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: In a system and method for reproducing a sound field the orientation of a computing device may be detected. Several orientation indications may be used to detect the computing device orientation. The detected orientation may be relative to a sound field that is a spatial representation of an audible environment associated with the computing device. Audio transducers associated with the computing device may be selected in order to reproduce the sound field based on the detected orientation. A received encoding may be processed and reproduced with the selected audio transducers.

Abstract: The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. The encoding device stores speaker arrangement information in a comment region in a PCE of an encoded bit stream and stores a synchronous word and identification information in the comment region such that other public comments and the speaker arrangement information stored in the comment region can be distinguished from each other. When an encoded bit stream is decoded, it is determined whether the speaker arrangement information is stored on the basis of the synchronous word and the identification information stored in the comment region. Audio data included in the encoded bit stream is output according to the arrangement of the speakers corresponding to the determination result. The present technique can be applied to an encoding device.

Abstract: In a system and method for encoding a sound field the orientation of a computing device may be detected. Several orientation indications may be used to detect the computing device orientation. The detected orientation may be relative to a sound field that is a spatial representation of an audible environment associated with the computing device. Microphones associated with the computing device may be selected in order to receive the sound field based on the detected orientation. The received sound field may be processed and encoded with associated descriptive information.

Abstract: The present technology relates to an encoding device and method, a decoding device and method, and a program that enable audio of a high audio quality to be obtained with a smaller code amount. The encoding device multiplexes low frequency encoding data obtained by encoding a low frequency component of an input signal and high frequency encoding data obtained by encoding data including an estimation coefficient to acquire a high frequency component of the input signal by estimation and outputs multiplexed data. When the input signal is encoded, a calculation unit calculates pseudo high frequency subband power to be an estimation value of power of the high frequency component from an estimation coefficient selected in a frame immediately before a frame of a processing target and the high frequency component of the input signal.

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

Abstract: An encoder is disclosed for use in a surround sound system wherein at least four audio input signals (FL, FR, RL, RR) representing an original sound field are encoded into two channel signals (L, and the encoded two channel signals are decoded into at least four audio output signals (FL?, FR?, RL?, RR?) corresponding to the four audio input signals. The encoder includes matrix means connected to receive the four audio input signals for encoding the four input signals into two channel (L and R) output signals. The matrix means includes means responsive to the four input signals for producing Lenc and Renc output signals as follows: Lenc=FL+kFR+jRL+jkRR Renc=FR+kFL?jRR?jkRL wherein k denotes a transformation or matrix coefficient having a value that is steered dynamically based on level of rear signal (RL+RR) content relative to front signal (FL+FR) content. An encoding method for use in a surround sound system is also disclosed.

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: A method of managing a power state of devices in a home entertainment system using a wireless home entertainment hub comprises obtaining a registration of a plurality of devices, where the plurality of devices includes at least one sink device and at least one source device. A selection of one of the source devices is received for presentation of program content to a user in the home entertainment system. A subset of the at least one sink device is determined for presentation of the program content to the user based on the selection of the source device. Each sink device in the subset of the at least one sink device is ensured to be active.

Abstract: A method for multi-channel processing in a multi-channel sound system, in which a channel or a channel mixture is first split into individual channels, the individual channels are limited by setting the values of the parameters channel fader, threshold, release, and output level and then encoding the individual channels. At least two channels are compressed and/or limited with a uniform output level value in method step, one channel is provided with a deviating output level value, which is set, depending on the audio material to be processed, and every further channel is compressed and/or limited in such a manner that it has an output level value that is at least one decibel less than the uniform output level value. The individual channels are combined into an encoded (coded) channel by setting a value of at least one of the parameters channel fader, threshold, release, and output level.

Abstract: The invention relates to an apparatus (1) for improving a stereo audio signal of an FM stereo radio receiver (10). It is suggested to employ components of a standard HE-ACC v2 decoder (40) in such apparatus (1) to efficiently recover the stereo audio signal (L\R?) while also exhibiting good signal quality. To that end, parametric stereo parameters (15) shall be quantized and submitted to such decoder (40) together with a mono-downmix signal (25). The decoder (25) can then easily recover the parametric stereo parameters (15) by de-quantizing them via a simple lookup table, and finally creating the stereo output signal (L?R?) from the de-quantized parametric stereo parameters and the mono-downmix signal (25). The apparatus (1) further includes a parameter estimation stage (5) which may again be embodied on the basis of components of a standard HE-ACC v2 encoder (65).

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

Abstract: A decoding method, medium, and device decoding an input signal, including compressed multi-channel signals as a mono or stereo signal, into 2-channel binaural signals. A full band channel level of each channel in the multi-channel system is calculated from channel level differences between the channels, and data of each channel included in the input signal is localized in directions corresponding to the channels based on the calculated full band channel levels of the channels. Accordingly, the input signal can be output as the 2-channel binaural signals by using simple operations without having to reconstruct multi-channel signals from the input signal in a quadrature mirror filter (QMF) domain.

