Abstract: An audio encoder for encoding an audio signal has: a first encoding processor for encoding a first audio signal portion in a frequency domain, having: a time frequency converter for converting the first audio signal portion into a frequency domain representation; an analyzer for analyzing the frequency domain representation to determine first spectral portions to be encoded with a first spectral resolution and second regions to be encoded with a second resolution; and a spectral encoder for encoding the first spectral portions with the first spectral resolution and encoding the second portions with the second resolution; a second encoding processor for encoding a second different audio signal portion in the time domain; a controller for analyzing and determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal havi

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer and a parameter determinator. The upmixer is configured to apply temporally variable upmix parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal, wherein the temporally variable upmix parameters include temporally variable smoothened phase values. The parameter determinator is configured to obtain one or more temporally smoothened upmix parameters for usage by the upmixer on the basis of a quantized upmix parameter input information. The parameter determinator is configured to combine a scaled version of a previous smoothened phase value with a scaled version of an input phase information using a phase change limitation algorithm, to determine a current smoothened phase value on the basis of the previous smoothened phase value and the phase input information.

Abstract: A processor for processing an audio signal has: an analyzer for deriving a window control signal from the audio signal indicating a change from a first asymmetric window to a second window, or indicating a change from a third window to a fourth asymmetric window, wherein the second window is shorter than the first window, or wherein the third window is shorter than the fourth window; a window constructor for constructing the second window using a first overlap portion of the first asymmetric window, wherein the window constructor is configured to determine a first overlap portion of the second window using a truncated first overlap portion of the first asymmetric window, or wherein the window constructor is configured to calculate a second overlap portion of the third window using a truncated second overlap portion of the fourth asymmetric window; and a windower for applying the first and second windows or the third and fourth windows to obtain windowed audio signal portions.

Abstract: An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Abstract: An apparatus for generating an enhanced signal from an input signal, wherein the enhanced signal has spectral values for an enhancement spectral region, the spectral values for the enhancement spectral regions not being contained in the input signal, includes a mapper for mapping a source spectral region of the input signal to a target region in the enhancement spectral region, the source spectral region including a noise-filling region; and a noise filler configured for generating first noise values for the noise-filling region in the source spectral region of the input signal and for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from the first noise values or for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from first noise values in the source region.

Abstract: An audio encoder for encoding an audio signal, includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal.

Abstract: An apparatus for generating an enhanced signal from an input signal, wherein the enhanced signal has spectral values for an enhancement spectral region, the spectral values for the enhancement spectral regions not being contained in the input signal, includes a mapper for mapping a source spectral region of the input signal to a target region in the enhancement spectral region, the source spectral region including a noise-filling region; and a noise filler configured for generating first noise values for the noise-filling region in the source spectral region of the input signal and for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from the first noise values or for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from first noise values in the source region.

Abstract: A coding efficiency of coding spectral coefficients of a spectrum of an audio signal is increased by en/decoding a currently to be en/decoded spectral coefficient by entropy en/decoding and, in doing so, performing the entropy en/decoding depending, in a context-adaptive manner, on a previously en/decoded spectral coefficient, while adjusting a relative spectral distance between the previously en/decoded spectral coefficient and the currently en/decoded spectral coefficient depending on an information concerning a shape of the spectrum.

Abstract: An encoder for encoding an audio signal includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder includes a converter for deriving converted prediction coefficients from the analysis prediction coefficients, a memory for storing a multitude of correction values and a calculator. The calculator includes a processor for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator includes a combiner for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients.

Abstract: For synchronizing multichannel extension data with an audio signal, wherein the audio signal includes block division information and the multichannel extension data include reference audio signal fingerprint information, the block division information in the audio signal is detected by means of a block detector. Thereupon, block division of the audio signal is performed by a fingerprint calculator according to the block division information in order to obtain a sequence of test audio signal fingerprints. In addition to that, a sequence of reference audio signal fingerprints is extracted from the reference audio signal fingerprint information of the multichannel extension data. Both sequences of fingerprints are correlated in order to obtain a correlation result, by which a compensator is controlled in order to reduce or eliminate a time offset between the multichannel extension data and the audio signal.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: A selected channel of a multi-channel signal which is represented by frames composed from sampling values having a high time resolution can be encoded with higher quality when a wave form parameter representation representing a wave form of an intermediate resolution representation of the selected channel is derived, the wave form parameter representation including a sequence of intermediate wave form parameters having a time resolution lower than the high time resolution of the sampling values and higher than a time resolution defined by a frame repetition rate. The wave form parameter representation with the intermediate resolution can be used to shape a reconstructed channel to retrieve a channel having a signal envelope close to that one of the selected original channel. The time scale on which the shaping is performed is shorter than the time scale of a framewise processing, thus enhancing the quality of the reconstructed channel.

