Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that is to be applied to the entire enhanced signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter for reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: At least one exemplary embodiment is directed to a new spatial audio enhancing system including a novel Adaptive Sound Upmixing System (ASUS). In some specific embodiments the ASUS provided converts a two-channel recording into an audio signal including four channels that can be played over four different loudspeakers. In other specific embodiments the ASUS provided converts a two-channel recording into an audio signal including five channels that can be played over five different loudspeakersn even other specific embodiments the ASUS provided converts a five-channel recording (such as those for DVD's) into an audio signal including eight channels that can be played over eight different loudspeakers. More generally, in view of this disclosure those skilled in the art will be able to adapt the ASUS to process and provide an arbitrary number of audio channels both at the input and the output.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. Method and systems provided for processing enhancing bass effect in audio signals. Said method and systems result in the bass enhancement being computationally less intensive. The bass effect enhancement techniques described are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on response of an exponential transfer function.

Abstract: In summary, this application describes a psycho-acoustically motivated, parametric description of the spatial attributes of multichannel audio signals. This parametric description allows strong bitrate reductions in audio coders, since only one monaural signal has to be transmitted, combined with (quantized) parameters which describe the spatial properties of the signal. The decoder can form the original amount of audio channels by applying the spatial parameters. For near-CD-quality stereo audio, a bitrate associated with these spatial parameters of 10 kbit/s or less seems sufficient to reproduce the correct spatial impression at the receiving end.

Abstract: A delay time calculation apparatus enabling all the speaker units of a delay array type speaker array to contribute to formation of a combined wavefront. Sound receiving points for acoustic waves output from the speaker units are set within a target area for the acoustic waves. For each speaker unit, an average value of differences between distances between the sound receiving points for the speaker units and other speaker units and distances between the sound receiving points for the speaker units and the speaker units is determined, and an average value of differences between distances from the other speaker units to sound receiving points for the other speaker units and distances from the speaker units to the sound receiving points for the other speaker units is determined. An average of the average values is converted into a delay time, thereby determining the delay time for each speaker unit.

Abstract: An apparatus and method of reproducing a virtual sound of two channels which adaptively reproduces a 2-channel stereo sound signal reproduced through a recording medium such as DVD, CD, or MP3 player etc., based on a listener's position. The method includes sensing a listener's position and recognizing distance and angle information about the listener's position, determining output gain values and delay values of two speakers based on the distance and angle information about the sensed listener's position and selecting localization filter coefficients in a predetermined table, and updating filter coefficients of a localization filter based on the selected localization filter coefficients and adjusting output levels and time delays of the two speakers from the determined gain values and delay values.

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 system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L?R signals.

Abstract: An exemplary embodiment of the invention can generate multiple output audio signals from multiple input audio signals, in which the number of output signals is equal to or higher than the number of input signals. The embodiment includes computing one or more independent sound subbands representing signal components which are independent between the input subbands; computing one or more localized direct sound subbands representing signal components which are contained in more than one of the input subbands and direction factors representing the ratios with which these signal components are contained in two or more input subbands; generating the output subband signals, where each output subband signal is a linear combination of the independent sound subbands and the localized direct sound subbands; and converting the output subband signals to time domain audio signals.

Abstract: Provided are a method and apparatus for sinusoidal audio coding, which employs a tracking method for further effective coding of sinusoids extracted in the process of a sinusoidal analysis of parametric coding. The sinusoidal audio coding method includes: extracting sinusoids of a current frame by performing a sinusoidal analysis on an input audio signal; with respect to each of the extracted sinusoids, setting a mode selected from a birth mode in which a sinusoid is newly generated irrespective of sinusoids of a previous frame, a continuation mode in which the sinusoid is only one sinusoid continued from one of the sinusoids of the previous frame, and a branch mode in which the sinusoid is one of a plurality of sinusoids continued from one of the sinusoids of the previous frame; and coding the extracted sinusoids according to the selected mode. Accordingly, a plurality of sinusoids that can be continued from one previous track component are set to the continuation mode or the branch mode.

Abstract: An electrical signal derived from an audio source is passed through an amplifier and is then divided into two discrete signal paths. The signal traveling along the first path travels through a high pass filter and is then split so that a portion is delayed and a portion passes through a harmonics generator so that the harmonics are advanced ahead of the delayed portion from the high pass filter. The split circuit is then combined in a summing amplifier. The second portion of the signal is also delayed before it also travels to the mixer. The combined signal is passed through an amplifier and then fed into an output transducer. The use of a delay both on the input signal and the portion of the signal which passed through the high pass filter output just before the summing amplifier enables the harmonics to not be masked by the lower frequencies, making the effect more apparent.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on the response of an exponential transfer function.

