Abstract: A method for processing audio signals for creating a three dimensional sound environment includes: receiving at least one input signal from at least one sound source; creating a simulated signal at least in part on the basis of the received at least one input signal, the simulated signal representing a simulation of at least one input signal reflecting from the ground or floor; and creating an output signal at least partly on the basis of the simulated signal and the at least one received input signal, the output signal including a plurality of audio channels; at least two channels of the audio channels of the output signal representing signals for sound transducers above a listener's ear level at a nominal listening position, and at least two channels of the audio channels of the output signal representing signals for sound transducers below a listeners ear level at a nominal listening position.

Abstract: A sound generator includes a diaphragm including n sub-diaphragms (n, the amount of the sub-diaphragms); n voice coils corresponding to the sub-diaphragms; a signal source for outputting signals of n channels; n high-pass filters for receiving the signals of n channels and outputting n high frequency signals; n low-pass filters for receiving the signals of n channels and outputting n low frequency signals; a first mixer for mixing the n low frequency signals and then outputting a low frequency signal; and n second mixers for mixing the low frequency signal and the n high frequency signals, and then outputting n driving signals. The n driving signals actuates the n corresponding sub-diaphragms for producing stereo sounds.

Abstract: A method of inducing a state of consciousness in a listener. The method includes providing first and second sound signals. The first sound signal is provided to one ear of the listener and the second sound signal is provided to the other ear of the listener. The second sound signal is different from the first sound signal and, when provided with the first sound signal, first and second sound signals cause the listener to perceive a first source of sound that is moving about the listener or as a tremolo effect.

Abstract: A method of optimizing stereo reception for an analog radio by applying the demodulated right sound signal (SD) and left sound signal (SG) as input to a decorrelation module having a variable decorrelation rate. The decorrelation rate of the decorrelation module is modified as a function of the reception quality coefficient “alpha” provided by the radio. The decorrelation module applies a higher decorrelation rate for a smaller reception quality coefficient “alpha” and applies a lower decorrelation rate for a larger reception quality coefficient “alpha. Also, a module for generating high-pitched sounds to recreate the high-frequency component (SHF) of the right or left sound signals which has been removed in the event of poor reception.

Abstract: Six input channels, to which are allocated 5.1-channel surround signals from a plurality of input sources, are grouped into a surround channel group, and these six input channels are connected to corresponding ones of six surround buses in a one-to-one relationship. Thus, merely grouping the input channels into a surround channel group allows the signals of the individual input channels (5.1-channel surround signals) to be taken out via a plurality of output destinations (5.1-channel speakers) corresponding to the surround buses. Once an instruction is given for changing a value of a parameter, the parameter is controlled in a ganged fashion in all of the input channels of the surround channel group except for an LFE input channel of the surround channel group.

Abstract: A sound-image control device includes a sound-image controller that processes a left sound signal and a right sound signal to localize a sound image at a predetermined position. The sound-image controller performs control for enhancing a sense of depth to a sound image originally included in an input sound signal.

Abstract: A method and apparatus of expanding a listening sweet spot. A method of expanding a listening sweet spot with respect to signals output from speakers includes: obtaining an HRTF (head related transfer function) at a position of a listener's ear; moving a first virtual ear around the position of the listener's ear; obtaining an HRTF at each position of the first virtual ear; and processing a signal to be input to the speakers using the obtained HRTFs to output to the speakers.

Abstract: An audio-signal processing device that processes an audio signal and supplies the audio signal to an audio output unit includes a characteristic-component extraction unit that extracts at least a high frequency component contained in the audio signal as a characteristic component. The audio signal and the extracted characteristic component are supplied to the audio output unit so that a sound image of the extracted characteristic component is localized closer to a listener than a sound image of the audio signal.

Abstract: The invention relates to an apparatus for improving a stereo audio signal of an FM stereo radio receiver. The apparatus comprises a parametric stereo (PS) parameter estimation stage. The parameter estimation stage is configured to determine one or more parametric stereo parameters based on the stereo audio signal in a frequency-variant or frequency-invariant manner. Preferably, these PS parameters are time- and frequency-variant. Moreover, the apparatus comprises an upmix stage. The upmix stage is configured to generate the improved stereo signal based on a first audio signal and the one or more parametric stereo parameters. The first audio signal is obtained from the stereo audio signal, e.g. by a downmix operation in a downmix stage. The PS parameter estimation stage may be part of a PS encoder. The upmix stage may be part of a PS decoder.

