Abstract: A radio configured to dynamically control cancellation of undesired signals in an audio stream. The radio includes a noise cancellation processor configured to receive an audio stream from a user and to alter information in the audio stream by filtering out undesired signals in the audio stream. The radio also includes a receiving component configured to receive a data packet from a remote device, to retrieve configuration information from the data packet, and to dynamically apply the configuration information, while the radio is being used by a user, to settings associated with the noise cancellation processor. A dynamically enabled noise cancellation processor suppresses undesired signals associated with a subsequent incoming audio stream provided by the user and transmits at least one of an altered audio stream or an unaltered audio stream to the remote device.

Abstract: A surround system includes a decoder that decodes an encoded audio data stream, a decorrelation unit that receives and decorrelates stereo signals decoded by the decoder so as to generate surround signals having a low-correlation component, an addition unit that adds high-correlation-component signals extracted from the stereo signals to the surround signals generated by the decorrelation unit, and a controller that controls addition of the high-correlation-component signals performed by the addition unit on the basis of the bit rate of the audio data stream.

Abstract: Provided is an audio coding apparatus and method that can selectively apply a operation mode of a coding module for stereo or multi-channel representation according to input signal characteristics of each channel, when voice or music signals are transmitted using an audio codec in portable terminals capable of stereo or multi-channel input and output. The audio coding apparatus includes a down-mixer for down-mixing multi-channel audio signals into mono signals; a coder for coding the mono signals; an input channel correlation analyzer for deciding whether to give them stereo effect based on their signal distribution characteristics, and outputting a control signal indicating whether to perform stereo representation process; and a stereo representation unit for performing stereo representation process onto the multi-channel audio signals when the control signal indicating to perform stereo representation process.

Abstract: A device for determining a component signal for a WFS system includes a provider for providing WFS parameters, a WFS parameter interpolator, and an audio signal processor. The provider provides WFS parameters for a component signal while using a source position and while using the loudspeaker position at a parameter sampling frequency smaller than the audio sampling frequency. The WFS parameter interpolator interpolates the WFS parameters so as to produce interpolated WFS parameters which are present at a parameter interpolation frequency that is higher than the parameter sampling frequency, the interpolated WFS parameters having interpolated fractions which have a higher level of accuracy than is specified by the audio sampling frequency. The audio signal processor is configured to apply the interpolated fractional values to the audio signal such that the component signal is obtained in a state of having been processed at the higher level of accuracy.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A method of processing a signal is disclosed, which includes receiving a downmix signal, object information and rendering information; and generating downmix processing information using the object information and the rendering information, the object information including modified object level information, wherein the modified object level information is determined using a level of a modified object generated by applying a downmix gain to an object. Accordingly, in an apparatus for processing a signal and method thereof according to the present invention, the signal is decoded using object information including modified object level information. Therefore, since a downmix gain is not transmitted, the present invention is able to reduce a used bit for object information and decode a signal using less information.

Abstract: A method for encoding multiple directional audio signals using an integrated codec by a wireless communication device is disclosed. The wireless communication device records a plurality of directional audio signals. The wireless communication device also generates a plurality of audio signal packets based on the plurality of directional audio signals. At least one of the audio signal packets includes an averaged signal. The wireless communication device further transmits the plurality of audio signal packets.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: A method and a system are described providing multiple display systems with an enhanced acoustics experience. A source audio signal having a plurality of source audio channels is generated from an audio signal source. The system includes a plurality of speakers connected to a plurality of display systems. A speaker configuration gatherer determines the spatial configuration of the speakers. An audio signal processor is provided to generate synthesized audio signal based on the contents of the source audio signal and spatial configuration of the speakers. The synthesized audio signal is mapped and delivered to the speakers to produce an enhanced sound field.

Abstract: A device for processing a signal includes a processing module used to process data in a domain. The device includes domain conversion hardware and memory, both separate from the processing module. The memory receives and sends data represented in a first domain to the domain conversion hardware. The domain conversion hardware converts the first domain data into the second domain data that may be used or processed by the processing module. The second domain data is then accessed and used by the processing module for processing.