Abstract: An input signal conveying encoded information representing one or more audio channels is decoded by determining the configuration of channels represented by the encoded information, obtaining from the channel configuration a channel selection mask that specifies which of the one or more audio channels are to be decoded, extracting encoded information from the input signal, and decoding the extracted encoded information for those audio channels specified in the channel selection mask.

Abstract: The invention relates to a method for processing a multi-channel audio signal which carries a plurality of audio channel signals. The method comprises determining a time-scaling position using the plurality of audio channel signals and time-scaling each audio channel signal of the plurality of audio channel signals according to the time-scaling position to obtain a plurality of time scaled audio channel signals.

Abstract: For flexibly signaling a synchronous mode or an asynchronous mode in the multi-channel parameter reconstruction, a parameter configuration cue is inserted in the data stream, which is used by a configurator on the side of a multi-channel decoder to configure a multi-channel reconstructor. If the parameter configuration cue has a first meaning, the configurator will look for further configuration information in its input data, while, when the parameter configuration cue has another meaning, the configurator performs a configuration setting of the multi-channel reconstructor based on information on a coding algorithm with which transmission channel data have been coded, so that it is ensured efficiently on the one hand and flexibly on the other hand that there will always be obtained a correct association between parameter data and decoded transmission channel data.

Abstract: A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.

Abstract: A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.

Abstract: An apparatus for merging a first spatial audio stream with a second spatial audio stream to obtain a merged audio stream comprising an estimator for estimating a first wave representation comprising a first wave direction measure and a first wave field measure for the first spatial audio stream, the first spatial audio stream having a first audio representation and a first direction of arrival. The estimator being adapted for estimating a second wave representation comprising a second wave direction measure and a second wave field measure for the second spatial audio stream, the second spatial audio stream having a second audio representation and a second direction of arrival.

Abstract: A method and apparatus for processing an audio signal is disclosed. Herein, the method includes receiving a downmix information having at least two independent objects and a background object downmixed therein; separating the downmix information into a first independent object and a temporary background object using a first enhanced object information; and extracting a second independent object from the temporary background object using a second enhanced object information.

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: A method for parametric spatial audio coding of a multi-channel audio signal comprising a plurality of audio channel signals is provided, the method comprising: calculating at least two different spatial coding parameters for an audio channel signal of the plurality of audio channel signals, selecting at least one spatial coding parameter of the at least two different spatial coding parameters associated with the audio channel signal on the basis of the values of the calculated spatial coding parameters; including a quantized representation of the selected spatial coding parameter into a parameter section of an audio bitstream; and setting a parameter type flag in the parameter section of the audio bitstream indicating the type of the selected spatial coding parameter being included into the audio bitstream.

Abstract: Provided are methods, systems, and apparatus for hierarchical decorrelation of multichannel audio. A hierarchical decorrelation algorithm is designed to adapt to possibly changing characteristics of an input signal, and also preserves the energy of the original signal. The algorithm is invertible in that the original signal can be retrieved if needed. Furthermore, the proposed algorithm decomposes the decorrelation process into multiple low-complexity steps. The contribution of these steps is generally in a decreasing order, and thus the complexity of the algorithm can be scaled.

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: The invention relates to a parametric audio encoder, comprising a parameter generator, the parameter generator being configured to determine a first set of encoding parameters and reference audio signal values, wherein the reference audio signal is another audio channel signal or a downmix audio signal derived from at least two audio channel signals of the plurality of multi-channel audio signals, to determine a first encoding parameter average based on the first set of encoding parameters of the audio channel signal, to determine a second encoding parameter average based on the first encoding parameter average of the audio channel signal and at least one other first encoding parameter average of the audio channel signal, and to determine the encoding parameter based on the first encoding parameter average of the audio channel signal and the second encoding parameter average of the audio channel signal.

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 representation of a multi-channel signal having several original channels includes a parameter set, which, when used together with at least one down-mix channel allows a multi-channel reconstruction. An additional level parameter is calculated such that an energy of the at least one downmix channel weighted by the level parameter is equal to a sum of energies of the original channels. The additional level parameter is transmitted to a multi-channel reconstructor together with the parameter set or together with a down-mix channel. An apparatus for generating a multi-channel representation uses the level parameter to correct the energy of the at least one transmitted downmix channel before entering the down-mix signal into an upmixer or within the up-mixing process.

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: Systems, methods, and apparatus for a unified approach to encoding different types of audio inputs are described.

