Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: The basic concept of the present invention is to extrapolate a partially known spatial covariance matrix of a multi-channel signal in the parameter domain. The extrapolated covariance matrix is used with the downcoded downmix signal in order to efficiently generate an estimate of a linear combination of the multi-channel signals.

Abstract: A network processing node (e.g., MGW, MRFP) and method are described herein that can: (1) receive packets on a first heterogeneous link (e.g., wireless link); (2) manipulate the received packets based on known characteristics about a second heterogeneous link (e.g., “Internet” link); and (3) send the manipulated packets on the second heterogeneous link (e.g., “Internet” link). For example, the network processing node can manipulate the received packets by adding redundancy, removing redundancy, frame aggregating (re-packetizing), recovering lost packets and/or re-transmitting packets.

Abstract: A method for spectrum recovery in spectral decoding of an audio signal, comprises obtaining (210) of an initial set of spectral coefficients representing the audio signal, and determining (212) a transition frequency. The transition frequency is adapted to a spectral content of the audio signal. Spectral holes in the initial set of spectral coefficients below the transition frequency are noise filled (214) and the initial set of spectral coefficients are bandwidth extended (216) above the transition frequency. Decoders and encoders being arranged for performing part of or the entire method are also illustrated.

Abstract: Information about excitation signals of a first signal encoded by CELP is used to derive a limited set of candidate excitation signals for a second correlated second signal. Preferably, pulse locations of the excitation signals of the first encoded signal are used for determining the set of candidate excitation signals. More preferably, the pulse locations of the set of candidate excitation signals are positioned in the vicinity of the pulse locations of the excitation signals of the first encoded signal. The first and second signals may be multi-channel signals of a common speech or audio signal. However, the first and second signals may also be identical, whereby the coding of the second signal can be utilized for re-encoding at a lower bit rate.

Abstract: A first signal representation of one or more of the multiple channels is encoded in a first encoding process, and a second signal representation of one or more of the multiple channels is encoded in a second, filter-based encoding process. Filter smoothing can be used to reduce the effects of coding artifacts. However, conventional filter smoothing generally leads to a rather large performance reduction and is therefore not widely used. It has been recognized that coding artifacts are perceived as more annoying than temporary reduction in stereo width, and that they are especially annoying when the coding filter provides a poor estimate of the target signal; the poorer the estimate, the more disturbing artifacts. Therefore, signal-adaptive filter smoothing is introduced in the second encoding process or a corresponding decoding process.

Abstract: A transient detector (100) analyzes (110) a given frame n of the input audio signal to determine, based on audio signal characteristics of the given frame n, a transient hangover indicator for a following frame n+1, and signals (120) the determined transient hangover indicator to an associated audio encoder (10) to enable proper encoding of the following frame n+1.

Abstract: In a method of perceptual transform coding of audio signals in a telecommunication system, performing the steps of determining transform coefficients representative of a time to frequency transformation of a time segmented input audio signal; determining a spectrum of perceptual sub-bands for said input audio signal based on said determined transform coefficients; determining masking thresholds for each said sub-band based on said determined spectrum; computing scale factors for each said sub-band based on said determined masking thresholds, and finally adapting said computed scale factors for each said sub-band to prevent energy loss for perceptually relevant sub-bands.

Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: The invention provides an efficient technique for encoding a multi-channel audio signal. The invention relies on the principle of encoding (S1) a signal representation of one or more of the multiple channels in a first encoding process, and encoding another signal representation of one or more channels in a second, filter-based encoding process. A basic idea according to the invention is to select (S2), for the second encoding process, a combination of i) frame division configuration of an overall encoding frame into a set of sub-frames, and ii) filter length for each sub-frame, according to a predetermined criterion. The second signal representation is then encoded (S3) in each sub-frame of the overall encoding frame according to the selected combination. The possibility to select frame division configuration and at the same time adjust the filter length for each sub-frame provides added degrees of freedom, and generally results in improved performance.

Abstract: Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.

