Abstract: A method is described which decodes a downmix matrix for mapping a plurality of input channels of audio content to a plurality of output channels, the input and output channels being associated with respective speakers at predetermined positions relative to a listener position, wherein the downmix matrix is encoded by exploiting the symmetry of speaker pairs of the plurality of input channels and the symmetry of speaker pairs of the plurality of output channels. Encoded information representing the encoded downmix matrix is received and decoded for obtaining the decoded downmix matrix.

Abstract: The present invention relates to an improved artificial neural network for predicting one or more next items in a sequence of items based on an input sequence item. The improved artificial neural network has greatly reduced memory requirements, making it suitable for use on electronic devices such as mobile phones and tablets. The invention includes an electronic device on which the improved artificial neural network operates, and methods of predicting the one or more next items in the sequence using the improved artificial neural network.

Abstract: A method of transmitting rendering data of a computer image to a client terminal via a communication channel includes: receiving rendering results of the computer image from at least one computer of a plurality of computers; identifying a partial region of the computer image based on transmission information; processing a subset of the rendering results, the subset corresponding to the identified partial region of the computer image; and transmitting, at a particular time, the processed subset of the rendering results to the client terminal via the communication channel.

Abstract: An apparatus for generating a plurality of audio channels for a speaker setup, comprises a processor repeating an energy distribution from a speaker not contained in the speaker setup to the speakers in the speaker setup to acquire a downmix information for a downmix to the speaker setup; and a renderer for generating the plurality of audio channels using the downmix information.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: An audio metadata providing apparatus and method and a multichannel audio data playback apparatus and method to support a dynamic format conversion are provided. Dynamic format conversion information may include information about a plurality of format conversion schemes that are used to convert a first format set by a writer of multichannel audio data into a second format that is based on a playback environment of the multichannel audio data and that are set for each of playback periods of the multichannel audio data. The audio metadata providing apparatus may provide audio metadata including the dynamic format conversion information. The multichannel audio data playback apparatus may identify the dynamic format conversion information from the audio metadata, may convert the first format of the multichannel audio data into the second format based on the identified dynamic format conversion information, and may play back the multichannel audio data with the second format.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

Abstract: An apparatus, system, and method, each of which: acquires sound data generated from a plurality of sound signals collected at a plurality of microphones; acquires, from one or more sensors, a result of detecting a position of the sound recording apparatus at a time point during a time period when the plurality of sound signals is collected; and stores, in a memory, position data indicating the position of the sound recording apparatus detected at the time point, and sound data generated based on a plurality of sound signals collected at the microphones at the time point at which the position was detected, in association with each other.

Abstract: A method is provided for acoustic source direction of arrival estimation and acoustic source separation, via spatial weighting of the dictionary based display of the steered response function calculated for a certain number of directions from spherical harmonic decomposition coefficients obtained from microphone array recordings of the sound field. The usage of spatial band limited functions of plane waves to represent more complex directional maps of the sound field constitutes the algorithm. These functions are calculated for pre-defined directions on an analysis surface (such as a sphere). The directions of arrival of sound sources are calculated with the same method in order to group source estimates to localize sound sources. Thereby, directions of arrival can be obtained from the recordings of the sound sources captured by means of a microphone array and following this, sound sources can be separated by using this direction information or predetermined source arrival directions.

Abstract: A method for adjusting sound (e.g., aural) output of a worn audio device includes obtaining orientation or position data that indicates an orientation or a position of the worn audio device. The method also includes determining at least one change in the orientation or position data indicating a change in orientation or position of the worn audio device. The method also includes adjusting or selecting a filter for adjusting aural output produced by the worn audio device using the orientation or position data, and operating the worn audio device to provide the sound output using the filter in response to the at least one change in the orientation or position data.

