Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.

Abstract: An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Abstract: An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.

Abstract: A downmixing device includes: a matrix conversion unit configured to perform a matrix operation for an input signal; a rotation correction unit configured to rotate an output signal of the matrix conversion unit; a spatial information extraction unit configured to extract spatial information from the output signal of the rotation correction unit; and an error calculation unit configured to calculate an error amount of the matrix operation result for the input signal by performing a matrix operation for the output signal of the rotation correction unit and the spatial information extracted by the spatial information extraction unit using a matrix that is inverse to the matrix used for the matrix operation by the matrix conversion unit.

Abstract: A method is provided for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources. The method comprises decomposing the multi-channel signal into frequency bands and the following performed per frequency band: obtaining data representative of the direction of the sound sources of the sound scene, selecting a set of sound sources constituting principal sources, adapting the data representative of the direction of the selected principal sources, as a function of restitution characteristics of the multi-channel signal, determining a matrix for mixing the principal sources as a function of the adapted data, matrixing the principal sources by the matrix determined so as to obtain a sum signal with a reduced number of channels and coding the data representative of the direction of the sound sources and forming a binary stream comprising the coded data, the binary stream being transmittable in parallel with the sum signal.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: A method for encoding a multichannel audio input signal, including steps of generating a downmix of low frequency components of a subset of channels of the input signal, waveform coding each channel of the downmix, thereby generating waveform coded, downmixed data, performing parametric encoding on at least some higher frequency components of each channel of the input signal, thereby generating parametrically coded data, and generating an encoded audio signal (e.g., an E-AC-3 encoded signal) indicative of the waveform coded, downmixed data and the parametrically coded data. Other aspects are methods for decoding such an encoded signal, and systems configured to perform any embodiment of the inventive method.

Abstract: A method of synthesizing a first (L) and a second (R) output signal from an input signal (x). The method comprises: filtering (201) the input signal to generate a filtered signal (Hx); obtaining a correlation parameter (?) indicative of a desired correlation between the first and second output signals; obtaining a level parameter (c) indicative of a desired level difference between the first and second input signals; and transforming the input signal and the filtered signal by a matrixing operation (203) into the first and second output signals, where the matrixing operation depends on the correlation parameter and the level parameter.

Abstract: A perceptually motivated, frequency domain active matrix decoder and decoding method which decodes N audio input signals to generate M audio output signals, where M is greater than N, including by generating M streams of output frequency components which determine the audio output signals, in response to N streams of input frequency components indicative of the audio input signals, determining power ratios from the input frequency components without use of feedback, including at least one power ratio for each critical frequency band in a set of critical frequency bands, and determining gain control values for each of the critical frequency bands from the power ratios including by shaping the power ratios in nonlinear fashion without use of feedback. An active matrix element is steered using the gain control values.

Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

Abstract: An audio system receives a multi-channel signal which is fed to a controller (121) that generates a first drive signal for a first sound emitter (111) by combining signals of a plurality of the channels. The first drive signal has a signal component contribution from a first bandwidth of each channel of the multi-channel signal. The multi-channel signal is also fed to another controller (115) which generates second drive signals for second sound emitters (101-109). The second drive signals are generated from a single channel signals of the multi-channel signal and in a second bandwidth having a lower cut-off frequency which is above 950 Hz for a 3 dB gain attenuation relative to an average gain for a frequency band extending 1 kHz above the lower cut-off frequency and higher than a lower cut-off frequency of the first bandwidth. A delay processor (125) introduces a delay for signal components of the first drive signal relative a corresponding second drive signal.

Abstract: This invention proposed a new algorithm. By multiply the proposed weight coefficients of this invention, CSP and CSS can be computed without computing for the mean(s) of the data. After the proposed weight coefficients of this invention undergo factorization, it can promote a new recursive and real time updatable computation method. To test the accuracy of the new invention, the StRD data were separately tested using SAS ver 9.0, SPSS ver15.0 and EXCEL 2007 for comparison. The results showed that the accuracy of the results of the proposed invention exceeds the level of accuracy of SAS ver9.0, SPSS ver15.0 and EXCEL 2007. Aside from an accurate computation, this new invented algorithm can also produce efficient computations.

Abstract: A surround sound virtualization apparatus and method. The surround sound virtualization apparatus may include an audio decoder to perform head-related transfer function (HRTF) filtering, and a time delay unit to provide a time delay to a plurality of output signals of the audio decoder.

Abstract: This invention relates to reformatting a plurality of audio input signals from a first format to a second format by applying them to a dynamically-varying transformatting matrix. In particular, this invention obtains information attributable to the direction and intensity of one or more directional signal components, calculates the transformatting matrix based on the first and second rules, and applies the audio input signals to the transformatting matrix to produce output signals.

Abstract: An encoder and encoding method for use in a surround sound system wherein at least four audio input signals representing an original sound field are encoded into two channel signals and the encoded two channel signals are decoded into at least four audio output signals corresponding to the four audio input signals. The encoder includes matrix structure connected to receive the four audio input signals for encoding the four input signals into two channel output signals. The matrix structure is responsive to the four input signals for producing L and R output signals as follows: L=FL+kFR+jRL+jkRR R=FR+kFL?jRR?jkRL wherein k denotes a transformation or matrix constant having a value approximately 0.207 and j denotes a 90 degree phase shift.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: 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 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: Biphase mark codes (BMC) are used in digital communications. Most BMC formats use preambles for rate determination and synchronization. A decoder compares the intervals of continuous high or continuous low voltages in a BMC stream to predetermined minimum and maximum values of half cell, full cell and one-and-a-half cell intervals for all supported sampling rates. If a pattern matching a preamble is found, the sampling rate is locked in and the decoder is synchronized to the BMC stream. Once locked, the decoder uses the predetermined minimum and maximum values at the locked rate to generate half cell, full cell and one-and-a-half cell indicators for a decoding state machine which decodes data in the BMC stream or decodes expected preambles.