Abstract: A device (100) for processing data, the device (100) comprising a detection unit (110) adapted for detecting individual reproduction modes indicative of a manner of reproducing the data separately for each of a plurality of human users, and a processing unit (120) adapted for processing the data to thereby generate reproducible data separately for each of the plurality of human users in accordance with the detected individual reproduction modes.

Abstract: Systems and methods for producing multi-dimensional parametric audio are provided.

Abstract: A data embedding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, extracting, from a code book storing a plurality of prediction coefficients, a plurality of candidates of prediction coefficients of a channel signal out of a plurality of channel signals, each candidate having a predictive error falling within a specific range of predictive error of predictive encoding of two other channel signals; and selecting from the extracted candidates a prediction coefficient as a result of the predictive encoding in accordance with a specific data embedding rule and embedding embed target data into the selected prediction coefficient.

Abstract: Method, apparatus and computer program product for processing an input stereophonic audio signal to thereby generate a converted stereophonic audio signal representing the input stereophonic audio signal, the input stereophonic audio signal comprising a left input audio signal and a right input audio signal, and the converted stereophonic audio signal comprising a first converted audio signal and a second converted audio signal. The first converted audio signal is generated based on the sum of the left input audio signal and the right input audio signal. The second converted audio signal is generated based on the difference between a first function of the left input audio signal and a second function of the right input audio signal. The first and second functions are adjustable to thereby adjust at least one characteristic of the converted stereophonic audio signal.

Abstract: Methods, systems and computer-readable medium reduce and/or eliminate errors in coding/decoding of streamed audio due to resetting of decoder state values based on playback buffer access. An inaudible compensation signal is included with the audio signal. The compensation signal is generated having a characteristic selected so that the encoded streamed audio signal substantially matches the reset state values at block boundaries. In an ADPCM example, the compensation signal is chosen such that the sum of the compensation signal and the original audio signal (=the compensated audio signal) has the characteristic that, at the block boundaries, the compensated audio signal matches the initial predictor value.

Abstract: A selected channel of a multi-channel signal which is represented by frames composed from sampling values having a high time resolution can be encoded with higher quality when a wave form parameter representation representing a wave form of an intermediate resolution representation of the selected channel is derived, the wave form parameter representation including a sequence of intermediate wave form parameters having a time resolution lower than the high time resolution of the sampling values and higher than a time resolution defined by a frame repetition rate. The wave form parameter representation with the intermediate resolution can be used to shape a reconstructed channel to retrieve a channel having a signal envelope close to that one of the selected original channel. The time scale on which the shaping is performed is shorter than the time scale of a framewise processing, thus enhancing the quality of the reconstructed channel.

Abstract: An audio decoder (100) comprising: effect means, decoding means, and rendering means. The effect means (500) generate modified down-mix audio signals from received down-mix audio signals. Said received down-mix audio signals comprise a down-mix of a plurality of audio objects. Said modified down-mix audio signals are obtained by applying effects to estimated audio signals corresponding to audio objects comprised in said received down-mix audio signals. Said estimated audio signals are derived from the received down-mix audio signals based on received parametric data. Said received parametric data comprise a plurality of object parameters for each of the plurality of audio objects. Said modified down-mix audio signals based on a type of the applied effect are decoded by decoding means or rendered by rendering means or combined with the output of rendering means.

Abstract: Frame elements which shall be made available for skipping may are transmitted more efficiently by arranging that a default payload length information is transmitted separately within a configuration block, with the length information within the frame elements, in turn, being subdivided into a default payload length flag followed, if the default payload length flag is not set, by a payload length value explicitly coding the payload length of the respective frame element. However, if the default payload length flag is set, an explicit transmission of the payload length may be avoided. Rather, any frame element, the default extension payload length flag of which is set, has the default payload length and any frame element, the default extension payload length flag of which is not set, has a payload length corresponding to the payload length value. By this measure, transmission effectiveness is increased.

Abstract: Systems, methods, and apparatus for a unified approach to encoding different types of audio inputs are described.

Abstract: An audio decoder for decoding an encoded audio signal, the encoded audio signal including a first channel element and a second channel element in a payload section of a data stream and first decoder configuration data for the first channel element and second decoder configuration data for the second channel element in a configuration section of the data stream, includes: a data stream reader for reading the configuration data for each channel element in the configuration section and for reading the payload data for each channel element in the payload section; a configurable decoder for decoding the plurality of channel elements; and a configuration controller for configuring the configurable decoder so that the configurable decoder is configured in accordance with the first decoder configuration data when decoding the first channel element and in accordance with the second decoder configuration data when decoding the second channel element.

