Abstract: A method and system for performing sample rate conversion is provided. The method may include configuring a system to convert a sample rate of a first audio channel of a plurality of audio channels to produce a first audio stream of samples. The system may be dynamically reconfigured to convert a sample rate of a second of the plurality of audio channels to produce a second audio stream of samples, wherein the first and second audio streams are output from the system at the same time. The method may further include arbitrating between request for additional data from the first and second audio stream of samples, where processing of the first channel is suspended when the request corresponds to a second channel that is of higher priority.

Abstract: Various techniques are disclosed for reducing artifacts generated by time compression. by adapting the time compression based on the state of the received audio. The amount of time compression may be bounded based on audio characteristics. Another feature provides a way of determining the most correlated portions of segments of audio. Voiced speech may be distinguished from unvoiced speech. Another feature provides a way of distinguishing between silence, voiced speech, and unvoiced speech. Time compression may be adapted during periods of lengthy silence. Another feature allows for reducing time compression during sensitive portions of the received audio. One or more of these features may be present in different embodiments.

Abstract: For a bandwidth extension of an audio signal, in a signal spreader the audio signal is temporally spread by a spread factor greater than 1. The temporally spread audio signal is then supplied to a demicator to decimate the temporally spread version by a decimation factor matched to the spread factor. The band generated by this decimation operation is extracted and distorted, and finally combined with the audio signal to obtain a bandwidth extended audio signal. A phase vocoder in the filterbank implementation or transformation implementation may be used for signal spreading.

Abstract: Example systems and methods concern a sub-block parser that is configured with a variable sized window whose size varies as a function of the actual or expected entropy of data to be parsed by the sub-block parser. Example systems and methods also concern a sub-block parser configured to compress a data sequence to be parsed before parsing the data sequence. One example method facilitates either actually changing the window size or effectively changing the window size by manipulating the data before it is parsed. The example method includes selectively reconfiguring a data set to be parsed by a data-dependent parser based, at least in part, on the entropy level of the data set, selectively reconfiguring the data-dependent parser, based, at least in part, on the entropy level of the data set, and parsing the data set.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: The present disclosure is directed towards methods and systems for virtualizing audio hardware for one or more virtual machines. A control virtual machine (VM) may translate a first stream of audio functions calls from a first VM hosted by a hypervisor. The translated first stream of audio function calls may be destined for a sound card of the computing device executing the hypervisor. The control VM may detect a second stream of audio functions calls from a second VM hosted by the hypervisor. The control VM may translate the second stream of audio functions calls from the second VM. The control VM may further merge the translated first stream of audio function calls and the translated second stream of the audio function calls in response to the detected second stream. The control VM may transmit the merged stream of audio function calls to the sound card.

Abstract: A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided.

Abstract: Systems (1600) and methods (1500) for frame synchronization. The methods involve: extracting bit sequences S0 and S1 from a Bit Stream (“BS”) of a Data Burst (“DB”); decoding S0 and S1 to obtain decoded bit sequences S?0 and S?1; using S?0 and S?1 to determine Bit Error Rate (“BER”) estimates (516, 518); combining the BER estimates to obtain a combined BER estimate; modifying S0 and S1 so that each includes at least one bit of BS which is not included in its current set of bits and so that it is absent of at least one of the bits in the current set of bits; iteratively repeating the decoding, using, combining and modifying steps to obtain more combined BER estimates; analyzing the combined BER estimates to identify a minimum combined BER estimate; and using the minimum combined BER estimate to determine a location of a vocoder voice frame within DB.

Abstract: A method comprising receiving at a user equipment encrypted content. The content is stored in said user equipment in an encrypted form. At least one key for decryption of said stored encrypted content is stored in the user equipment.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: An audio decoder includes an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically encoded representation thereof, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code representing a spectral value, or a most significant bit-plane thereof, in a decoded form, in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. It evaluates a hash table, entries of which define both significant state values amongst the numeric context values and boundaries of intervals of numeric context values, in order to select the mapping rule, wherein the hash table ari_hash_m is defined as given in FIGS. 22(1), 22(2), 22(3) and 22(4).