Abstract: The signal processing is based on the concept of using a time-domain aliased (12, TDA) frame as a basis for time segmentation (14) and spectral analysis (16), performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall ?segmented? time-to-frequency transform can thus be changed by simply adapting the time segmentation to obtain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

Abstract: A method for perceptual spectral decoding comprises decoding of spectral coefficients recovered from a binary flux into decoded spectral coefficients of an initial set of spectral coefficients. The initial set of spectral coefficients are spectrum filled. The spectrum filling comprises noise filling of spectral holes by setting spectral coefficients in the initial set of spectral coefficients not being decoded from the binary flux equal to elements derived from the decoded spectral coefficients. The set of reconstructed spectral coefficients of a frequency domain formed by the spectrum filling is converted into an audio signal of a time domain. A perceptual spectral decoder comprises a noise filler, operating according to the method for perceptual spectral decoding.

Abstract: Signals of different channels are combined into one mono signal. A set of adaptive filters, preferably one for each channel, is derived in a respective filter adaptation unit. When an adaptive filter is applied to the mono signal it reconstructs the signal of the respective channel under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal, in encoded form, and the parameters of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.

Abstract: A vector quantizer includes a lattice quantizer (10) approximating a vector x by a lattice vector belonging to a lattice ?0. A lattice vector decomposer (14) connected to the lattice quantizer successively decomposes the lattice vector into a sequence of quotient vectors y, and a sequence of remainder vectors ri on successive lattices ?l?1, by lattice division with a corresponding predetermined sequence of integers pi?2, where i=1 . . . k and k is a positive integer representing the number of elements in each sequence.

Abstract: In signal processing using digital filtering the representation of a filter is adapted depending on the filter characteristics. If e.g. a digital filter is represented by filter coefficients for transform bands Nos. 0, 1, . . . , K in the frequency domain, a reduced digital filter having coefficients for combined transform bands, i.e. subsets of the transformed bands, Nos. 0, 1, . . . , L, is formed and only these coefficients are stored. When the actual filtering in the digital filter is to be performed, an actual digital filter is obtained by expanding the coefficients of the reduced digital filter according to a mapping table and then used instead of the original digital filter.

Abstract: The basic concept of the present invention is to extrapolate a partially known spatial covariance matrix of a multi-channel signal in the parameter domain. The extrapolated covariance matrix is used with the downcoded downmix signal in order to efficiently generate an estimate of a linear combination of the multi-channel signals.

Abstract: A method for audio coding and decoding comprises primary encoding of a present audio signal sample into an encoded representation (T(n)), and non-causal encoding of a first previous audio signal sample into an encoded enhancement representation (ET(n?N+)). The method further comprises providing of the encoded representations to an end user. At the end user, the method comprises primary decoding of the encoded representation (T*(n)) into a present received audio signal sample, and non-causal decoding of the encoded enhancement representation (ET*(n?N+)) into an enhancement first previous received audio signal sample. The method further comprises improving of a first previous received audio signal sample, corresponding to the first previous audio signal sample, based on the enhancement first previous received audio signal sample. Devices and systems for audio coding and decoding are also presented.

Abstract: The invention provides a highly efficient technique for encoding a multi-channel audio signal. The invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoder (130) and encoding a second signal representation of one or more of the multiple channels in a second, multi-stage, encoder (140). This procedure is significantly enhanced by providing a controller (150) for adaptively allocating a number of encoding bits among the different encoding stages of the second, multi-stage, encoder (140) in dependence on multi-channel audio signal characteristics.

Abstract: The invention concerns a frequency-domain error concealment technique for information that is represented, on a frame-by-frame basis, by coding coefficients. The basic idea is to conceal an erroneous coding coefficient by exploiting coding coefficient correlation in both time and frequency. The technique is applicable to any information, such as audio, video and image data, that is compressed into coding coefficients and transmitted under adverse channel conditions. The error concealment technique proposed by the invention has the clear advantage of exploiting the redundancy of the original information signal in time as well as frequency. For example, this offers the possibility to exploit redundancy between frames (inter-frame) as well as within frames (intra-frame). The use of coding coefficients from the same frame as the erroneous coding coefficient is sometimes referred to as intra-frame coefficient correlation and it is a special case of the more general frequency correlation.