Abstract: An audio signal decoder for providing a decoded multi-channel audio signal representation on the basis of an encoded multi-channel audio signal representation has a time warp decoder configured to selectively use individual audio channel specific time warp contours or a joint multi-channel time warp contour for a reconstruction of a plurality of audio channels represented by the encoded multi-channel audio signal representation.

Abstract: An intermediate channel representation of a multi-channel signal can be reconstructed highly efficient and with high fidelity, when upmix parameters for upmixing a transmitted downmix signal to the intermediate channel representation are derived that allow for an upmix using the same upmixing algorithms as within the multi-channel reconstruction. This can be achieved when a parameter re-calculator is used to derive the upmix parameters that takes into account also parameters having information on channels that are not included in the intermediate channel representation.

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: According to one embodiment, generating images according to intersection points includes obtaining samples of signals from transmitter-receiver pairs. A transmitter-receiver pair is configured to transmit a signal and receive the signal reflected by an object. Intersection points are determined, where an intersection point indicates an intersection for integer multiples of a sample-time distance. A subset of samples corresponding to the intersection points is selected. Image data is generated from the selected subset of samples, where the image data is used to generate an image of the object.

Abstract: Provided is an output control apparatus connected to an audio output apparatus controlling output of audio data, including: a receiving unit configured to receive encoded data obtained by encoding multi-channel audio data; a decoding unit configured to decode the encoded data; an output control unit configured to control output of a predetermined number of channels of audio data among the multi-channel audio data obtained by the decoding; a transmitting unit configured to transmit any one of audio data different from the audio data of which the output is controlled by the output control unit among the multi-channel audio data obtained by the decoding and the encoded data to an audio output apparatus connected through an integrated cable formed by integrating at least a video line, an audio line, and a control line; and a control unit configured to control the audio output apparatus through the integrated cable.

Abstract: An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal.

Abstract: A parametric representation of a multi-channel audio signal describes the spatial properties of the audio signal well with compact side information when a coherence information, describing the coherence between a first and a second channel, is derived within a hierarchical encoding process only for channel pairs including a first channel having only information of a left side with respect to a listening position and including a second channel having only information from a right side with respect to a listening position. As within the hierarchical process the multiple audio channels of the audio signal are downmixed iteratively into monophonic channels, one can pick the relevant parameters from an encoding step involving only channel pairs carrying the information needed to describe the spatial properties of the multi-channel audio signal.

Abstract: A method of audio matrix decoding in which a moving sound image is restored includes decoding multichannel signals from stereo signals, extracting strengths and positions of virtual sound sources existing between channels based on power vectors of the decoded multichannel signals, comparing the strengths and positions of the extracted previous and current virtual sound sources to predict position movement and the strengths of the virtual sound sources, and redistributing powers to positions of channel speakers in a multichannel arrangement based on the predicted position of a sound image.

Abstract: A first signal representation of one or more of the multiple channels is encoded in a first encoding process, and a second signal representation of one or more of the multiple channels is encoded in a second, filter-based encoding process. Filter smoothing can be used to reduce the effects of coding artifacts. However, conventional filter smoothing generally leads to a rather large performance reduction and is therefore not widely used. It has been recognized that coding artifacts are perceived as more annoying than temporary reduction in stereo width, and that they are especially annoying when the coding filter provides a poor estimate of the target signal; the poorer the estimate, the more disturbing artifacts. Therefore, signal-adaptive filter smoothing is introduced in the second encoding process or a corresponding decoding process.

Abstract: Provided is an apparatus and method of encoding and decoding multiple channel signals based upon phase information and one or more residual signals.

Abstract: A parametric stereo upmix apparatus (300, 400) generating a left signal (206) and a right signal (207) from a mono downmix signal (204) based on spatial parameters (205). Said parametric stereo upmix being characterized in that it comprises a means (310) for predicting a difference signal (311) comprising a difference between the left signal (206) and the right signal (207) based on the mono downmix signal (204) scaled with a prediction coefficient (321). Said prediction coefficient is derived from the spatial parameters (205). Said parametric stereo upmix apparatus (300, 400) further comprises an arithmetic means (330) for deriving the left signal (206) and the right signal (207) based on a sum and a difference of the mono downmix signal (204) and said difference signal (311).