Abstract: An encoder for encoding an audio signal comprising at least two channels, the encoder configured to generate an encoded signal comprising at least a first part, a second part and a third part, wherein the encoder is further configured to: generate the first part of the encoded signal dependent on at least one combination of first and second channels of the at least two channels; generate the second part of the encoded signal dependent on at least one difference between the first and second channels of the at least two channels; and generate the third part of the encoded signal dependent on at least one energy ratio of the first and second channels of the at least two channels.

Abstract: An audio processing apparatus and method for a mobile device are provided. The audio processing apparatus and method may appropriately determine sound source localizations corresponding to a voice signal and an audio signal, and thereby may simultaneously provide a voice call service and a multimedia service. Also, the audio processing apparatus and method may guarantee quality of the voice call service even when simultaneously providing the voice call service and the multimedia service.

Abstract: A method of decoding an audio signal is disclosed, The present invention includes the steps of receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal, extracting the ambient component signal and the source component signal of each of the channels based on correlation between the channel signals, modifying the ambient component signal using surround effect information, and generating the audio signal including a plurality of channels using the modified ambient component signal and the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal.

Abstract: A device for determining a component signal for a WFS system includes a provider for providing WFS parameters, a WFS parameter interpolator, and an audio signal processor. The provider provides WFS parameters for a component signal while using a source position and while using the loudspeaker position at a parameter sampling frequency smaller than the audio sampling frequency. The WFS parameter interpolator interpolates the WFS parameters so as to produce interpolated WFS parameters which are present at a parameter interpolation frequency that is higher than the parameter sampling frequency, the interpolated WFS parameters having interpolated fractions which have a higher level of accuracy than is specified by the audio sampling frequency. The audio signal processor is configured to apply the interpolated fractional values to the audio signal such that the component signal is obtained in a state of having been processed at the higher level of accuracy.

Abstract: A device for processing parameters representing Head-Related Transfer Functions includes an input stage configured to receive audio signals of sound sources, a determinor configured to receive reference parameters representing Head-Related Transfer Functions and configured to determine, from the audio signals, position information representing positions and/or directions of the sound sources. A processor is configured to process the audio signals; and an influencer is configured to influence the processing of the audio signals based on the position information yielding an influenced output audio signal.

Abstract: A head-related transfer function measurement method includes the steps of: first measuring, including placing an acousto-electric conversion unit nearby both ears of a listener where placement of an electro-acoustic conversion unit is assumed, picking up sound waves emitted at a perceived sound source position with the acousto-electric conversion unit in a state with a dummy head or a human at the listener position, and measuring a head-related transfer function from only the sound waves directly reaching the acousto-electric conversion unit; second measuring, including picking up sound waves emitted at a perceived sound source position with the acousto-electric conversion unit, with no dummy head or human at the listener position, and measuring a natural-state transfer property from only the sound waves directly reaching the acousto-electric conversion unit; normalizing the head-related transfer function with the natural-state transfer property to obtain a normalized head-related transfer function; which is

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of generating source mapping information corresponding to each source of multi-sources by using spatial information indicating features between the multi-sources; generating at least one rendering information by using the source mapping information and filter information having a surround effect; and performing interpolation by using neighbor rendering information of the at least one rendering information.

Abstract: A system and method for generating a sound that simulates a fictitious sound perceived to emanate from a fictitious sound location is disclosed. In at least one aspect, the system includes a first surface serving to at least partially enclose a first region, a first speaker positioned on the first surface, a second region within the first region, and a sound reflecting boundary to enclose the first region. The system further includes a first fictitious source location within the first region, the location being outside of the second region, and a control device coupled at least indirectly within the first speaker. The control device generates control signals configured to cause the first speaker to generate sounds that in turn produce actual sounds that simulate fictitious sounds perceived to emanate from fictitious sound locations.