Abstract: A voice-coded in-band communication device monitors a voice-coded channel to detect data to present to a user. During operation, the communication device can detect a data-encoding signal from the voice-coded channel, such that the voice-coded channel can carry an audio signal that includes a voice signal and the data-encoding signal. The device decodes the data-encoding signal to detect a data element. The data element can include information that is to be presented to a local user, a request from a remote device for information about the local user, or information that the system can use to establish a peer-to-peer connection with the remote device over a separate data channel. The device can also generate a filtered audio signal to present to the user by removing the detected data-encoding signal from the voice-coded channel, and then reproduces the filtered audio signal for the user.

Abstract: A time-domain system and method of modifying the time scale of digital audio signals includes a pre-processor. The pre-processor forms a synthesized signal for processing with minimum computation and that has optional features to give preference to certain audio channels and/or frequency bands, a mechanism of adaptively characterizing the temporal features of the synthesized signal by its normalized power and zero-crossing count, and a mechanism of identifying a segment of the synthesized signal where the time scale can be modified without introducing artifacts or losing content.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

Abstract: A computing device identifies a first codeword in a first codebook to represent short-timescale information of frames in a time-based data item segmented at intervals and identifies a second codeword in a second codebook to represent long-timescale information of the frames. The computing device generates a third codebook based on the first codeword and the second codeword for the frames to add long-timescale information context to the short-timescale information of the frames.

Abstract: In one embodiment, a method provides for monitoring and analyzing communications of a monitored user on behalf of a monitoring user, to determine whether the communication includes a violation. For example, SMS messages, MMS messages, IMs, e-mails, social network site postings or voice mails of a child may be monitored on behalf of a parent. In one embodiment, an algorithm is used to analyze a normalized version of the communication, which algorithm is retrained using results of past analysis, to determine a probability of a communication including a violation.

Abstract: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state 715 located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation 745 of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: Downmixing multi-channel audio signals to target channels by pre-downmixing frequency coefficients that are encoded using a most frequently used block type in stereo channels in the frequency domain, thereby reducing an amount of calculations and an amount of power required to downmix the multi-channel audio signals.

Abstract: Adaptive Gain Control (AGC) is performed directly in a coded domain. A Coded Domain Adaptive Gain Control (CD-AGC) system modifies at least one parameter of a first encoded signal, resulting in corresponding modified parameter(s). The CD-VQE system replaces the parameter(s) of the first encoded signal with the modified parameter(s), resulting in a second encoded signal. In a decoded state, the second encoded signal approximates a target signal that is a function of two signals, including the first encoded signal and a third encoded signal, in at least a partially decoded states. Thus, the first encoded signal does not have to go through intermediate decode/re-encode processes, which can degrade overall speech quality. Computational resources required for a complete re-encoding are not needed. Overall delay of the system is minimized. The CD-AGC system can be used in any network in which signals are communicated in a coded domain, such as a Third Generation (3G) wireless network.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

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 coding apparatus includes a generation unit configured to generate first coding information used for first coding of a first audio signal and second coding information used for second coding of a second audio signal, and generate third coding information used for the first coding of the second audio signal and fourth coding information used for the second coding of a third audio signal; a first coding unit configured to generate first data and second data; a second coding unit configured to generate third data and fourth data by performing the second coding on the third audio signal; and a multiplexing unit configured to generate a stream of the first audio signal and a stream of the second audio signal. The third data is decoded in place of the second data in a case where a loss or an error has occurred in the stream of the second audio signal.