Abstract: A method for reducing howling in a communication system containing collocated mobile devices is presented. In a transmitter, an audio signal is received at a microphone. Acoustic feedback is removed from the audio signal and the resulting signal is encoded and transmitted either using direct or trunked mode operation to a receiver. The encoded signal is decoded at the transmitter, in addition to at the receiver, and fed back to an echo canceller with sufficient delay to account for substantially the entirety of a loop delay from encoding of the audio signal to reception of the acoustic feedback at the microphone to enable removal of the acoustic feedback. An estimate of the acoustic feedback is used to initially remove the acoustic feedback, the error being fed back to the processor to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

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: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a down-mix signal and spatial information regarding a plurality of channels from an input bitstream, and generating a three-dimensional (3D) down-mix signal by performing a 3D rendering operation on the down-mix signal using the spatial information and a filter, wherein the sum of the number of valid signals of the down-mix signal, the number of valid signals of the spatial information, and the number of valid signals of co-efficients of the filter is less than the number of valid signals of the 3D down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: Two digital data sets are combined by equating a first subset of samples to neighboring samples from a second subset which is interleaved with the first subset of samples where the equated samples of the two digital data sets do not correspond in time, and by subsequently adding corresponding samples from both digital data sets. This results in a third digital data set that allows the unraveling of the two digital data sets. The third digital data set, when combining two digital audio streams into a single digital audio stream, is still a good mono representation of the two combined digital audio streams and can thus be reproduced on regular reproduction equipment, yet the use of a decoder according to the invention allows the unraveling of the two digital data sets from the third digital data set.

Abstract: Embodiments of the present invention disclose an audio decoding method, including: determining that bitstreams to be decoded are monophony coding layer and first stereo enhancement layer bitstreams; decoding the monophony coding layer to obtain a monophony decoded frequency-domain signal; reconstructing left and right channel frequency-domain signals in a first sub-band region by utilizing the monophony decoded frequency-domain signal after an energy adjustment; and reconstructing left and right channel frequency-domain signals in a second sub-band region by utilizing the monophony decoded frequency-domain signal without the energy adjustment.

Abstract: An apparatus for processing an audio signal and method thereof, comprising receiving a downmix signal comprising at least one normal object signal, and bitstream including object information determined when the downmix signal is generated; extracting extension type identifier indicating whether the downmix signal further comprises a multi-channel object signal, from extension part of the bitstream; when the extension type identifier indicates that the downmix signal further comprise multi-channel object signal, extracting first spatial information from the bitstream; and, transmitting at least one of the first spatial information and the second spatial information; wherein the first spatial information is determined when a multi-channel source signal are downmixed into the multi-channel object signal, wherein the second information is generated using the object information and mix information, are disclosed.

Abstract: A decoded sound analysis unit (104) calculates, regarding the frequency-region stereo signals L(b) and R(b) decoded by the PS decoding unit (103), a second degree of similarity (109) and a second intensity difference (110) from the decoded sound signals. A spectrum correction unit (105) detects a distortion added by the parametric-stereo conversion by comparing the second degree of similarity (109) and the second intensity difference (110) calculated at the decoding side with the first degree of similarity (107) and the first intensity difference (108) calculated and transmitted from the encoding side, and corrects the spectrum of the frequency-region stereo decoded signals L(b) and R(b).

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes an audio signal receiving unit receiving the audio signal including at least one object; a preset metadata receiving unit receiving preset metadata from preset information; a preset rendering data receiving unit obtaining preset matrix from the preset information; a display unit displaying the preset metadata; an input unit receiving command for selecting one of the preset metadata; and an object adjusting unit adjusting output level of the object according to the output channel by using the preset matrix corresponding to the selected preset metadata. Accordingly, without user's setting for each object, if preset metadata to be applied to an audio signal is selected with reference to preset metadata and preset matrix, levels of objects included in the audio signal can be easily adjusted using preset rendering data corresponding to the selected preset metadata.

Abstract: There is provided a method of encoding audio and including said encoded audio into a digital transport stream, comprising receiving at an encoder input a plurality of temporally co-located audio signals, assigning identical time stamps per unit time to all of the plurality of temporally co-located audio signals and incorporating the identically time stamped audio signals into the digital transport stream. There is also provided a method decoding said encoded data, and encoding apparatus and decoding apparatus.

Abstract: The invention concerns a method and a system for sound spatialization of a first set of not less than one of the audio channels encoded on of a number of frequency subbands (SBk) and decoded in a transformed domain (Fl, C, Fr, Sr, SI, Ife) into a second set of not less than two (Bl, Br) sound channels in the time domain, from modelling filters converted into a gain and a delay applicable in the transformed domain involving: filtering (A) through equalization, subband delay of the signal by applying at least one gain and one delay to generate from each of said encoded channels an equalized and delayed component; adding (B) a subset of equalized and delayed signals to create a number of filtered signals corresponding to not less than two; synthesizing (C) each of said filtered signals to obtain the second set of not less than two reproduction sound channels (Bl, Br) in the time domain.