Abstract: A head-related transfer function (HRTF) convolution method arranged, when an audio signal is reproduced acoustically by an electro-acoustic conversion unit disposed in a nearby position of both ears of a listener, to convolute an HRTF into the audio signal, which allows the listener to listen to the audio signal such that a sound image is localized in a perceived virtual sound image localization position, the method including the steps of: measuring, when a sound source is disposed in the virtual sound image localization position, and a sound-collecting unit is disposed in the position of the electro-acoustic conversion unit, a direct-wave direction HRTF regarding the direction of a direct wave, and reflected-wave direction HRTFs regarding the directions of selected one or more reflected waves, from the sound source to the sound-collecting unit, separately beforehand; and convoluting the obtained direct-wave direction HRTF, and the reflected-wave direction HRTFs into the audio signal.

Abstract: An audio reproducing system includes a pair of speaker units, a mounting unit for mounting the pair of speaker units, without being attached to a baffle board, to the vicinity of a listener's ears in a manner such that sounds emitted from the front and back of a diaphragm of each speaker unit are mixed, and an audio signal output unit for virtual sound imaging an input audio signal and outputting the virtual sound imaged signal to the pair of speaker units in a manner such that the listener listens to a sound reproduced by the pair of speaker units feeling as if the sound is emitted from a different speaker device.

Abstract: A down-mixing device is provided. 5.1-channel sounds are down-mixed into 4-channel sounds with high-quality multi-channel surrounding sound effect reappeared. A common-acoustic-pole and zero (CAPZ) model is used to rebuild frequency responses of original head-related transfer functions (HRTF). Direct sounds, reflection sounds and late reverberation in a spatial sound field are gathered. Thus, varieties of sounds become rich and 3-dimensional hearing experiences are enhanced with low cost for a 4-channel earphones system.

Abstract: A method for simulating spatially extended sound sources comprising: panning a first input signal over a plurality of output channels to generate a first multi-channel directionally encoded signal; panning a second input signal over the plurality of output channels to generate a second multi-channel directionally encoded signal; combining the first and second multi-channel directionally encoded signals to generate a plurality of loudspeaker output channels; and applying a bank of decorrelation filters on the loudspeaker output channels.

Abstract: Provided are sound field reproducing apparatus and method. The sound field reproducing apparatus includes an input unit for receiving reflection reproducing information, a signal processor for selecting loudspeakers and calculating reflection signal for applying a panning algorithm based on the reflection reproducing information, a signal treatment unit for localizing the calculated reflection signal on a virtual sound image according to the panning algorithm; and a reproducer for reproducing the localized reflection signals through the selected loudspeakers.

Abstract: An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving a downmix signal generated from a plural-channel signal, mix information and phase shift information on the plural-channel signal, upmixing the downmix signal into the plural-channel signal by applying the mix information to the downmix signal, and generating an original plural-channel signal by shifting a phase of at least one channel of the plural-channel signal based on the phase shift information. According to the present invention compensate a reconstructed original plural-channel signal using compensation information (phase shift information), thereby compensating a phase or a gain lost in the plural-channel signal reconstructed by upmixing using mix information.

Abstract: An apparatus and method for generating a panoramic sound are provided. The panoramic sound generation apparatus may include a panning coefficient calculation unit to calculate a panning coefficient that represents directivity of a sound source using an input signal, a masker determination unit to determine a direction masker that extracts a sound source of a desired direction based on the panning coefficient, and a channel separation unit to separate to be used as an output signal, output to a sound output device, using the direction masker.

Abstract: A front surround reproduction system improving the stereo effect of mid and low frequency signals by using a psychoacoustic model, and a method thereof.

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.

Abstract: An audio information processing apparatus for processing input audio information is adapted to input an audio signal composed of front area audio information that should be input to a front speaker and rear area audio information that should be input to a plurality of speakers. A mixer is adapted to mix the rear area audio information in accordance with an instruction from an instruction circuit.