Abstract: A method, a device and a system for voice encoding/decoding are disclosed in the present invention. The method includes: assembling an input pulse code modulation signal into one signal according to a designated time slot and assembly manner; and encoding the assembled signal according to a designated encoding manner to output an encoded voice signal. In the present invention, because a process of assembling or splitting the signal may be implemented through software, in the case that hardware in a current network does not need to be replaced, an effect of encoding/decoding voice with a 7 K spectrum may be achieved in the current network.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: Encoding and decoding of residual signals are provided. In a method of encoding a residual signal of an audio signal, the residual signal is divided into a plurality of sections having different sizes, based on a change of the residual signal. Then, section division information representing information about the divided sections and section-by-section residual signal information representing characteristics of the sections of the residual signal are acquired. Thereafter, the residual signal is encoded based on the section division information and the section-by-section residual signal information.

Abstract: A representation of an audio signal having a first, a second and a third frame is derived by estimating first warp information for the first and second frames and second warp information for the second and third frames, the warp information describing pitch information of the audio signal. First or second spectral coefficients for first and second frames or second and third frames are derived using first or second warp information and a first or second weighted representation of the first and second frames or second and third frames, the first or second weighted representation derived by applying a first or second window function to the first and second frames or second and third frames, wherein the first or second window function depends on the first or second warp information. The representation of the audio signal is generated including the first and the second spectral coefficients.

Abstract: An audio encoding device includes a time-frequency converting unit that conducts time-frequency conversion of channel signals included in an audio signal having a plurality of channels in frame units having a certain length of time to convert the channel signals to respective frequency signals; a downmixing unit that generates a main signal representing a major component of a first channel and a second channel among the plurality of channels, and a residual signal that is a component orthogonal to the main signal; a weight determining unit that obtains a decoding value predicted and a decoding value predicted, obtains signal components affecting each other between the first channel and the second channel; a weighting unit that uses the weighting coefficient; a residual signal encoding unit that encodes the weighted residual signal the weighting coefficient; and a main signal encoding unit that encodes the main signal.

Abstract: The present disclosure relates to coding and decoding technologies, and discloses a preprocessing method, a preprocessing apparatus, and a coding device. The preprocessing method includes: obtaining characteristic information of a current frame signal; identifying whether the current frame signal requires no coding operation of removing LTC according to the characteristic information of the current frame signal and preset information; and if identifying that the current frame signal requires no coding operation of removing LTC, performing the coding operation of removing STC for the current frame signal; and if identifying that the current frame signal requires the coding operation of removing LTC, performing the coding operations of removing both LTC and STC for the current frame signal. Through the technical solution provided herein, the coding operation of removing LTC is performed for only part of the input frame signals.

Abstract: A computation apparatus includes: a range calculation section for calculating a range of an input value that can give a predetermined discrete value obtained by discretizing a computation result of a nonlinear operation; and a discrete value output section for outputting, when the input value is input, the predetermined discrete value corresponding to the range in which the input value that has been input is contained.

Abstract: Codebook Arrangement for use in coding an input sound signal includes First and Second Codebook Stages. First Codebook Stage includes one of a time-domain CELP codebook and a transform-domain codebook. Second Codebook Stage follows the first codebook stage and includes the other of the time-domain CELP codebook and the transform-domain codebook. Codebook Stage includes an adaptive codebook may be provided before First Codebook Stage. A selector may be provided to select an order of the time-domain CELP codebook and the transform-domain codebook in First and Second Codebook Stages, respectively, as a function of characteristics of the input sound signal. The selector may also be responsive to both the characteristics of the input sound signal and a bit rate of the codec using Codebook Arrangement to bypass Second Codebook Stage. Codebook Arrangement can be used in a coder of an input sound signal.

Abstract: [PROBLEMS] To provide a high-quality audio signal encoding technique by controlling the number of time/frequency groups in a frame. [MEANS FOR SOLVING PROBLEMS] An audio encoding device includes: a time group boundary candidate position extraction unit (101) for analyzing a sub-band signal (2001) obtained by frequency-changing an input signal and calculating a candidate position of the time group boundary by comparing change in energy of three successive time groups; a time group quantity generation unit (103) for outputting a maximum value of the time group quantity; a time group selection unit (102) for generating a time group quantity not greater than the maximum time group quantity by using the candidate position; and a frequency group generation unit (104) for generating a frequency group by using the generated time group information.

