Abstract: There are provided methods and an apparatus for processing audio signals. According to one aspect of the present invention there is included a method for processing audio signals having the steps of receiving at least one audio signal having at least a center channel signal, a right side channel signal, and a left side channel signal; processing the right and left side channel signals with a first virtualizer processor, thereby creating a right virtualized channel signal and a left virtualized channel signal; processing the center channel signal with a spatial extensor to produce distinct right and left outputs, thereby expanding the center channel with a pseudo-stereo effect; and summing the right and left outputs with the right and left virtualized channel signals to produce at least one modified side channel output.

Abstract: In a method and an acoustic system that generate a stereo signal for each of multiple. separate sources, a blind source separation of at least two microphone signals is conducted to acquire BSS filters. Each of the microphone signals is filtered with its own filter transfer function that is the quotient of a power density spectral portion of the respective sound source and the overall power density spectrum of the respective microphone signal, such that two stereo signals are obtained for each microphone signal. An approximation of the signals to be separated, for example for each of two hearing devices, is thereby possible.

Abstract: A spectral management system may be used in an audio system to receive and process audio signals having multiple distributed audio channels, such as a right, left, center, right side, left side, right rear and left rear channels. The spectral management system may separate and route a frequency range of audio content included in one or more of the distributed audio channels to other distributed audio channels. The separated and routed frequency range of audio content may be combined with audio content present on the other distributed audio channels to which the separated frequency range of audio content is routed. Separation, routing and combination may include bass audio content routing, mid-bass audio content routing, subwoofer audio content routing and treble audio content routing.

Abstract: A mobile communication terminal is provided comprising a receiver for receiving broadcast content over a first broadcast channel; and a controller for controlling reproduction of the broadcast content on a display of the mobile terminal. In response to detecting a first event, the controller receives a first request to delay reproduction of broadcast content in real-time. In response to receiving the first request the controller stores, in a storage medium, first broadcast content received between a time T1 and T2. In response to detecting a second event, the controller reproduces the first broadcast content. The controller stores second broadcast content received after a time T3. After the first broadcast content is reproduced, the controller reproduces the second broadcast content.

Abstract: A compact audio reproduction system for two input signals includes at least four loudspeakers disposed at the vertices of a quadrilateral not more than two feet on any side and such that no two loudspeakers are located at a distance from one another which is less than one-fourth the greatest distance between any two loudspeakers. The two input signals are connected to alternate speakers such that no two loudspeakers at adjacent vertices of the quadrilateral produce the same signal such that a listener at an arbitrary location perceives a sound source larger than the quadrilateral and significant stereo image. The signals received by two loudspeakers located at adjacent vertices may receive signals which are equalized separately from the signals received by the other loudspeakers for the purpose of reducing comb filtering and improving the tolerance of the device to placement near walls and other obstructions.

Abstract: There is described a method of encoding input signals (CHI to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data comprising down-mix output signals (610, 620) together with complementary parametric data (600). The method includes a first step of down-mixing input signals (CHI to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CHI to CH3; 400 to 450) during down-mixing to generate said parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CHI to CH3; 400 to 450) involves including information in the down-mix signals (610, 620) which is useable during subsequent decoding of the down-mix output signals (610, 620) and the parametric data (600) to determine at least some parameter data and thereby enabling representations of the input signals (CHI to CH3; 400 to 450) to be subsequently regenerated.

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 method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: The invention relates to methods and units supporting a multichannel audio extension. In order to allow an efficient extension requiring a low computational complexity, it is proposed that at an encoding end, at least state information is provided as side information for a provided mono audio signal (M) generated out of a multichannel audio signal. The state information indicates for each of a plurality of frequency bands how a predetermined or equally provided gain value is to be applied in the frequency domain to the mono audio signal (M) for obtaining first and a second channel signals (L,R) of a reconstructed multichannel audio signal.