Abstract: An apparatus for generating a plurality of audio channels for a first speaker setup is characterized by an imaginary speaker determiner, an energy distribution calculator, a processor and a renderer. The imaginary speaker determiner is configured to determine a position of an imaginary speaker not contained in the first speaker setup to obtain a second speaker setup containing the imaginary speaker. The energy distribution calculator is configured to calculate an energy distribution from the imaginary speaker to the other speakers in the second speaker setup. The processor is configured to repeat the energy distribution to obtain a downmix information for a downmix from the second speaker setup to the first speaker setup. The renderer is configured to generate the plurality of audio channels using the downmix information.

Abstract: The directivity of a loudspeaker describes how sound produced by the speaker varies with angle and frequency. Low-frequency sound tends to be relatively omnidirectional, while high-frequency sound tends to be more strongly directional. Because the two ears of a listener are in different spatial positions, the direction-dependent performance of the speakers can produce unwanted differences in volume or spectral content between the two ears. For example, high-frequency sounds may appear to be muffled in one ear, compared to the other. A multi-speaker sound system can employ binaural directivity compensation, which can compensate for directional variations in performance of each speaker, and can reduce or eliminate the difference in volume or spectral content between the left and right ears of a listener. The binaural directivity compensation can optionally be included with spatial audio processing, such as crosstalk cancellation, or can optionally be included with loudspeaker equalization.

Abstract: The present disclosure relates to a method of converting an audio signal in an intermediate signal format to a set of speaker feeds suitable for playback by an array of speakers. The audio signal in the intermediate signal format is obtainable from an input audio signal by means of a spatial panning function. The method comprises determining a discrete panning function for the array of speakers, determining a target panning function based on the discrete panning function, wherein determining the target panning function involves smoothing the discrete panning function, and determining a rendering operation for converting the audio signal in the intermediate signal format to the set of speaker feeds, based on the target panning function and the spatial panning function. The present disclosure further relates to a corresponding apparatus and a corresponding computer-readable storage medium.

Abstract: A multi-channel decorrelator for providing a plurality of decorrelated signals on the basis of a plurality of decorrelator input signals is configured to premix a first set of N decorrelator input signals into a second set of K decorrelator input signals, wherein K<N. The multi-channel decorrelator is configured to provide a first set of K? decorrelator output signals on the basis of the second set of K decorrelator input signals. The multi-channel decorrelator is further configured to upmix the first set of K? decorrelator output signals into a second set of N? decorrelator output signals, wherein N?>K?. The multi-channel decorrelator can be used in a multi-channel audio decoder. A multi-channel audio encoder provides complexity control information for the multi-channel decorrelator.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

Abstract: An apparatus for generating a plurality of audio channels for a first speaker setup is characterized by an imaginary speaker determiner, an energy distribution calculator, a processor and a renderer. The imaginary speaker determiner is configured to determine a position of an imaginary speaker not contained in the first speaker setup to obtain a second speaker setup containing the imaginary speaker. The energy distribution calculator is configured to calculate an energy distribution from the imaginary speaker to the other speakers in the second speaker setup. The processor is configured to repeat the energy distribution to obtain a downmix information for a downmix from the second speaker setup to the first speaker setup. The renderer is configured to generate the plurality of audio channels using the downmix information.

Abstract: A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output.

Abstract: This disclosure falls into the field of object-based audio content, and more specifically it is related to the field of conversion of multi channel audio content into object-based audio content. This disclosure further relates to method for processing a time frame of audio content having a spatial position.

Abstract: 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 in dependence on a rendering information has an object parameter determinator. The object parameter determinator is configured to obtain inter-object-correlation values for a plurality of pairs of audio objects. 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 obtain inter-object-correlation values for a plurality of pairs of related audio objects, or to obtain inter-object-correlation values for a plurality of pairs of related audio objects using a common inter-object-correlation bitstream parameter value.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: An apparatus for encoding a multi-channel signal having at least three channels includes an iteration processor, a channel encoder and an output interface. The iteration processor is configured to calculate inter-channel correlation values between each pair of the at least three channels, for selecting a pair including a highest value or including a value above a threshold, and for processing the selected pair using a multi-channel processing operation to derive first multi-channel parameters for the selected pair and to derive first processed channels. The iteration processor is configured to perform the calculating, the selecting and the processing using at least one of the processed channels to derive second multi-channel parameters and second processed channels. The channel encoder is configured to encode channels resulting from an iteration processing to obtain encoded channels.