Abstract: The invention relates to sonic steerable antennae and their use to achieve a variety of effects. The invention comprises a method and apparatus for taking an input signal, replicating it a number of times and modifying each of the replicas before routing them to respective output transducers such that a desired sound field is created. This sound field may comprise a directed beam, focus beam or a simulated origin. Further, “anti-sound” may be directed so as to create nulls (quiet spots) in an already existing sound field. The input signal replicas may also be modified in way which changes their amplitude or they may be filtered to provide the desired delaying.

Abstract: There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.

Abstract: A stereo sound generation apparatus and method of reproducing multi-channel sound input signals through two-channel speakers.

Abstract: An acoustic signal output control section extracts, from an acoustic signal, a low-frequency-component acoustic signal and a high-frequency-component acoustic signal. A sound image localization processing filter section performs a sound image localization process on the low-frequency-component acoustic signal. The low-frequency-component acoustic signal having been subjected to the sound image localization process is outputted from loudspeakers in an AV rack loudspeaker apparatus. A television sound image localization processing filter section performs a sound image localization process on the high-frequency-component acoustic signal. The high-frequency-component acoustic signal having been subjected to the sound image localization process is outputted from loudspeakers in a television via HDMI communication sections.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, a first multi-channel information, and an object information; processing the downmix signal using the object information and a mix information; and, transmitting one of the first multi-channel information and a second multi-channel information according to the mix information, wherein the second channel information is generated using the object information and the mix information is disclosed.

Abstract: A spatial audio system for implementing a head-related transfer function (HRTF). A first stage implements a lateral HRTF that reproduces the median frequency response for a sound source located at a particular lateral distance from a listener, and second stage implements a vertical HRTF that reproduces the spectral changes when the vertical distance of a sound source changes relative to the listener. The system improves the vertical localization accuracy provided by an arbitrary measured HRTF by introducing an enhancement factor into the second processing stage. The enhancement factor increases the spectral differentiation between simulated sound sources located at different positions within the same “cone of confusion.

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: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of generating rendering information by using spatial information indicating features between multi-sources and filter information having a surround effect; and generating a surround signal by applying the rendering information to a downmix signal generated by downmixing the multi-sources, wherein a tab number of a filter used in applying the rendering information to generate the surround signal is variable.

Abstract: An apparatus for spatially separating sounds from at least two sound sources includes a sound detection unit and a sound control unit. The sound detection unit detects a first sound from a first sound source while a second sound is outputted from a second sound source. The sound control unit performs a head-related transfer function (HRTF) on the second sound to move the second sound to a user's desired position according to the detection result, thereby the second sound is set as a background sound.

Abstract: To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.

Abstract: A system and method for rendering a virtual sound source using a plurality of speakers in an arbitrary arrangement includes expanding a wave field of the virtual sound source and a composite wave field generated by the plurality of speakers into multi-pole expansions, and equating the multi-pole expansions to each other to form a continuous equation. A set of constraints is applied to the continuous equation that minimize an error in a sweet spot region resulting in a linear system of equations. The linear system of equations is solved using singular-value decomposition to arrive at a pseudo-inverse representing a multi-dimensional transfer function between the virtual sound source and the plurality of speakers. The multi-dimensional transfer function is then applied to a signal of the virtual sound source to render the virtual sound source in the sweet spot region using the plurality of speakers in the arbitrary arrangement.

Abstract: An audio output device includes: a main unit including a terminal configured to connect an external speaker; a speaker unit configured to be detachably attached to the main unit while the terminal is covered with the speaker unit; and an amplification unit configured to amplify an audio signal so that the amplified audio signal is input to the terminal and the speaker unit.

