Abstract: A sensor is configured to determine at least one operating condition of a device and a selector is configured to select an audio coding process for the device, based on the operating condition. The operating condition may include remaining battery life of the device and/or ambient noise level. The selected audio coding process may consume less power than another possible audio coding process during audio processing. The audio may include voice and/or audio playback, e.g., music playback.

Abstract: An digital watermark embedding device including an interface unit configured to acquire content in a digital form and digital watermark information, the device includes, a codeword generating unit configured to generate a base codeword including a bit sequence including the digital watermark information; a shifting unit configured to generate a plurality of correcting codewords differing from one another by permutating an arrangement in the bit sequence included in the base codeword depending on a plurality of shift amounts, the plurality of shift amounts differing from one digital watermark information to another, under a predetermined permutation rule; and a watermark superimposing unit configured to embed the plurality of correcting codewords in the content.

Abstract: A method and apparatus for processing a signal are discussed. According to an embodiment, the method includes receiving extension information and a downmix signal decoded by either an audio coding scheme or a speech coding scheme, the downmix signal having a bandwidth of a low frequency signal; generating an upmixing signal from the downmix signal by using channel extension; determining an extension base signal corresponding to partial band of the upmixing signal based on the extension information; and generating an extended upmixing signal by applying the extension information to the extension base signal, the extended upmixing signal having a bandwidth extended by reconstructing a high frequency signal.

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: An apparatus for obtaining a parameter describing a variation of a signal characteristic of a signal on the basis of actual transform-domain parameters describing the audio signal in transform-domain includes a parameter determinator. The parameter determinator is configured to determine one or more model parameters of a transform-domain variation model describing an evolution of the transform-domain parameters in dependence on one or more model parameters representing a signal characteristic, such that a model error, representing a deviation between a modeled temporal evolution of the transform-domain parameters and an evolution of the actual transform-domain parameters, is brought below a predetermined threshold value or minimized.

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: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract: A packet generator for generating packets from an input signal configured to: generate at least one first signal, dependent on the input signal, the first signal comprising a first relative time value; generate at least one second signal, dependent on the input signal and associated with the at least one first signal; and generate at least one indicator associated with each of the at least one second signal, each indicator dependent on the first relative time value.

Abstract: A method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions.

Abstract: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

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

Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes receiving filter information, applying spatial information to the filter information to generate surround converting information, and outputting the surround converting information. The apparatus for decoding an audio signal includes a filter information receiving part receiving filter information; an information converting part applying spatial information to the filter information to generate surround converting information, and a surround converting information output part outputting the surround converting information.

Abstract: A method and apparatus for prediction in a speech-coding system extends a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, to a multi-tap LTP filter. From another perspective, a conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Such a multi-tap LTP filter offers a number of advantages over the prior-art. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients (?i's) of the multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.

Abstract: A method of receiving an audio signal includes measuring a periodicity of the audio signal to determine a checked periodicity. At least one best available subband is determined. At least one extended subband is composed, wherein composing includes reducing a ratio of composed harmonic components to composed noise components if the checked periodicity is lower than a threshold, and scaling a magnitude of the at least one extended subband based on a spectral envelope on the audio signal.

Abstract: A method of processing a signal is disclosed. The present invention includes receiving extension information and at least one downmix signal of a first downmix signal decoded by a audio coding scheme and a second downmix signal decoded by a speech coding scheme; determining an extension base signal corresponding to a partial region of the downmix signal based on the extension information; and generating an extended downmix signal having a bandwidth extended by reconstructing a high frequency region signal using the extension base signal and the extension information. According to a signal processing method and apparatus of the present invention, signal corresponding to a partial frequency region of the downmix signal is used as the extension base signal. Therefore, the high frequency region of the downmix signal is reconstructed by using the extension base signal having variable bandwidth.

Abstract: A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data.