Abstract: Mechanisms for navigation via three dimensional audio effects are described. A current location of a device and a first point of interest are determined. The point of interest may be determined based on a web service and the current location of the device may be determined via mobile device signals. A zone that includes the point of interest may be determined. A three dimensional audio effect that simulates a sound being emitted from the zone may be generated. The three dimensional audio effect may be transmitted to speakers capable of simulating three dimensional audio effects. The transmitted three dimensional audio effect may aid in navigation from a current location to the point of interest.

Abstract: A process and system for enhancing and customizing sound includes receiving an input audio sound and enhancing the voice audio input in two or more harmonic and dynamic ranges by re-synthesizing the audio into a full range PCM wave. A tone adjusting circuit is provided which includes a first section for adjusting a low frequency tone, a second section for adjusting a mid frequency tone, a third section for adjusting a high frequency tone and mixing the audio outputs processed by the first, second and third sections to produce an output audio sound. The enhancement includes the parallel processing the input audio via a low pass filter with dynamic offset, an envelope controlled bandpass filter, a high pass filter, adding an amount of dynamic synthesized sub bass to the audio and combining the four treated audio signals in a summing mixer with the original audio. The low frequency tone has a frequency of 100 Hz and a bandwidth of 0.5.

Abstract: A signal processing device is provided, including a prediction error calculating unit that calculates an error signal between a left signal l(n) and a prediction signal of the left signal l (n) predicted from a right signal r(n), a gain adjusting unit that makes a gain adjustment and outputs an error signal, a first adder that adds the left signal l(n) and the error signal and outputs, and a second adder that adds the right signal r(n) and the error signal in opposite phase and outputs.

Abstract: Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes extracting a downmix signal and spatial information from a received audio signal, generating surround converting information using the spatial information and rendering the downmix signal to generate a pseudo-surround signal in a previously set rendering domain, using the surround converting information. The apparatus for decoding an audio signal includes a demultiplexing part extracting a downmix signal and spatial information from a received audio signal, an information converting part generating surround converting information using the spatial information and a pseudo-surround generating part rendering the downmix signal to generate a pseudo-surround signal in a previously set rendering domain, using the surround converting information.

Abstract: A circuit for operating loudspeakers includes a first, second, third and fourth loudspeaker circuit, having one input each for injecting a signal and one output each for connecting a loudspeaker input. The loudspeaker circuits are designed to amplify the injected signal and to provide the amplified signal at the outputs thereof. The loudspeaker circuits can, for example, be used for a 2.1 sound system. The three channels for a 2.1 sound system can be implemented by an amplifier circuit with four loudspeaker circuits, one loudspeaker circuit each being required for the two stereo channels left and right. A subwoofer channel can be driven differentially by two loudspeaker circuits. The stereo channels are, by contrast, only still connected to one loudspeaker circuit each, and so the stereo channels require at least one further common ground cable.

Abstract: The present invention provides an arrangement, e.g. a method, an apparatus and a computer program. The arrangement comprises obtaining information indicative of spatial positions of one or more sound sources within an audio image of a composite audio signal, obtaining information indicative of the types of the one or more sound sources, determining, for each of the one or more sound sources, a position in a display of the apparatus indicative of the spatial position of the sound source and causing an apparatus to display an item representing the sound source in the determined position in the display, wherein the item representing the sound source is determined on basis of the type of the sound source.

Abstract: The present invention provides methods and systems for digitally processing audio signals in two-channel audio systems and/or applications. In particular, the present invention includes a first filter structured to split a two-channel audio input signal into a low frequency signal and a higher frequency signal. A M/S splitter is then structured to split the higher frequency signal into a middle and a side signal. A detection module is then configured to create a detection signal from the middle signal, which is used in a compression module configured to modulate the side signal to create a gain-modulated side signal. A processing module is then structured to combine the low frequency signal, middle signal, and the gain-modulated side signal to form a final output signal.

Abstract: An audio system, and an audio apparatus and a channel mapping method thereof are provided. The audio apparatus includes a interface configured to receive a plurality of test tone signals from each of a plurality of sound output apparatuses configured to output multichannel audio signals, a channel determiner configured to analyze the plurality of test tone signals determine locations of the plurality of sound output apparatuses, and generate a determination result based on the determined locations, an switch configured to output one test tone to each of the plurality of sound output apparatuses, and perform channel mapping to map channels through which the multichannel audio signals are output according to the determination result, and an output unit configured to be randomly connected to the plurality of sound output apparatuses, and output the multichannel audio signals by adjusting the random connection according to the channel mapping.