Abstract: The invention relates to a method and device for reproducing sound from a first input audio signal (1) using a plurality of first loudspeakers (4) and producing a target binaural impression to a listener (6) within a listening area (55).

Abstract: An acoustic processing device, includes an inputting section which receives a first audio signal and a second audio signal, a first input signal amplifying section which amplifies the first audio signal at an amplification factor, a second input signal amplifying section which amplifies the second audio signal at an amplification factor, a first initial delayed amplifying section, a first adding section which adds the first audio signal amplified by the first amplifying portion and the first audio signal amplified by the first input signal amplifying section, a second adding section which adds the first audio signal amplified by the second amplifying portion and the second audio signal amplified by the second input signal amplifying section, a first supplying section which supplies an audio signal output from the first adding section to a first sound emitting section, and a second supplying section which supplies an audio signal output from the second adding section to a second sound emitting section differen

Abstract: A sound device includes an input terminal for electronic signal input; a line out terminal adapted to output electronic signals input on the input terminal to an external area; a delay arrangement adapted to delay electronic signals input on the input terminal for a certain time; a main output arrangement adapted to output electronic signals delayed by the delay arrangement. The delay arrangement delays signals in order to accommodate a time difference between (1) electronic signals output from an external sound device connected with the line output terminal and (2) electronic signals output from the main output arrangement. The time difference between sound that is output from the sound device and the one that is output from an external sound device that is connected with line connection terminal can be eliminated. Thus, it can prevent sound quality decrease due to time difference from happening.

Abstract: Provided are an array speaker system and a method of implementing the same. The array speaker system includes a group setting unit setting one or more combinations of a plurality of speakers included in an array speaker system into at least one group according to a predetermined adjustment factor, a signal assignment unit dividing an input signal into at least one sound source signal according to the adjustment factor and selectively assigning the divided sound source signal(s) to the set group(s), a sound adjustment unit adjusting a sound source signal to be output through speakers within the group according to sound source characteristics of the group, and a signal output unit outputting the adjusted sound source signal through the speakers within the group.

Abstract: Provided is a sound image localization control apparatus allowing, when sound is reproduced so as to perform sound image localization for a plurality of users, each of the plurality of users to variably adjust an acoustical effect individually without diminishing a sound image localization effect. The sound image localization control apparatus includes a processing characteristic setting means (13; 14) setting a processing characteristic in a controlling means (12), such that each of acoustic transfer functions, for at least two predetermined positions, represents a desired characteristic. Further, the controlling means (12) of sound image localization control apparatus receives an acoustic signal and the processing characteristic which is set by the processing characteristic setting means and performs signal processing. Additionally, the sound image localization control apparatus includes a sound reproducing means (3) receiving an output from the controlling means.

Abstract: A method of generating parameters representing Head-Related Transfer Functions, the method comprising the steps of a) sampling with a sample length (n) a first time-domain HRTF impulse response signal using a sampling rate (fs) yielding a first time-discrete signal, b) transforming the first time-discrete signal to the frequency domain yielding a first frequency-domain signal, c) splitting the first frequency-domain signal into sub-bands, and d) generating a first parameter of the sub-bands based on a statistical measure of values of the sub-bands.

Abstract: The method and apparatus described herein generates realistic audio for a virtual reality simulation based on the position (location and orientation) of a participant's head. The audio may be generated based on independent and dependent audio profiles. The independent audio profile represents the participant-independent propagation of sound from a virtual source to each of one or more virtual objects in the simulation. The dependent audio profile represents the propagation of the sound from each of the one or more virtual objects to the head or ears of the participant based on a position of the participant's head or ears. An audio processor generates the desired audio signal at the head of the participant by combining the dependent and independent audio profiles to determine a total audio profile for the virtual source, and filtering an audio wave corresponding to the virtual source based on the total audio profile.

Abstract: A plural-channel audio signal (e.g., a stereo audio) is processed to modify a gain (e.g., a volume level or loudness) of an estimated dialogue signal (e.g., dialogue spoken by actors in a movie) relative to other signals (e.g., reflected or reverberated sound). In some aspects, a classifier is used to classify component signals in the plural-channel audio signal or the estimated dialogue signal. In some aspects, a desired volume level for the dialogue signal is maintained relative to the plural-channel audio signal or other component signals.