Abstract: The present research relates to controlling rendering of multi-object or multi-channel audio signals. The present research provides a method and apparatus for controlling rendering of multi-object or multi-channel audio signals based on spatial cues in a process of decoding the multi-object or multi-channel audio signals. To achieve the purpose, the method suggested in the research controls rendering in a spatial cue domain in the process of decoding the multi-object or multi-channel audio signals.

Abstract: Disclosed is an audio signal processing device. The audio signal processing device includes a receiving unit configured to receive a first audio signal corresponding to a sound collected by a first sound collecting device and a second audio signal corresponding to a sound collected by a second sound collecting device, a processor configured to process the second audio signal based on a correlation between the first audio signal and the second audio signal, and an output unit configured to output a processed second audio signal. The first audio signal is a signal for reproducing an output sound of a specific sound object, and the second audio signal is a signal for ambience reproduction of a space in which the specific sound object is positioned.

Abstract: A decoder for generating an audio output signal having one or more audio output channels from a downmix signal having one or more downmix channels is provided. The downmix signal encodes one or more audio object signals. The decoder has a threshold determiner for determining a threshold value depending on a signal energy and/or a noise energy of at least one of the of or more audio object signals and/or depending on a signal energy and/or a noise energy of at least one of the one or more downmix channels. Moreover, the decoder has a processing unit for generating the one or more audio output channels from the one or more downmix channels depending on the threshold value, by computing eigenvalues of a downmix channel cross correlation matrix, wherein each eigenvalue except largest eigenvalue is compared to the threshold value, and omitted if they are smaller.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: Method for processing a stereo signal includes encoding an N-channel audio signal in a stereo signal (Lo, Ro) and spatial parameters (wl, wr), processing the stereo signal using the spatial parameters for generating a processed stereo signal (low, Row). The matrix of the processed stereo signal is described as the matrix of the stereo signal, multiplied by a filter matrix (H) having element that are filter functions (H1, H2, H3, H4) operated with spatial parameters (wl, wr) and a constant (a). The filter functions are time invariant and selected so that the matrix is invertible.

Abstract: Disclosed is an audio signal processing method. The audio signal processing method according to the present invention comprises the steps of: receiving a bit-stream including at least one of a channel signal and an object signal; receiving a user's environment information; decoding at least one of the channel signal and the object signal on the basis of the received bit-stream; generating the user's reproducing channel information on the basis of the user's received environment information; and generating a reproducing signal through a flexible renderer on the basis of at least one of the channel signal and the object signal and the user's reproducing channel information.

Abstract: Sound scenes in 3D can be synthesized or captured as a natural sound field. For decoding, a decode matrix is required that is specific for a given loudspeaker setup and is generated using the known loudspeaker positions. However, some source directions are attenuated for 2D loudspeaker setups like e.g. 5.1 surround. An improved method for decoding an encoded audio signal in soundfield format for L loudspeakers at known positions comprises steps of adding (10) a position of at least one virtual loudspeaker to the positions of the L loudspeakers, generating (11) a 3D decode matrix (D?), wherein the positions (Formula I) of the L loudspeakers and the at least one virtual position (Formula II) are used, downmixing (12) the 3D decode matrix (D?), and decoding (14) the encoded audio signal (i14) using the downscaled 3D decode matrix (Formula III). As a result, a plurality of decoded loudspeaker signals (q14) is obtained.

Abstract: Disclosed is a method and a system to determine an audio precompensation controller for an associated sound generating system including a total of N?2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L?1 inputs for L input signals and N outputs for N controller output signals, one to each loudspeaker. For each one of at least a subset of the N loudspeaker inputs, an impulse response is estimated at each measurement position. For each one of the L input signal(s), a selected one of the N loudspeakers is specified as a primary loudspeaker and a selected subset S including at least one of the N loudspeakers as support loudspeaker(s).