Abstract: A crosstalk suppression arrangement for a stereo audio system, wherein the stereo audio system comprises a left stereo channel and a right stereo channel, each comprising a driving circuit, an output impedance of the driving circuit comprising at least a first output impedance part, and a load impedance, and wherein the load impedances of the left and the right stereo channels are connected to a reference voltage via a common reference voltage impedance, the crosstalk suppression arrangement comprising: a crosstalk suppression impedance having an impedance value based on the reference voltage impedance, at least one of the load impedances and at least one of the output impedances, wherein the crosstalk suppression impedance is connected at one end to a point between the load impedance and the output impedance of the left stereo channel and at another end to a point between the load impedance and the output impedance of the right stereo channel.

Abstract: A method of encoding multi-channel signals having two or more channels into a first signal and a second signal, and an apparatus to perform the method, the method including generating the first signal by performing a first operation using a first channel signal in the multi-channel signals; and generating the second signal by performing a second operation using a combination of the first channel signal and a second channel signal in the multi-channel signals.

Abstract: The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

Abstract: An audio encoder (109) has a hierarchical encoding structure and generates a data stream comprising one or more audio channels as well as parametric audio encoding data. The encoder (109) comprises an encoding structure processor (305) which inserts decoder tree structure data into the data stream. The decoder tree structure data comprises at least one data value indicative of a channel split characteristic for an audio channel at a hierarchical layer of the hierarchical decoder structure and may specifically specify the decoder tree structures to be applied by a decoder. A decoder (115) comprises a receiver (401) which receives the data stream and a decoder structure processor (405) for generating the hierarchical decoder structure in response to the decoder tree structure data. A decode processor (403) then generates output audio channels from the data stream using the hierarchical decoder structure.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: A parametric representation of a multi-channel audio signal having parameters suited to be used together with a monophonic downmix signal to calculate a reconstruction of the multi-channel audio signal can efficiently be derived in a stereo-backwards compatible way when a parameter combiner is used to generate the parametric representation by combining a one or more spatial parameters and a stereo parameter resulting in a parametric representation having a decoder usable stereo parameter and an information on the one or more spatial parameters that represents, together with the decoder usable stereo parameter, the one or more spatial parameters.

Abstract: A configurable common filterbank processor applicable for various audio standards and its processing method. Inverse modified discrete cosine transform (IMDCT) and window and overlap-add (WOA) decoding operations required by AC-3 and AAC, and IMDC, WOA, and matrixing decoding operations required by MP3 are divided into several different modes, and a quick algorithm is provided for expediting the operation of these modes, and a hardware architecture is designed universally for these modes, so that the hardware architecture can be applicable for the decoding operations of three different audio standards, respectively AC-3, AAC and MP3, to expand the scope of applicability of a decoder.

Abstract: A method and a device are described for processing a stereo signal obtained from an encoder, which encodes an N-channel audio signal into spatial parameters (P) and a stereo down-mix comprising first and second stereo signals (L0, R0). A first signal and a third signal are added in order to obtain a first output signal (Low), wherein the first signal (L0wL) comprises the first stereo signal (L0) modified by a first complex function (g1), and the third signal (L0wR) comprises the second stereo signal (R0) modified by a third complex function (g3). A second signal and a fourth signal are added to obtain a second output signal (R0w). The fourth signal (R0wR) comprises the second stereo signal (R0) modified by a fourth complex function (g4), and the second signal (R0wL) comprises the first stereo signal (L0) modified by a second complex function (g2).

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein a combined cue code (e.g., a combined inter-channel correlation (ICC) code) is generated by combining two or more estimated cue codes, each estimated cue code estimated from a group of two or more channels. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels. Received cue codes include a combined cue code (e.g., a combined ICC code). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein two or more derived cue codes are derived from the combined cue code, and each derived cue code is applied to generate two or more synthesized channels.

Abstract: There was no method of positioning virtual sound sources of object signals obtained from received coded object information, in a listening space on a receiving site side.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Provided is a method for controlling synchronizing downmix signals and MPEG surround side information signals by controlling a delay according to the kind of downmix audio signals in an MPEG surround decoder. When multi-channel audio signals are restored using an HE-AAC decoder and a low-power MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. Anther delay unit delays spatial parameters to compensate for a delay caused in QMF and Nyquist banks when time-domain downmix signals are used. Also, when multi-channel audio signals are restored using an HE-AAC decoder and a high-quality MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter.