Abstract: In one embodiment, C input audio channels are encoded to generate E transmitted audio channel(s), where one or more cue codes are generated for two or more of the C input channels, and the C input channels are downmixed to generate the E transmitted channel(s), where C>E?1. One or more of the C input channels and the E transmitted channel(s) are analyzed to generate a flag indicating whether or not a decoder of the E transmitted channel(s) should perform envelope shaping during decoding of the E transmitted channel(s). In one implementation, envelope shaping adjusts a temporal envelope of a decoded channel generated by the decoder to substantially match a temporal envelope of a corresponding transmitted channel.

Abstract: An apparatus for processing audio data comprising an interaural time delay correction factor unit for receiving a plurality of channels of audio data and generating an interaural time delay correction factor. An interaural time delay correction factor insertion unit for modifying the plurality of channels of audio data as a function of the interaural time delay correction factor.

Abstract: A method for stereo expansion includes a step to remove the effects of actual relative speaker to listener positioning and head shadow and a step to introduce an artificial effect based on a desired virtual relative speaker to listener positioning using the inter-aural delay and the head-shadow models for the virtual speakers at desired angles relative to the listener thereby creating the impression of a widened and centered sound stage and an immersive listening experience. Known methods drown out vocals and add mid-range coloration thereby defeating equalization. The present method includes the integration of a novel binaural listening model and speaker-room equalization techniques to provide widening while not defeating equalization.

Abstract: A speaker array apparatus includes a speaker array that emits sounds of a plurality of channels, a beam formation calculating section that performs a calculation for controlling phases of the sounds so that the speaker array emits sound beams in directions set for the respective channels, a sound source localization applying section that performs a calculation for controlling the phases of the sounds emitted from the speaker array so as to form a plurality of virtual point sound sources, and performs a calculation of auditory sensation characteristics at a listening position on a basis of a head-related transfer function, a selecting section that selects one of the beam formation calculating section and the sound source localization applying section, and a phase controlling section that controls the phases of the sounds emitted from the speaker array on a basis of a calculation result of the beam formation calculation section which is selected by the selecting section or applies the auditory sensation charact

Abstract: A sound processor includes N (N is an integer of five or more) speakers, a sound source configured to output N sound signals an additional sound source configured to output an additional sound signal and a coefficient data input section configured to input N pieces of position information indicating respectively positions of the N speakers and N coefficients indicating respectively volumes of sounds outputted from the N speakers based on the additional sound signal a coefficient data analysis section configured to generate M (M is an integer equal to or greater than two and smaller than N) adjustment coefficients based on said N pieces of position information and the N coefficients wherein the M adjustment coefficients indicate volumes of sounds outputted from M speakers of the N speakers based on the additional sound signal.

Abstract: An apparatus for processing an input signal, wherein the input signal comprises a plurality of subsignals associated to discrete transmitters or receivers, wherein the discrete transmitters or receivers are disposed at predetermined geometrical positions with regard to a room, comprises a means for providing a plurality of wave-field components, wherein a superposition of the plurality of wave-field components results in a composite wave field, wherein the composite wave field can be propagated in the room, wherein the plurality of wave-field components are derived from the input signal by wave field decomposition based on orthogonal wave field base functions and the predetermined geometrical positions, a plurality of single filters, wherein a wave-field component of the plurality of wave-field components is associated to a single filter, wherein the single filter is formed to influence the associated wave-field component such that with regard to the plurality of single filters a plurality of filtered wave-fi

Abstract: This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

Abstract: An audio output device includes: a speaker unit including at least first to third speakers; and an audio signal processor configured to select either one of a first mode for outputting a stereo audio signal from the first and second speakers and a second mode for outputting a monaural audio signal from the third speaker and the first or second speaker.

Abstract: The sound field reproducing apparatus includes: a signal obtaining unit which obtains input audio signals of plural channels; an effect adding unit which gives an effect on an output to a sound field; and an effect amount control unit which controls an amount of the effect based on a characteristic of the input audio signal. The sound field reproducing apparatus receives input audio signals of plural channels from a medium such as a DVD, and gives a sound effect to them to output. The amount of the effect given to the input audio signal is controlled based on the characteristic of the input audio signal, especially the cross correlation of the reverberation component between the channels.

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: A sound field controlling device for supplying audio signals to a plurality of speakers provided in a space to form a sound field in the space, includes a measuring unit which measures levels of indirect sounds, which are outputted from the speakers, reflected from a wall surface of the space, and reach a listening position respectively, a reverberation applying unit which generates a reverberation simulation signal for reinforcing the indirect sounds on the basis of the audio signals, and a reverberation balance adjusting unit which controls the level of the reverberation simulation signal and supplies the controlled reverberation simulation signal to the corresponding speakers on the basis of the levels of the indirect sounds outputted from the speakers so that respective synthesized levels of the indirect sounds and the reverberation simulation signal are balanced between the speakers.