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: 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 can be applied to the entire broadened signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter allowing the reshaping the reconstituted high-frequency part and the correction of compression artifacts 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: An apparatus for processing an audio signal and method thereof are disclosed.

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

Abstract: A multi-channel input signal having at least three original channels is represented by a parameter representation of the multi-channel signal. A first balance parameter, a first coherence parameter, or a first inter-channel time difference between a first channel pair and a second balance parameter, or a second coherence parameter, or a second inter-channel time difference parameter between a second channel pair are calculated. This set of parameters is the parameter representation of the original signals. The first channel pair has two channels, which are different from two channels of a second channel pair. Furthermore, each channel of the two channel pairs is one of the original channels, or a weighted combination of the original channels, and the first channel pair and the second channel pair include information on the three original channels.

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: 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 source mapping information corresponding to each source of multi-sources by using spatial information indicating features between the multi-sources; generating sub-rendering in formation by applying filter information giving a surround effect to the source mapping in formation per the source; generating rendering information for generating a surround signal by integrating the at least one of the sub-rendering information; and generating the surround signal by applying the rendering information to a downmix signal generated by downmixing the multi-sources.

Abstract: An apparatus and method for generating a multichannel audio signal showing a 3-dimensional (3D) effect by editing an existing multichannel audio signal. The apparatus may include an output location determiner to determine location to output an object audio signal using editing information of the object audio signal, and a multichannel object audio signal generator to generate a multichannel object audio signal by allocating the object audio signal to channels corresponding to the location to output the object audio signal.

Abstract: A sound processing system for a vehicle includes a sound processor that is configured to mix at least one real audio input signal to form at least one virtual output signal. At least one audio signal that is available to drive at least one loudspeaker may be formed using the combination of the virtual output signal and the real audio input signal. The virtual output signal may be post processed to form a predetermined frequency range of the audio signal prior to being combined with the real audio input signal. The audio signal may be created by mixing the real audio input signal with the post processed virtual output signal. Alternatively, the audio signal may be formed by mixing the real audio input signal to form a real audio output signal, and then summing the real audio output signal with the post processed virtual output signal. Mixing may be performed with a crossbar mixer included in the sound processor.

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: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: 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: The present invention includes an audio signal receiving unit receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal; an ambient component signal extracting unit extracting the ambient component signal of each of the channels based on correlation between the channel signals; an ambient component signal modifying unit modifying the ambient component signal using surround effect information; a source component signal extracting unit extracting the source component signal of each of the channels based on the correlation between the channel signals; a first signal output unit outputting the modified ambient component signal and the source component signal; and a second signal output unit outputting the audio signal or the source component signal.

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: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: A digital audio signal processor uses a matrix transform to produce a multi-channel synthetic reverberation signal based on variably-delayed versions of an input signal. The matrix transform outputs are mixed with the inputs to the signal delaying mechanisms, and some of the outputs are further processed to create the reverberation 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: 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: An electronic device for converting a multi-channel audio file to a dual channel audio file and vice versa. The multichannel audio file includes a right channel group and a left channel group of channel signals. The electronic device respectively mixes the channel signals of the right channel group and the left channel group according to a mixed matrix to form N mixed signals, and cross embeds the N mixed signals to from a left channel audio signal and a right channel audio signal to compose the dual channel audio file. The electronic device samples, recombines and decodes the left channel audio signal and the right channel audio signal according to a decoding matrix, which is the inverse of the mixed matrix, to revert to the original multi-channel audio file.

Abstract: An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Abstract: The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit that codes the first downmix signal; a first t-f converting unit that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit that generates spatial information for generating a multi-channel audio signal from a downmix signal.

Abstract: Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device includes: a decorrelated signal generation unit, and a matrix operation unit and a third arithmetic unit. The decorrelated signal generation unit generates a decorrelated signal w? indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit and the third arithmetic unit generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w? generated by the decorrelated signal generation unit, using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.

Abstract: Multiple channels of audio are combined either to a monophonic composite signal or to multiple channels of audio along with related auxiliary information from which multiple channels of audio are reconstructed, including improved downmixing of multiple audio channels to a monophonic audio signal or to multiple audio channels and improved decorrelation of multiple audio channels derived from a monophonic audio channel or from multiple audio channels. Aspects of the disclosed invention are usable in audio encoders, decoders, encode/decode systems, downmixers, upmixers, and decorrelators.

Abstract: A method and an apparatus to reproduce a 7.1 channel encoded sound through a 5.1 channel speaker system are provided. The apparatus includes a decoder to separate a 7.

Abstract: A sound source separation apparatus includes a transfer function storage unit that stores a transfer function from a sound source, a sound change detection unit that generates change state information indicating a change of the sound source on the basis of an input signal input from a sound input unit, a parameter selection unit that calculates an initial separation matrix on the basis of the change state information generated by the sound change detection unit, and a sound source separation unit that separates the sound source from the input signal input from the sound input unit using the initial separation matrix calculated by the parameter selection unit.

Abstract: An active audio matrix decoding method and apparatus to generate multi-channel audio signals from a stereo channel audio signal.

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: 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.

Abstract: Processing an audio signal associated with a sound recording made available to be rendered to an end user is disclosed. The audio signal is received. A high-order perceptual attribute of the audio signal as rendered is changed by modifying the audio signal. The modification may be based on real-time analysis of the audio signal.