Abstract: A binaural cue coding scheme in which cue codes are derived from the transmitted audio signal. In one embodiment, an encoder downmixes C input channels to generate E transmitted channels, where C>E>1. A decoder derives cue codes from the transmitted channels and uses those cue codes to synthesize playback channels. For example, in one 5-to-2 BCC embodiment, the encoder downmixes a 5-channel surround signal to generate left and right channels of a stereo signal. The decoder derives stereo cues from the transmitted stereo signal, maps those stereo cues to surround cues, and applies the surround cues to the transmitted stereo channels to generate playback channels of a 5-channel synthesized surround signal.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A reproduction apparatus comprises an inverse characteristics converting section that converts the acoustic characteristics of an acoustic space into inverse characteristics thereof, a reproducing section that reproduces music data, an acoustic characteristics output section that outputs acoustic characteristics different from the acoustic characteristics of the acoustic space, an adjusting section that adjusts the music data reproduced by the reproduction section on the basis of the inverse acoustic characteristics of the acoustic space converted by the inverse characteristics conversion section and the acoustic characteristics output from the acoustic characteristics output section and an output section that outputs the music data adjusted by the adjustment section to the acoustic space.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: For synthesizing at least three output channels using two stereo input channels, the stereo input channels are analyzed to detect signal components occurring in both input channels. A signal generator is operative to introduce at least a part of the detected signal components into the second channel associated with a second speaker in an intended speaker scheme, which is positioned between a first and a third speaker in the speaker scheme. When, however, feeding of the complete detected signal components would result in a clipping situation, then only a part of the detected signal components is fed into the second channel as a real center channel and the remainder is located in the first and third channels as a phantom center channel.

Abstract: A system and method for processing multiple channel audio signals to create a realistic soundscape in a space largely independent of the number of loudspeakers and audio source channels is provided. The system includes an encoding system in the recording process and a decoding system in the local listening area.

Abstract: For a level correction in a wave field synthesis system having a wave field synthesis module and an array of loudspeakers for providing sound to a presentation region, a correction value which is based on a set amplitude state in a presentation region is determined, the set amplitude state depending on a position of the virtual source or a type of the virtual source, and the actual amplitude state in the presentation region depending on the component signals for the loudspeakers due to the virtual source. The correction value determined is fed to a manipulator manipulating the audio signal associated to the virtual source before feeding to the wave field synthesis module, or the component signals for the individual loudspeakers due to the virtual source are manipulated to reduce a deviation between a set amplitude state and an actual amplitude state at one point or several in the presentation region.

Abstract: Head Related Transfer Functions (HRTFs) of an individual are measured in rapid fashion in an arrangement where a sound source is positioned in the individual's ear channel, while microphones are arranged in the microphone array enveloping the individual's head. The pressure waves generated by the sounds emanating from the sound source reach the microphones and are converted into corresponding electrical signals which are further processed in a processing system to extract HRTFs, which may then be used to synthesize a spatial audio scene. The acoustic field generated by the sounds from the sound source can be evaluated at any desired point inside or outside the microphone array.

Abstract: An audio reproduction system is divided into a central wave-field synthesis module and a plurality of loudspeaker modules disposed in a distributed way, wherein synthesis signals for the individual loudspeakers as well as corresponding channel information associated to the synthesis signals are calculated in the central wave-field synthesis module. The synthesis signals for a loudspeaker as well as associated channel information will then be transmitted to respective loudspeaker modules via a transmission path, wherein every loudspeaker module obtains the synthesis signals and associated channel information intended for the loudspeaker associated to the loudspeaker module. A distributed audio rendering and digital/analog converting takes place in the loudspeaker module to generate the actually analog loudspeaker signals in a distributed way in spatial proximity to every loudspeaker.

Abstract: The invention concerns the processing of audio data. The invention is characterized in that it consists in: (a) encoding signals representing a sound propagated in three-dimensional space and derived from a source located at a first distance (P) from a reference point, to obtain a representation of the sound through components expressed in a spherical harmonic base, of origin corresponding to said reference point, (b) and applying to said components compensation of a near-field effect through filtering based on a second distance (R) defining, for sound reproduction, a distance between a reproduction point (HPi), and a point (P) of auditory perception where a listener is usually located.

Abstract: To provide a surround sound reproducing system capable of obtaining high realistic sensation even in the case where an integral surround speaker cannot be installed rearward of a listening position. A surround system 100 is constructed by: a sound source output device 110 for outputting bit stream data of a predetermined format; a signal processor 120 for performing a signal process on every audio signal of each channel; and a speaker system 130 made of various speakers corresponding to various channels and including an integral surround speaker. The signal processor 120 has a signal processing unit 200 including: a switch control unit 203 for selecting one of right and left surround signals amplified from a surround speaker; a frequency correcting circuit 204 for correcting the frequency characteristic of the surround signal selected by the switch control unit 203; and an adder 205 for adding the surround signal whose frequency characteristic was corrected to a main signal.

Abstract: The invention describes a method for processing sound signals for a surround left channel (SL) and a surround right channel (SR). Thereby, a continually varying delay between the resulting signals of the surround right (SR) and surround left channels (SL) is generated. Furthermore the invention describes a delay management unit, a sound processing system, an acoustic system comprising such a sound processing system, a mixing unit for such an acoustic system and a studio system comprising such a sound processing system.