Abstract: An audio signal decoder for providing a decoded audio signal representation on the basis of an encoded audio signal representation has a decoder preprocessing stage for obtaining a plurality of frequency band signals from the encoded audio signal representation, a clipping estimator, a level shifter, a frequency-to-time-domain converter, and a level shift compensator. The clipping estimator analyzes the encoded audio signal representation and/or side information relative to a gain of the frequency band signals in order to determine a current level shift factor. The level shifter shifts levels of the frequency band signals according to the level shift factor. The frequency-to-time-domain converter converts the level shifted frequency band signals into a time-domain representation. The level shift compensator acts on the time-domain representation for at least partly compensating a corresponding level shift and for obtaining a substantially compensated time-domain representation.

Abstract: An upmixer for upmixing a downmix audio signal into an upmixed audio signal describing one or more upmixed audio channels includes a parameter applier configured to apply upmixing parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal. The parameter applier is configured to apply a phase shift to the downmix audio signal to obtain a phase-shifted version of the downmix audio signal, while leaving a decorrelated signal unmodified by the phase shift. The parameter applier is further configured to combine the phase-shifted version of the downmix audio signal with the decorrelated signal to obtain the upmixed audio signal.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer and a parameter determinator. The upmixer is configured to apply temporally variable upmix parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal, wherein the temporally variable upmix parameters include temporally variable smoothened phase values. The parameter determinator is configured to obtain one or more temporally smoothened upmix parameters for usage by the upmixer on the basis of a quantized upmix parameter input information. The parameter determinator is configured to combine a scaled version of a previous smoothened phase value with a scaled version of an input phase information using a phase change limitation algorithm, to determine a current smoothened phase value on the basis of the previous smoothened phase value and the phase input information.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer and a parameter determinator. The upmixer is configured to apply temporally variable upmix parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal, wherein the temporally variable upmix parameters include temporally variable smoothened phase values. The parameter determinator is configured to obtain one or more temporally smoothened upmix parameters for usage by the upmixer on the basis of a quantized upmix parameter input information. The parameter determinator is configured to combine a scaled version of a previous smoothened phase value with a scaled version of an input phase information using a phase change limitation algorithm, to determine a current smoothened phase value on the basis of the previous smoothened phase value and the phase input information.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer configured to apply temporally variable upmixing parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal. The apparatus also includes a parameter interpolator, wherein the parameter interpolator is configured to obtain one or more temporally interpolated upmix parameters to be used by the upmixer on the basis of a first complex-valued upmix parameter and a subsequent second complex-valued upmix parameter.

Abstract: An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Abstract: For synchronizing multichannel extension data with an audio signal, wherein the audio signal includes block division information and the multichannel extension data include reference audio signal fingerprint information, the block division information in the audio signal is detected by means of a block detector. Thereupon, block division of the audio signal is performed by a fingerprint calculator according to the block division information in order to obtain a sequence of test audio signal fingerprints. In addition to that, a sequence of reference audio signal fingerprints is extracted from the reference audio signal fingerprint information of the multichannel extension data. Both sequences of fingerprints are correlated in order to obtain a correlation result, by which a compensator is controlled in order to reduce or eliminate a time offset between the multichannel extension data and the audio signal.

Abstract: A selected channel of a multi-channel signal which is represented by frames composed from sampling values having a high time resolution can be encoded with higher quality when a wave form parameter representation representing a wave form of an intermediate resolution representation of the selected channel is derived, the wave form parameter representation including a sequence of intermediate wave form parameters having a time resolution lower than the high time resolution of the sampling values and higher than a time resolution defined by a frame repetition rate. The wave form parameter representation with the intermediate resolution can be used to shape a reconstructed channel to retrieve a channel having a signal envelope close to that one of the selected original channel. The time scale on which the shaping is performed is shorter than the time scale of a framewise processing, thus enhancing the quality of the reconstructed channel.