Abstract: To provide an audio signal processing system, a surround signal generation method and so on that are capable of obtaining stable output signal level and feeling of soundscape without depending on an audio input signal. A DSP inputs audio input signals respectively corresponding to a plurality of channels, and generates surround signals that have reflection sound and reverberate sound and that respectively correspond to the plurality of channels. The DSP generates the surround signal corresponding to one channel based on an audio input signal corresponding to the one channel, a variable number changing within the predetermined range in response to a signal level of the surround signal at a preceding predetermined time corresponding to the one channel, and the audio input signal corresponding to the other channel.

Abstract: An audio apparatus includes an input unit for inputting stereo signals in which multi-channel surround-sound signals are mixed in accordance with a predetermined algorithm and a parameter indicating encoded contents of the algorithm, a surround-sound signal generator for generating left-side surround-sound signals and right-side surround-sound signals by decorrelating left-side signals and right-side signals contained in the stereo signals, and a controller for controlling the decorrelation performed by the surround-sound signal generator based on the parameter.

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: A speaker array is provided which can increase the orientation when reproducing front channels of a surround sound, increase the density effect, improve the narrow directivity when reproducing a stereo sound and increase the selectivity in selecting reproduction methods which match setting environments. When reproducing a stereo sound by a speaker array for reproducing a surround sound by converting a sound into a beam of sound, the speaker array is divided into a sound reproducing region for an L system and a sound reproducing region for an R system at a central portion thereof. Then, each of the reproducing regions so divided is further divided in to bands. In addition, since a high frequency reproducing region has a high directivity and a strong orientation when the sound is attempted to be reproduced simultaneously by the plurality of speakers as is described above, a reproducing region is limited to part of the reproducing regions.

Abstract: Systems, methods, and apparatus, including computer program products, for audio editing are provided. In some implementations, a method is provided. The method includes receiving audio data having a first audio channel and a second audio channel. The audio data is separated, into a plurality of blocks. An amount of misalignment is determined between the first audio channel and the second audio channel for the portion of the audio data in each block using a phase difference between the first and second audio channels for each of a plurality of frequency bands. The first and second channels are aligned using the determined misalignment.

Abstract: A sound device includes an input terminal for electronic signal input; a line out terminal adapted to output electronic signals input on the input terminal to an external area; a delay arrangement adapted to delay electronic signals input on the input terminal for a certain time; a main output arrangement adapted to output electronic signals delayed by the delay arrangement. The delay arrangement delays signals in order to accommodate a time difference between (1) electronic signals output from an external sound device connected with the line output terminal and (2) electronic signals output from the main output arrangement. The time difference between sound that is output from the sound device and the one that is output from an external sound device that is connected with line connection terminal can be eliminated. Thus, it can prevent sound quality decrease due to time difference from happening.

Abstract: A headphone down mix signal can be efficiently derived from a parametric down mix of a multi-channel signal, when modified HRTFs (head related transfer functions) are derived from HRTFs of a multi-channel signal using a level parameter having information on a level relation between two channels of the multi-channel signals such that a modified HRTF is stronger influenced by the HRTF of a channel having a higher level than by the HRTF of a channel having a lower level. Modified HRTFs are derived within the decoding process taking into account the relative strength of the channels associated to the HRTFs. The HRTFs are thus modified such that a down mix signal of a parametric representation of a multi-channel signal can directly be used to synthesize the headphone down mix signal without the need of an intermediate full parametric multi-channel reconstruction of the parametric down mix.

Abstract: A sound field in a vehicle can be further readily and optimally corrected without forcing a user to do troublesome work. As an embodiment of the present invention, a measurement sound emitted through a left speaker or a right speaker at a further position in a view from a driver's listening point that was assumed on the head touching surface of the headrest of the driver's seat or the passenger seat is picked up, with a microphone provided as buried on a front panel of a head unit to be attached to the almost center position between the driver's seat and the passenger seat in the vehicle. And the frequency characteristic of the above sound is corrected, based on a tendency that the frequency characteristic of an audio sound reaching from the left speaker or the right speaker to the microphone is almost approximate to the virtual frequency characteristic of an audio sound that will reach from the left speaker or the right speaker to the driver's listening point assumed on the head touching surface.

Abstract: An audio signal processing apparatus includes a localization direction detector that detects localization directions of two-channel input audio signals, a localization direction distribution calculator that calculates a distribution of the localization directions detected by the localization direction detector, a gain table information recording unit that records gain table information defining weights corresponding to respective localization angles, a gain generator that generates a gain corresponding to an output audio signal on the basis of the distribution calculated by the localization direction distribution calculator and the gain table information recorded in the gain table information recording unit, and a synthesizing unit that synthesizes the two-channel input audio signals using the gain generated by the gain generator.