Abstract: A sound reproducing apparatus for providing an optimal virtual sound source and a sound reproducing method therefor are provided. The sound reproducing apparatus includes a virtual sound signal generation unit for generating more than one virtual sound signal corresponding to locations and the number of target virtual sound sources on the basis of more than one inputted sound signal, and a virtual sound signal downmix unit for downmixing said more than one virtual sound signal to virtual sound signal outputs corresponding to a predetermined number of output channels. The number and locations of diversely variable virtual sound sources are provided, thereby obtaining an effect of providing an optimal virtual sound source adaptively depending on each different environment.

Abstract: For controlling a wave field synthesis renderer arranged in a wave field synthesis system, a scene description, in which not an absolute position or an absolute time instant, but a time span or location span within which the audio object may vary is indicated for a source, is used. Furthermore, there is provided a monitor, which monitors a utilization situation of the wave field synthesis system. An audio object manipulator finally varies the starting point of the audio object to be considered by the wave field synthesis renderer or the actual position of the audio object within the time span and/or location span, in order to avoid capacity bottlenecks on the transmission lines or in the renderer.

Abstract: A method for processing sound signal including an L side enhanced difference signal comprising a sum of a signal being removed low frequency component from L?R signal and an L+R signal, an R side enhanced difference signal comprising a sum of a signal being removed low frequency component from R?L signal and an R+L signal, an L side output signal comprising a subtraction of signals, in which said R side enhanced difference signal being delayed to remove low and high ranges and amplified by a predetermined amplifying rate for a purpose of cross cancellation talk is subtracted from said L side enhanced difference signal and an R side output signal comprising a subtraction of signals, in which said L side enhanced difference signal being delayed to remove low and high ranges and amplified by a predetermined amplifying rate for a purpose of cross cancellation talk is subtracted from said R side enhanced difference signal.

Abstract: The input signal X of one channel is divided by a multi-stage delay processing device Z?1 and each of the outputs is superimposed by a specified coefficient by a coefficient processing device W0, W1, . . . , Wk. The results are added by an adder, thereby providing a correlation eliminating filter for extracting a signal component from the input signal X of one channel having a high correlation with the input signal Y of the other channel. There is provided a coefficient updating processing device 5 for successively changing the feature of the correlation eliminating filter according to an error signal e obtained from the output signal RES and the input signal Y from the other channel, and the input signal X of one channel. A surround signal is obtained from a difference between the output RES from the correlation eliminating filter and the input signal Y of the other channel.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: According to one embodiment, during mixing of an N-channel input signal to generate an M-channel output signal, in at least one frequency sub-band, magnitude equalization is applied to the mixed channel signals such that an amplitude sum magnitude for the N input channels (e.g., the magnitude of a sum of estimated amplitudes of the N input channels) is approximately equal to an amplitude sum magnitude for the M output channels (e.g., the magnitude of a sum of estimated amplitudes of the M output channels). In one implementation, magnitude equalization is applied to one or more sub-bands (e.g., those below 1 kHz), and power equalization is applied to one or more other sub-bands (e.g., those above 1 kHz) to reduce coloration effects in the output signal.

Abstract: A method to process audio signals, an apparatus accepting audio signals, a carrier medium that carried instructions for a processor to implement the method to process audio signals, and a carrier medium carrying filter data to implement a filter of audio signals. The method includes filtering a pair of audio input signals by a process that produces a pair of output signals corresponding to the results of: filtering each of the input signals with a HRTF filter pair, and adding the HRTF filtered signals. The HRTF filter pair is such that a listener listening to the pair of output signals through headphones experiences sounds from a pair of desired virtual speaker locations.

Abstract: A method to process audio signals, an apparatus accepting audio signals, a carrier medium that carried instructions for a processor to implement the method to process audio signals, and a carrier medium carrying filter data to implement a filter of audio signals. The method includes filtering a pair of audio input signals by a process that produces a pair of output signals corresponding to the results of: filtering each of the input signals with a HRTF filter pair, and adding the HRTF filtered signals. The HRTF filter pair is such that a listener listening to the pair of output signals through headphones experiences sounds from a pair of desired virtual speaker locations.