Abstract: The present technology relates to an encoding device and method, a decoding device and method, and a program which can obtain high quality audio with a less code amount. A signal encoding unit encodes audio signals and outputs a resultant signal code string. A coefficient encoding unit encodes mixing coefficients for use in a mixing process of the audio signals and outputs a resultant coefficient code string. A multiplexing unit multiplexes the signal code string and the coefficient code string and outputs a resultant output code string. The coefficient encoding unit rearranges the mixing coefficients at the time of encoding the mixing coefficients on the basis of distances between input-side sound source positions and speaker positions on a reproduction side and calculates differential values between the mixing coefficients on the basis of arrangement order of the mixing coefficients, thereby encoding the mixing coefficients.

Abstract: An audio system includes a level controller receiving an input audio signal from an audio medium. The level controller includes a gain adjust module producing an output audio signal based on the input audio signal. A filter arrangement is communicatively coupled to the level controller. The filter arrangement receives the output audio signal and divides the output signal into a plurality of filtered signals. Each of the filtered signals corresponds to a respective frequency band. A multiband dynamics controller is communicatively coupled to the filter arrangement. The multiband dynamics controller receives the filtered signals and applies a respective gain characteristic to each of the filtered signals to thereby produce a plurality of gain adjust signals. The gain adjust signals are combined into a multiband dynamics controller output signal.

Abstract: M audio input channels, each associated with a spatial direction, are translated to N audio output channels, each associated with a spatial direction, wherein M and N are positive whole integers, M is three or more, and N is three or more, by deriving the N audio output channels from the M audio input channels, wherein one or more of the M audio input channels is associated with a spatial direction other than a spatial direction with which any of the N audio output channels is associated, and at least one of the one or more of the M audio input channels is mapped to a respective set of at least three of the N output channels. At least three output channels of a set may be associated with contiguous spatial directions.

Abstract: A method and apparatus for performing an adaptive down-mixing of a multichannel audio signal comprising a number of input channels, wherein a signal adaptive transformation of said input channels is performed by multiplying the input channels with a downmix block matrix comprising a fixed block for providing a set of backward compatible primary channels and a signal adaptive block for providing a set of secondary channels.

Abstract: Gapless looping playback of an audio sound seamlessly without any silence between loops. An uncompressed digital audio file is divided and encoded using an MP3 encoder to produce two encoded files, A and B. The MP3 encoder adds a padding delay at the beginning and end of each file. Two timers are defined: T1=A?(a+b) and T2=B?(d+c), where a is the padding delay at the end of A, b is the delay at the beginning of B, c is the delay at the beginning of A, and d is the delay at the end of B. Simultaneously, a decoder starts to decode A on Track 1 and T1 is started. Immediately after T1 expires, a decoder starts to decode B on Track 2 and T2 is started. Thanks to the timers, the respective end and beginning of the sound portions of A and B temporally align so that no silence is heard.

Abstract: An encoder is disclosed for use in a surround sound system wherein at least four audio input signals (FL, FR, RL, RR) representing an original sound field are encoded into two channel signals (L, R) and the encoded two channel signals are decoded into at least four audio output signals (FL?, FR?, RL?, RR?) corresponding to the four audio input signals. The encoder includes matrix means connected to receive the four audio input signals for encoding the four input signals into two channel (L and R) output signals. The matrix means includes means responsive to the four input signals for producing Lenc and Renc output signals as follows: Lenc=FL+kFR+jRL+jkRR Renc=FR+kFL?jRR?jkRL wherein k denotes a transformation or matrix coefficient having a value that is steered dynamically based on level of rear signal (RL+RR) content relative to front signal (FL+FR) content. An encoding method for use in a surround sound system is also disclosed.

Abstract: Provided are a method and apparatus for encoding/decoding stereo audio. In the method for encoding stereo audio, stereo audio is encoded based on at least one of the phase difference between first and second channel audios and information on an angle made by a vector on the intensity of mono-audio and a vector on the intensity of the first channel audio or a vector on the intensity of the second channel audio. Thus, the number of encoded parameters is minimized so that a compression ratio in the encoding of the stereo audio is improved.