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: A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.

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: Aspects of a method and system for an asynchronous pipeline architecture for multiple independent dual/stereo channel PCM processing are provided. Asynchronously pipeline processing of audio information comprised within a decoded PCM frame may be based on metadata information generated from the decoded PCM frame and an output decoding rate. The asynchronously pipeline processing may comprise mixing a primary audio information portion and a secondary audio information, portion, sample rate converting the audio information, and buffering the audio information. The asynchronously pipeline processing may comprise multiple pipeline stages. Feeding back an output of one of the pipeline stages to an input of a previous one of the pipeline stages may be enabled. The metadata information may comprise a frame start indicator associated with the decoded PCM frame and/or a plurality of mixing coefficients.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: The present invention is directed to system(s), method(s), and apparatus for accurate fast forward rate when performing trick play with variable distance between frames. In one embodiment, there is presented a circuit for providing a fast forward video sequence. The circuit comprises a system time clock for providing a time reference, said time reference incremented at a predetermined fast forward rate; a comparator for comparing the time reference with timing information associated with a picture; and a controller for determining whether to display the picture based at least in part on the comparison between the timing information and the time reference.

Abstract: A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit.

Abstract: An apparatus for calculating bandwidth extension data of an audio signal in a bandwidth extension system, in which a first spectral band is encoded with a first number of bits and a second spectral band different from the first spectral band is encoded with a second number of bits, the second number of bits being smaller than the first number of bits, has a controllable bandwidth extension parameter calculator for calculating bandwidth extension parameters for the second frequency band in a frame-wise manner for a sequence of frames of the audio signal. Each frame has a controllable start time instant. The apparatus additionally includes a spectral tilt detector for detecting a spectral tilt in a time portion of the audio signal and for signaling the start time instant for the individual frames of the audio signal depending on spectral tilt.

Abstract: An uplink or downlink audio processor contains a multi band compressor that receives an input, uplink or downlink, audio signal. The multi-band compressor has a band splitter that splits the input audio signal into a number of different band signals. Each band signal is input to a respective compressor block, which is independently programmable so that its audio frequency response (a) differs from a linear response in at least two non-overlapping windows of its input signal, and (b) differs from the frequency response of another one of the compressor blocks. Other embodiments are also described and claimed.

Abstract: An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal.

Abstract: Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider.

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: This invention relates to indexing an input vector contained in a set of vectors of a plurality of sets of vectors. The indexing comprises performing, in case that the input vector is contained in a set of vectors of a pre-defined group of one or more sets of vectors of the plurality of sets of vectors, a specific processing that is adapted to a characteristic of the sets of vectors in the pre-defined group of sets of vectors and is only applicable in case of input vectors contained in sets of vectors with the characteristic. The indexing further comprises performing, in case that the input vector is not contained in a set of vectors of the pre-defined group of sets of vectors, a general processing. The invention further relates to an according determining of a target vector contained in a set of vectors of a plurality of sets of vectors based on an index associated with said target vector.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a window-based signal transformer configured to map a time-frequency representation, which is described by the encoded audio information, to a time-domain representation. The window-based signal transformer is configured to select a window, out of a plurality of windows including windows of different transition slopes and windows of different transform length, on the basis of a window information. The audio decoder includes a window selector configured to evaluate a variable-codeword-length window information in order to select a window for a processing of a given portion of the time-frequency representation associated with a given frame of the audio information.

Abstract: A particular method includes loading one or more memory images into a multi-way cache. The memory images are associated with an audio decoder, and the multi-way cache is accessible to a processor. Each of the memory images is sized not to exceed a page size of the multi-way cache.