Abstract: A system enhances spatialization in which spatial information about sound sources at an originating location is represented in an audio signal. The system applies a phase difference analysis to the signals received from an array of spaced apart input devices or microphones to derive spatial or directional information about the relative directions of one or more satellite input devices or microphones. The signals from the satellite input devices or microphones are mixed by a function of their respective directions to generate a multichannel output signal. When processed by a remote or local system, the output signal provides a representation of the relative directions of the sound sources at the originating location at a receiving location.

Abstract: In accordance with an example embodiment of the invention, a method is disclosed. Near field communication is detected between at least two devices. A location of at least one of the at least two devices is determined based on the detected near field communication. An audio channel of a multi-channel audio file is assigned based on the determined location of the at least one of the at least two devices.

Abstract: An audio enhancement technique comprises a stereo audio source. The stereo audio source is split in two identical sound signals, with one of the split signals sent to a compare block and the other to an audio block. Output from the stereo audio source is fed to a stereo bus A block to create a processed audio. A bypass block for turning the processed audio on or off is provided. The output from the bypass block is fed to the compare block. The processed audio is outputted for use by user.

Abstract: An input multi-channel representation is converted into a different output multi-channel representation of a spatial audio signal, in that an intermediate representation of the spatial audio signal is derived, the intermediate representation having direction parameters indicating a direction of origin of a portion of the spatial audio signal; and in that the output multi-channel representation of the spatial audio signal is generated using the intermediate representation of the spatial audio signal.

Abstract: Systems and methods that use ultrasonic emitters for producing multi-dimensional parametric audio are provided. The systems and methods can be configured to determine HRTF filters for the left and right ears of a listener using an optical imaging system to scan a profile of a listener. Audio content may be encoded into a left and right channel for producing a three dimensional sound effect for the listener of the audio content by: processing the sound channel into left and right input channel signals; applying the HRTF filters and acoustic crosstalk cancellations filters to the left and right channel signals to generate output left and right channel signals; and modulating the left and right output channel signal frequencies onto an ultrasonic carrier.

Abstract: A system and method for acoustically reproducing k electrical audio signals (where k=2, 3, 4, . . . ) and establishing k sound zones are provided, in each of which one of k reception sound signals occurs that is an individual pattern of the reproduced and transmitted k electrical audio signals, comprising processing the k electrical audio signals to provide k processed electrical audio signals and converting the k processed electrical audio signals into corresponding k acoustic audio signals with k loudspeakers that are arranged at positions separate from each other and within or adjacent to the k sound zones. Each of the k acoustic audio signals is transferred according to a transfer matrix from each of the k loudspeakers to each of the k sound zones, where they contribute to the corresponding reception sound signals.

Abstract: A one-cabinet stereo loudspeaker, wherein at least first and second loudspeaker units (LL, LR) are arranged in a cabinet structure (CB) so as to radiate sound with at least three lobes of the first acoustic signal (L) alternating with respect to horizontal angle with at least three lobes of the second acoustic signal (R). This provides a spacious sound experience at all locations around the loudspeaker.

Abstract: An encoding device, a decoding device, and related methods are provided that eliminate the loss of synchronization of the adaptive filters of a terminal at the encoding end and a terminal at the decoding end caused by transmission errors. Deterioration of the sound quality is suppressed when a multiple channel signal is encoded with high efficiency using an adaptive filter. In the terminal at the encoding end, a buffer stores updated filter coefficients. When packet loss detection information indicating whether there is any packet loss in the terminal at the decoding end indicates that there is packet loss, a switch outputs the past filter coefficients of the previous (NX+1) frames from the buffer to an adaptive filter. The adaptive filter uses the past filter coefficients of the previous (NX+1) frames to conduct filtering.

Abstract: When two loudspeakers play the same signal, a “phantom center” image is produced between the speakers. However, this image differs from one produced by a real center speaker. In particular, acoustical crosstalk produces a comb-filtering effect, with cancellations that may be in the frequency range needed for the intelligibility of speech. Methods for using phase decorrelation to fill in these gaps and produce a flatter magnitude response are described, reducing coloration and potentially enhancing dialogue clarity. These methods also improve headphone compatibility and reduce the tendency of the phantom image to move toward the nearest speaker.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: The invention discloses a method and an encoder for processing a digital audio stereo signal. A digital audio encoder for coding such audio signal comprises a predictive Temporal Noise Shaping (TNS) filter, a Mid-/Side (M/S) coding unit, a control unit for determining a first prediction gain related to the unmodified L/R signal processed by the TNS filter and for determining a second prediction gain related to the M/S-coded L/R signal processed by the TNS filter, wherein the control unit is adapted to disable TNS-filtering—i.e. to bypass the TNS filter—for a current signal frame, if the first and second prediction gains differ by more than a pre-determined mismatch range.