Abstract: An system, method, and method of encoding/decoding a multi-channel audio signal, including a decoding level generation unit producing decoding-level information that helps a bitstream including a number of audio channel signals and space information to be decoded into a number of audio channel signals, wherein the space information includes information about magnitude differences and/or similarities between channels, and an audio decoder decoding the bitstream according to the decoding-level information. Accordingly, even a single input bitstream can be decoded into a suitable number of channels depending on the type of a speaker configuration used. Scalable channel decoding can be achieved by partially decoding an input bitstream. In the scalable channel decoding, a decoder may set decoding levels and outputs audio channel signals according to the decoding levels, thereby reducing decoding complexity.

Abstract: A method for producing an optimum sound field of loudspeakers is revealed. Firstly enclose a surface of a human head with a first closed geometric shape. Then enclose the first closed geometric shape with at least one second closed geometric shape. Next set up reference points on the first and second closed geometric shapes respectively. Use signal strength of the reference point on the second closed geometric shape and a gradient of the corresponding reference points on the first and second closed geometric shapes to obtain a transfer function of each point between a virtual loudspeaker/a real loudspeaker and the sound field respectively. Then create a virtual sound field and a real sound field according to the above transfer functions. Finally, get an optimum signal of the real loudspeaker by minimizing an error between the virtual sound field and the real sound field with a boundary condition.

Abstract: A decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: A method and an apparatus for adjusting a channel delay parameter of a multi-channel signal are provided in the embodiments of the present invention. The method includes: performing down-mixing processing on a multi-channel signal to obtain a processed signal (101); calculating energy distribution of the processed signal (102); and judging whether a comb filtering effect occurs in the processed signal according to the energy distribution of the processed signal, and adjusting a channel delay parameter of the multi-channel signal if the comb filtering effect occurs in the processed signal (103).

Abstract: A method for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources is provided. This method comprises decomposing the multi-channel signal into frequency bands and, per frequency band, obtaining directivity information per sound source of the sound scene, the information being representative of the spatial distribution of the sound source in the sound scene, of selecting a set of sound sources of the sound scene constituting principal sources, of matrixing the selected principal sources to obtain a sum signal with a reduced number of channels and, of coding the directivity information and of forming a binary stream comprising the coded directivity information, the binary stream being transmittable in parallel with the sum signal. A decoding method is also provided that is able to decode the sum signal and the directivity information to obtain a multi-channel signal, to an adapted coder and an adapted decoder.

Abstract: Provided is a signal separation system including a rendering unit which receives a first and a second input signal and positions the first input signal according to rendering information.

Abstract: An improved decorrelator is disclosed that processes an input audio signal in two separate paths. In one path, a banded phase-flip filter is applied to lower frequencies of the input audio signal. In a second path, a frequency-dependent delay is applied to higher frequencies of the input audio signal. Signals from the two paths are combined to obtain an output signal that is psychoacoustically decorrelated with the input audio signal. The decorrelated signal can be mixed with the input audio signal without generating audible artifacts.

Abstract: A method and apparatus to evaluate a quality of an audio signal, in which the number of effective channels is determined for each of a reference signal of a current frame and a test signal indicative of the reference signal that has passed through an audio codec, and an audio quality evaluation score of the current frame is calculated by evaluating an audio quality of the current frame based on the determined number of effective channels for each of the reference signal and the test signal by means of a predetermined evaluator.

Abstract: Disclosed is a sound processing apparatus including a factorization unit and an extraction unit. The factorization unit is configured to factorize frequency information obtained by performing time-frequency transformation on sound signals of a plurality of channels into a channel matrix expressing properties in a channel direction, a frequency matrix expressing properties in a frequency direction, and a time matrix expressing properties in a time direction. The extraction unit is configured to compare the channel matrix with a threshold and extract components specified by a result of the comparison from the channel matrix, the frequency matrix, and the time matrix to generate the frequency information on a sound from a desired sound source.

Abstract: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.

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: An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.