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: Inputting of a parametrically encoded audio signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information describing a multi-channel sound image and including channel configuration information is shown along with deriving, from the channel configuration information, audio source location data describing at least one of horizontal and vertical positions of audio sources in the binaural audio signal; selecting, from a predetermined set of head-related transfer function filters, a left-right pair of head-related transfer function filters matching closest to the audio source location data, wherein the left-right pair of head-related transfer function filters is searched in a stepwise motion in a horizontal plane; and synthesizing a binaural audio signal from the at least one processed signal according to side information and the channel configuration information.

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: An acoustical-signal processing apparatus includes a feature extracting unit that extracts feature data common to each channel signal which forms a multichannel acoustical signal, based on a composite similarity obtained by combining similarities calculated from each channel signal; and a time-base companding unit that executes time compression and time expansion of the multichannel acoustical signal based on the extracted feature data.

Abstract: A method comprising selecting a subset of audio sources from a plurality of audio sources, and transmitting signals from said selected subset of audio sources to an apparatus, wherein said subset of audio sources is selected in dependence on information provided by said apparatus.

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: An audio processor for converting a multi-channel audio input signal, such as a B-format sound field signal, into a set of audio output signals, such as a set of two or more audio output signals arranged for headphone reproduction or for playback over an array of loudspeakers. A filter bank splits each of the input channels into frequency bands. The input signal is decomposed into plane waves to determine one or two dominant sound source directions. The(se) are used to determine a set of virtual loudspeaker positions selected such that the dominant direction(s) coincide(s) with virtual loudspeaker positions. The input signal is decoded into virtual loudspeaker signals corresponding to each of the virtual loudspeaker positions, and the virtual loudspeaker signals are processed with transfer functions suitable to create the illusion of sound emanating from the directions of the virtual loudspeakers.

Abstract: The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: The invention discloses a decoding apparatus for decoding an analog audio signal. The decoding apparatus includes an RF tuner, an analog to digital (A/D) converter, a digital down converter, and a programmable digital signal processor (DSP). The RF tuner is used for receiving the analog audio signal and for providing an analog sound intercarrier frequency (SIF) signal indicative thereof. The analog to digital (A/D) converter is used for sampling the analog SIF signal and for converting the signal into a digital SIF signal. The digital down converter is used for down converting the digital SIF signal to generate a baseband signal. The programmable digital signal processor (DSP) is used for demodulating the baseband signal according to a demodulation procedure of a predetermined standard and decoding the demodulated baseband signal to output an output signal in compliance with a decoding procedure of the predetermined standard.

Abstract: A multi-channel audio signal decoding method and device is provided. The multi-channel audio signal decoding method includes receiving a first multi-channel audio signal; performing a first decoding procedure on the first multi-channel audio signal to generate a second multi-channel audio signal; performing a second decoding procedure on a first single-channel audio data of the second multi-channel audio signal to generate a first single-channel audio signal when the first single-channel audio data belongs to a first classification; and performing a third decoding procedure on a second single-channel audio data of the second multi-channel audio signal to generate a second single-channel audio signal when the second single-channel audio data belongs to a second classification. The number of instructions of the third decoding procedure is less than that of the second decoding procedure.

Abstract: A television audio signal encoder includes a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission.

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: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: A stereo decoder includes a first digital filter for producing a first filtered composite audio signal and a second digital filter for producing a second filtered composite audio signal. A sum/difference network produces a left channel signal and a right channel signal. A processor executes operational instructions that calculate a first corner frequency of a first digital filter based on a signal quality of the FM signal, calculate a first set of filter coefficients for the first digital filter based on the first corner frequency and an approximation of a filter transform, calculate a second corner frequency of a second digital filter based on the signal quality, and calculate a second set of filter coefficients for the second digital filter, based on the second corner frequency and the approximation of the filter transform.

Abstract: The present invention relates to a method of processing a media signal and apparatus therefor. A media signal decoding method according to the present invention includes detecting a channel having a valid value of the multi-channels to be generated and generating the detected channel having the valid value from the downmix signal and the spatial information signal. Accordingly, the present invention is able to reduce a decoding operation quantity by detecting which one of the channels to be generated from a transferred media signal is set to a virtual value and omitting decoding for the generation of the channel set to the virtual value.

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.