Abstract: Provided are a method and apparatus for localizing a multichannel sound signal. The method includes: obtaining a multichannel sound signal to which sense of elevation is applied by applying a first filter to an input sound signal; determining at least one frequency range of a dynamic cue according to change of a head-related transfer function (HRTF) indicating information regarding paths from a spatial location of an actual speaker to ears of an audience; and applying a second filter to at least one sound signal, corresponding to the determined at least one frequency range, of at least one channel in the multichannel sound signal to change the at least one sound signal so as to remove or to reduce the dynamic cue when the multichannel sound signal is output.

Abstract: Method and devices for providing surround audio signals are provided. Surround audio signals are received and are binaurally filtered by at least one filter unit. In some embodiments, the input surround audio signals are also processed by at least one equalizing unit. In those embodiments, the binaurally filtered signals and the equalized signals are combined to form output signals.

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: Systems and methods to deliver live sound with enhanced quality conveyed by a mixing desk to a mobile computing device user contemporaneously with external live sounds. The mobile computing device is operable to receive enhanced audio signals produced by a remote audio signal processing device through a communication work. A memory resident application is configured to playback the enhanced audio signals in phase with the external sounds using an audio rendering device. An attendee at the live event can hear the sounds from the playback through an earphone coupled to the mobile computing device as well as from the ambient environment.

Abstract: A speaker array apparatus includes a speaker array and a control section that controls audio signals corresponding to channels respectively so that the speaker array emits sound beams for the audio signal of a part of the channels in a plurality of directions and sound beams for the audio signals of the channels other than the part of the channels in a plurality of directions. One of the sound beams for the audio signal of the part of the channels is substantially identical in direction with one of the sound beams for the audio signals of the channels other than the part of the channels. Sound images for emitting sounds for the audio signals of the channels other than the part of the channels are formed and a phantom sound image for the audio signal of the part of the channels is formed at a position which is deviated from positions of the sound images.

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.

Abstract: The invention is aimed at improving the quality of the filtering by transfer functions of HRTF type of signals (L, R) compressed in a transformed domain, for binaural playing on two channels (L-BIN, R-BIN), using a combination of HRTF filters (hL,L, hL,R) including a decorrelated version (HRTF-C*, HRTF-E*) of a few of these filters. For this purpose, a decorrelation cue is given with spatialization parameters (SPAT) accompanying the compressed signals (L, R). The Decorrelation comprises applying a different phase shift to each subband of the input signal combined with addition of an overall delay. The invention makes it possible to improve the broadening in the binaural rendition of audio scenes initially in a multi-channel format.

Abstract: A parametric stereo upmix method for generating a left signal and a right signal from a mono downmix signal based on spatial parameters includes predicting a difference signal comprising a difference between the left signal and the right signal based on the mono downmix signal scaled with a prediction coefficient. The prediction coefficient is derived from the spatial parameters. The method further includes deriving the left signal and the right signal based on a sum and a difference of the mono downmix signal and said difference signal.

Abstract: A system and method for generating virtual microphone signals having a particular number and configuration for channel playback from an intermediate set of signals that were recorded in an initial format that is different from the channel playback format. In one embodiment, an initial set of intermediate are Bark-banded such that each intermediate signal may lead to a corresponding power spectral density (PSD) signal representative of the initial intermediate signal. Further, one may generate cross-correlations signals for each pair of intermediate signals. Next, from the PSDs and cross correlations, one may more efficiently calculate corresponding channel signals to be used for playback on respective channel speakers. Thus, the PSDs of each channel signal may be generated at chosen angles (as well as other design factors). Further, each channel signal may also be further modified with a corresponding cancellation signal that further enhances the resultant signal in each channel.

Abstract: An acoustic echo suppression unit according to an embodiment of the present invention includes and input interface for extracting a downmix signal from an input signal, the input signal including the downmix signal and parametric side information, wherein the downmix and the parametric side information together represent a multichannel signal, a calculator for calculating filter coefficients for an adaptive filter, wherein the calculator is adapted to determine the filter coefficients based on the downmix signal and a microphone signal or a signal derived from the microphone signal, and an adaptive filter adapted to filter the microphone signal or the signal derived from the microphone signal based on the filter coefficients to suppress an echo caused by the multichannel signal in the microphone signal.