Abstract: According to the present invention, multiple parametrically encoded audio signals can be efficiently combined using an audio signal generator, which generates an audio output signal by combining the down-mix channels and the associated parameters of the audio signals directly within the parameter domain, i.e. without reconstructing or decoding the individual input audio signals prior to the generation of the audio output signal. This is achieved by direct mixing of the associated down-mix channels of the individual input signals. It is one key feature of the present invention that the combination of the down-mix channels is achieved by simple, computationally inexpensive arithmetic operations.

Abstract: In an audio signal reproduction device and an audio signal reproduction system of the present invention, an SLch sound is reproduced as a mixed sound that is adjusted appropriately by two speakers 5a, 5d located at Lch and SBLch speaker positions, and an SRch sound is reproduced as a mixed sound that is adjusted appropriately by two speakers 5b, 5e located at Rch and SBRch speaker positions. This configuration can achieve 5.1ch virtual reproduction with high sound quality in which sounds for 5.1 channels of LPCM 7.1ch audio information included in the contents are output as they are and sounds for the remaining 2 channels are reproduced artificially even if speakers compatible with the 5.1ch sound field system are connected.

Abstract: An audio signal is supplied to a loudspeaker array to perform wavefront synthesis. A virtual sound source is produced at an infinite distance using wavefront synthesis. A propagation direction of a sound wave emitted from the virtual sound source is changeable.

Abstract: An ASC generates a control signal based on an engine rotation frequency detected by an engine rotational speed detecting means, and a speaker outputs a sound effect depending on the control signal into a vehicle. The speaker is disposed in a substantially central area of a dashboard, in front of a driver seat and a front passenger seat of the vehicle, in the transverse direction of the vehicle.

Abstract: An audio signal processing circuit comprising: a holding circuit configured to receive a clock signal and set data corresponding to the clock signal, and to hold the set data; a processing circuit configured to process at least one of a first audio signal and a second audio signal input in parallel, based on the set data of the holding circuit; and a set data output circuit configured to output the clock signal to the holding circuit based on the first audio signal corresponding to the clock signal, and output the set data to the holding circuit based on the second audio signal corresponding to the set data.

Abstract: A signal processing device includes a filter that is set to frequency characteristics in which a dip existing in an intermediate and high frequency range is smoothed in the frequency characteristics of a virtual characteristic applying filter for applying transfer characteristics of a space transfer path to a sound signal, the space transfer path extending from a virtually set orientation of a sound image to an ear of a listener, an equalizer that forms the dip by cutting a part of the intermediate and high frequency range, and an adjusting unit that adjusts at least a central frequency of the dip. An input signal is passed through the filter and the equalizer.

Abstract: An acoustic image localization apparatus according to the present invention that outputs sound from a plurality of speakers so as to localize an acoustic image at a predetermined position on a space as viewed from a listener, the acoustic image localization apparatus comprising: amplitude characteristic adjusting means for adjusting an amplitude frequency characteristic of an inputted acoustic signal such that the acoustic image is localized at a position rotated by a first angle about a position of a listener toward an upper direction from a facing position of the listener; and a plurality of level adjusting means, provided so as to respectively correspond to the plurality of speakers, for adjusting a level of the acoustic signal outputted from the amplitude characteristic adjusting means and for outputting, to a corresponding speaker, the acoustic signal whose level has been adjusted, wherein each of the level adjusting means adjusts the level of the acoustic signal, which is outputted from the amplitude ch

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: A digital signal processing apparatus in which a first digital filter reproduces that part of an impulse response that responds fast, and a decimation filter converts the output of a delay device of the first digital filter to a digital signal having a sampling rate of ½. The digital signal is supplied to the second digital filter that reproduces that part of the impulse response that responds slowly and outputs data representing the response characteristic of this part of the impulse response. An interpolation filter converts an input signal to a signal having the same sampling rate as the digital audio signal input to the digital signal processing apparatus, and the output signal of the interpolation filter is supplied to an adder circuit.

Abstract: Audio signals that represent a sound field with increased spatial resolution are obtained by deriving signals that represent the sound field with high-order angular terms. This is accomplished by analyzing input audio signals representing the sound field with zero-order and first-order angular terms to derive statistical characteristics of one or more angular directions of acoustic energy in the sound field. Processed signals are derived from weighted combinations of the input audio signals in which the input audio signals are weighted according to the statistical characteristics. The input audio signals and the processed signals represent the sound field as a function of angular direction with angular terms of one or more orders greater than one.