Abstract: A method of disassembling a pair of input signals L(t) and R(t) to form subband representations of N output channel signals o1(t), o2(t), . . . , oN(t), wherein t is time. The method includes the steps of generating a subband representation of the signal L(t) containing a plurality of subband components Lk(t) where k is an integer ranging from 1 to M; generating a subband representation of the signal R(t) containing a plurality of subband components Rk(t); and constructing the subband representation for each of the plurality of output channel signals, each of those subband representations containing a plurality of subband components oj,k(t), wherein oj,k(t) represents the kth subband of the jth output channel signal and is constructed by combining components of the input signals L(t) and R(t) according to an output construction rule: oj,k(t)=f(Lk(t),Rk(t)) for k=1, 2, . . . , M and j=1, 2, . . . , N.

Abstract: A personal listening device used in connection with the amplification of signals for enhancing the perceived quality and ambiance of sound heard by listener using stereo headphones or small speaker systems utilizes a difference signal that is cross-fed between stereo channels. The invention provides a perception of spatial presence in music sound when used with either stereo headphones or small speaker systems. The circuitry, wiring and packaging provide for multiple output jacks allowing for convenient sharing of the enhanced output signal from a personal listening device with other persons using headphones. The invention provides effects similar to stereo enhancement, surround sound, and psychoacoustic imagery in a small, inexpensive package for enhancing a user's audio experience.

Abstract: A method for generating a sound giving a sensation of depth by applying, after extraction, a transfer function quadrille onto electric sound signals on the left and right thereof. The transfer functions simulate the trajectories taken by the sound associated with the electric signal to be processed in order to reach two receivers, if this sound was emitted in air. The signals that have been processed, one by one, by one of the four transfer functions of the quadrille, are combined with each other, then the sound signal thus obtained is mixed with the original electric sound signal to be processed after a temporal reset.

Abstract: A method and device are provided for determining a reproduction position of a source of sound for audio-visual reproduction of a film scene from a plurality of individual pictures with regard to a reproduction surface having a predetermined width, and with regard to a projection source having a projection reference point. A recording position of the source of sound, a camera position during recording, and an aperture angle of the camera during recording are obtained and the determined recording position is transformed to a camera coordinate system. A reproduction position for the source of sound is calculated and used to produce sound-source positioning in a reproduction room using wave-field synthesis methods.

Abstract: Techniques of making a recording of or transmitting a sound field from either multiple monaural or directional sound signals that reproduce through multiple discrete loud speakers a sound field with spatial harmonics that substantially exactly match those of the original sound field. Monaural sound sources are positioned during mastering to use contributions of all speaker channels in order to preserve the spatial harmonics. If a particular arrangement of speakers is different than what is assumed during mastering, the speaker signals are rematrixed at the home, theater or other sound reproduction location so that the spatial harmonics of the sound field reproduced by the different speaker arrangement match those of the original sound field. An alternative includes recording or transmitting directional microphone signals, or their spatial harmonic components, and then matrixing these signals at the sound reproduction location in a manner that takes into account the specific speaker arrangement.

Abstract: A system, and an apparatus of speaker, and a method for forming a virtual audio is provided. An audio source is processed by a second audio source processing program according to parameters which are entered by the user, and filtered to be a high frequency audio signal. The audio source with orientation is outputted by the orientation speaker.

Abstract: In this signal processing apparatus, a first-audio-parameter calculating unit calculates a first audio parameter based on two audio signals. A second-audio-parameter calculating unit calculates a second audio parameter based on the two audio signals. A surround-signal generating unit generates surround components to be respectively allocated to surround signals.

Abstract: A vehicle sound system encompasses a combination of speaker configuration, speaker placement, and sound processing to improve sound quality. A pair of speakers (or rows of speakers) are placed close together and located in the front of the console or dashboard with their geometric center on or near the vehicle's central axis. A sound processor acts to “spread” the sound image produced by the two closely spaced speakers by employing a cross-cancellation technique in which the cancellation signal is derived from the difference between the left and right channels. The resulting difference signal is scaled, delayed (if necessary), and spectrally modified before being added in opposite polarities to the left and right channels. The pair of speakers may be placed on a common baffle or mounting surface or in a common housing enclosure, with sound being carried through one or more ducts and emanating out of a slot.

Abstract: A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

Abstract: An audio enhancement system and method for use receives a group of multi-channel audio signals and provides a simulated surround sound environment through playback of only two output signals. The multi-channel audio signals comprise a pair of front signals intended for playback from a forward sound stage and a pair of rear signals intended for playback from a rear sound stage. The front and rear signals are modified in pairs by separating an ambient component of each pair of signals from a direct component and processing at least some of the components with a head-related transfer function. Processing of the individual audio signal components is determined by an intended playback position of the corresponding original audio signals. The individual audio signal components are then selectively combined with the original audio signals to form two enhanced output signals for generating a surround sound experience upon playback.