Abstract: An audio signal processing device includes: head related transfer function convolution processing units convoluting head related transfer functions with audio signals of respective channels of plural channels, which allow the listener to listen to sound so that sound images are localized at assumed virtual sound image localization positions concerning respective channels of the plural channels of two or more channels when sound is reproduced by electro-acoustic transducer means; and 2-channel signal generation means for generating 2-channel audio signals to be supplied to the electro-acoustic transducer means from audio signals of plural channels from the head related transfer function convolution processing units, wherein, in the head related transfer function convolution processing units, at least a head related transfer function concerning direct waves from the assumed virtual image localization positions concerning a left channel and a right channel in the plural channels to both ears of the listener is n

Abstract: A method and system for multiplexing audio signals into a single channel uses frequency division multiplexing. The frequency division multiplexing method herein is based on a frequency transform algorithm using FFT shifting that does not require a carrier signal for the modulation. In one embodiment, two input microphone audio signals are frequency shifted and the resulting single audio channel is directed over a standard input connection to a computing device, for instance, a smart phone using a wired TRRS connection. The TRRS analog input of the computing devices exhibits a high-pass characteristic, and the frequency shifting method enables the low frequency audio components of input audio signals to be received and processed by a software application on the smart phone. Other embodiments are disclosed.

Abstract: Method and system for generating output signals for reproduction by two physical speakers in response to input audio signals indicative of sound from multiple source locations including at least two rear locations. Typically, the input signals are indicative of sound from three front locations and two rear locations (left and right surround sources). A virtualizer generates left and right surround outputs useful for driving front loudspeakers to emit sound that a listener perceives as emitting from rear sources. Typically, the virtualizer generates left and right surround outputs by transforming rear source inputs in accordance with a head-related transfer function. To ensure that virtual channels are well heard in the presence of other channels, the virtualizer performs dynamic range compression on rear source inputs. The dynamic range compression is preferably accomplished by amplifying rear source inputs or partially processed versions thereof in a nonlinear way relative to front source inputs.

Abstract: A method and system for producing an acoustic spatial projection by creating audio channels for producing an acoustic field by mixing, on a reflective surface, sounds associated with the audio channels is provided. In one embodiment, a method includes the step of using audio information to determining a set of audio channels. Each audio channel is associated with a sound source, such as one or more loudspeakers, and for a subset of the audio channels, the associated sound sources emit sound waves directed at a reflective surface prior to being received at a listening location. The method further includes steps of determining an acoustic response of a listening environment; steps of determining a delay to apply to one or more channels of the set of audio channels; and steps of determining a frequency compensation to apply to one or more channels of the audio channels.

Abstract: An information processing apparatus includes a forward deciding unit that makes a decision as to a user's forward according to the user's orientation information, a sound generating unit that creates sound data assigned to each of virtual sound sources placed in a plurality of directions preset in advance, a compressing unit that performs compression on the created sound data by the sound generating unit in different ways between the created sound data corresponding to the user's forward obtained by the forward deciding unit and the created sound data corresponding to a direction other than the user's forward, and a communication unit that transmits the compressed sound data by the compressing unit.

Abstract: Systems and methods for generating and performing a three-dimensional audio CAPTCHA are provided. One exemplary system can include a decoy signal database storing a plurality of decoy signals. The system also can include a three-dimensional audio simulation engine for simulating the sounding of a target signal and at least one decoy signal in an acoustic environment and outputting a stereophonic audio signal based on the simulation. One exemplary method includes providing an audio prompt to a resource requesting entity. The audio prompt can have been generated based on a three-dimensional audio simulation of the sounding of a target signal containing an authentication key and at least one decoy signal in an acoustic environment. The method can include receiving a response to the audio prompt from the resource requesting entity and comparing the response to the authentication key.

Abstract: An audio output circuit includes an on-chip left channel amplifier module, an on-chip center channel amplifier module, and an on-chip right channel amplifier module. A left channel IC pin is operably coupled to an output of the on-chip left channel amplifier module. A right channel IC pin is operably coupled to an output of the on-chip right channel amplifier module. A center channel IC pin is operably coupled to an output of the on-chip center channel amplifier module. A center channel feedback IC pin is operably coupled to an input of the on-chip center channel amplifier module to provide a feedback loop. A left jack connection is operably coupled to the left channel IC pin. A right jack connection is operably coupled to the right channel IC pin. A jack return connection coupled to the center feedback IC pin. An inductor has a first node coupled to the jack return connection and a second node coupled to the center channel IC pin.

Abstract: An apparatus comprises a processor configured to receive a first audio signal and first location data, the first location data relating to a location of a source of the first audio signal; receive a second audio signal and second location data, the second location data relating to a location of a source of the second audio signal; receive selected location data relating to a selected location; and generate a multichannel signal in dependence on the first and second audio signals, the first and second location data and the selected location data.

Abstract: An apparatus for calculating driving coefficients for loudspeakers of a loudspeaker arrangement for an audio signal associated with a virtual source is described.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

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: 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.