal
may occur based at least in part on a linear predictor applied to at least some of the plurality of real-valued subband signals;

an energy estimator that estimates an energy of at least some of the plurality of copied real-valued subband signals;
an aliasing reducer that modifies a gain to be applied to at least some of the identified subbands signals based at least
in part on the estimated energy; and

a real-valued synthesis filterbank that combines the plurality of real-valued subband signals with the highband portion to
produce the full bandwidth audio signal, the full bandwidth audio including real-valued time domain samples,

wherein the apparatus is implemented at least in part with one of more hardware elements.

1. A method, comprising:
receiving geo origination data and non-locational sensor data from a geo-tagged media device of a user;
selecting, based at least in part on the geo origination data, one or more geo-tagged media content elements from a plurality
of geo-tagged media content elements;

generating, based at least in part on the one or more selected geo-tagged media content elements and the geo origination data,
geo-referenced rendering data for the one or more selected geo-tagged media content elements, the geo-referenced rendering
data to be used for rendering media content from the one or more selected geo-tagged media content elements perceivable to
the user of the geo-tagged media device;

selecting, based at least in part on the non-locational sensor data, one or more second geo-tagged media content elements
from the plurality of geo-tagged media content elements;

generating, based at least in part on the one or more second selected geo-tagged media content elements and the non-locational
sensor data, second geo-referenced rendering data for the one or more second selected geo-tagged media content elements, the
second geo-referenced rendering data to be used for rendering second media content from the one or more second selected geo-tagged
media content elements perceivable to the user of the geo-tagged media device;

wherein the method is performed by one or more computing devices.