Abstract: A spatial audio processing uses a matrix of channels and conditioning amplifiers. Four channels are created from two by using a left, right, mono and a differential amp. A timing gate is added and set to the bass frequency to switch the matrix. The matrix starts with channel one receiving the left program. On the next timing cycle, channel two receives the left program, while channel one receives the left rear program. Thereafter, channel three receives the right program and channel four receives the right rear program. This condition continues through the timing order until every channel has received all four programs and then repeats itself. The constant shift in program content eliminates standing waves in any area and opens up the sound in an area that becomes spatial or orbital.

Abstract: An audio reproducing apparatus includes the following elements: an output circuit operable to generate audio signals to be supplied to first and second speakers placed near the ears of a listener; and a detector operable to detect the presence of sound output from other speakers placed at positions further away from the listener than the first and second speakers. When it is determined that there is no sound output from the other speakers on the basis of a detection result obtained by the detector, the output circuit performs a control operation so that audio signals to be supplied to the other speakers are supplied to the first and second speakers.

Abstract: A downmixer provides a listener of an output signal with a substantially accurate rendition of the apparent direction and relative loudness of an input signal. Downmixing certain channels of the input signal independently may substantially preserve the energy and intended direction of the input signal. The downmixer may include a test downmixer that operates over a limited frequency range to more accurately reflect the loudness of the input signal at the output, as perceived by a listener. The downmixer may demand fewer resources, freeing up resources for use in other operations.

Abstract: The present invention relates to a method for selecting auditory signal components for reproduction by means of one or more supplementary sound reproducing transducers, such as loudspeakers, placed between a pair of primary sound reproducing transducers, such as left and right loudspeakers in a stereophonic loudspeaker setup or adjacent loudspeakers in a surround sound loudspeaker setup, the method comprising the steps of (i) specifying azimuth angle range within which one of said supplementary sound reproducing transducers is located or is to be located and a listening direction; (Ii) based on said azimuth angle range and said listening direction, determining left and right interaural level difference limits and left and right interaural time difference limits, respectively; (iii) providing a pair of input signals for said pair of primary sound reproducing transducers; (iv) pre-processing each of said input signals, thereby providing a pair of pre-processed input signals; (v) determining interaural level dif

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of extracting a downmix signal from a bitstream; generating a decorrelated downmix signal by applying a decorrelator to the downmix signal; and generating a surround signal by applying rendering information for generating a surround signal to the downmix signal and the decorrelated downmix signal.

Abstract: A frequency domain method for phase-amplitude matrixed surround decoding of 2-channel stereo recordings and soundtracks, based on spatial analysis of 2-D or 3-D directional cues in the recording and re-synthesis of these cues for reproduction on any headphone or loudspeaker playback system.

Abstract: A multi-channel audio system is constituted using at least one line array speaker unit, in which plural speakers are arrayed in line, wherein the same audio signal is supplied with a prescribed delay time to each of the speakers, thus forming plural sound beams. The plural sound beams are reflected on a wall surface and a ceiling of a room so as to form plural virtual sound sources surrounding a listening position, and emission directions and intensities of the sound beams are controlled so as to localize a phantom at a prescribed position based on the plural virtual sound sources. By appropriately arranging plural line array speaker units horizontally, vertically, and slantingly in such a way that each line array speaker unit forms sound beams distributed and spread in a sectorial form, it is possible to realize a surround audio system having a high degree of freedom with regard to setup positions for forming virtual sound sources.

Abstract: A first matrix (W(k)) indicating frequency characteristics of a separation filter is calculated from input signals of a plurality of channels. A second matrix (Ws(k)) is calculated by using the restriction coefficients (Ci(k)) for restricting the separation filter and the first matrix, and separation filter coefficients (wsij(s)) are calculated by using the second matrix. With use of the separation filter coefficients, separation signals (ysi(t)) are then calculated from the input signals. A third matrix (Ws?1(k)) is then calculated by transforming the second matrix into an inverse matrix at each frequency, and reproduction filter coefficients (a?I1(s), a?I2(s)) are calculated by using the third matrix. With use of the reproduction filter coefficients, the synthesized signal of each channel is calculated by using the separation signals.

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.