Abstract: Information useful for modifying the dynamics of an audio signal is derived from one or more devices or processes operating at one or more respective nodes of each of a plurality of hierarchy levels, each hierarchical level having one or more nodes, in which the one or more devices or processes operating at each hierarchical level takes a measure of one or more characteristics of the audio signal such that the one or more devices or processes operating at each successively lower hierarchical level takes a measure of one or more characteristics of progressively smaller subdivisions of the audio signal.

Abstract: Codebook indices for a scalable speech and audio codec may be efficiently encoded based on anticipated probability distributions for such codebook indices. A residual signal from a Code Excited Linear Prediction (CELP)-based encoding layer may be obtained, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal may be transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum. The transform spectrum is divided into a plurality of spectral bands, where each spectral band having a plurality of spectral lines. A plurality of different codebooks are then selected for encoding the spectral bands, where each codebook is associated with a codebook index. A plurality of codebook indices associated with the selected codebooks are then encoded together to obtain a descriptor code that more compactly represents the codebook indices.

Abstract: The present disclosure includes processing a signal to generate a first sub-set of data, transmitting the first sub-set of data for generation of a reconstructed audio signal, the reconstructed audio signal having a fidelity relative to the signal, processing the signal to generate a second sub-set of data and a third sub-set of data, the second sub-set of data defining a second portion of the signal and comprising data that is different than data of the first sub-set of data, and the third sub-set of data defining a third portion of the signal and comprising data that is different than data of the first and second sub-sets of data, comparing a priority of the second sub-set of data to a priority of the third sub-set of data, and transmitting one of the second sub-set of data and the third sub-set of data over the network for improving the fidelity.

Abstract: An audio encoding method and apparatus and an audio decoding method and apparatus are provided. The audio signal decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating a modified downmix signal based on the downmix signal and extracted information which is extracted from the object-based side information; generating channel-based side information based on the object-based side information and control data for rendering the downmix signal; and generating a multi-channel audio signal based on the modified downmix signal and the channel-based side information.

Abstract: A method of encoding a time-domain audio signal is presented. A device transforms the time-domain signal into a frequency-domain signal including a sequence of sample blocks, wherein each block includes a coefficient for each of multiple frequencies. The coefficients of each block are grouped into frequency bands. For each frequency band of each block, a scale factor is estimated for the band, and the energy of the band for the block is compared with the energy of the band of an adjacent sample block, wherein the blocks may be adjacent to each other in either or both of an interchannel and a temporal sense. If the ratio of the band energy for the first block to the band energy for the adjacent block is less than some value, the scale factor of the band for the first block is increased. The coefficients of the band for each block are quantized based on the resulting scale factor. The encoded audio signal is generated based on the quantized coefficients and the scale factors.

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.

Abstract: A stereo encoding method and apparatus are provided, so as to reduce distortion caused by delay adjustment. The stereo encoding method includes: extracting a current interchannel delay of a stereo signal and a previous delay adjacent to the current interchannel delay; performing adjustment frame judgment according to characteristics of the current stereo signal when the current delay and the previous delay are different; and performing delay adjustment on the stereo signal by using the current interchannel delay if it is judged that a frame where the current delay occurs is an adjustment frame.

Abstract: A method and an apparatus for generating a lattice vector quantizer codebook are disclosed. The method includes: storing an eigenvector set that includes amplitude vectors and/or length vectors, where the amplitude vectors and/or length vectors are different from each other and correspond to a root leader of a lattice vector quantizer; storing storage addresses of the amplitude vectors and length vectors, where the amplitude vectors and length vectors correspond to the root leader and are in the eigenvector set; and generating a lattice vector quantizer codebook according to the eigenvector set and the storage addresses.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: A method and an apparatus for processing a signal are provided. The method includes: obtaining an energy average value of each sub-band for a current frame frequency-domain signal; obtaining a current frame modification coefficient of each sub-band for the current frame frequency-domain signal according to a spectral envelope and the energy average value of each sub-band; obtaining a weighted modification coefficient of each sub-band for the current frame frequency-domain signal by using the current frame modification coefficient and a relevant frame modification coefficient; and modifying the spectral envelope of each sub-band for the current frame frequency-domain signal by using the weighted modification coefficient.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: An audio encoder/decoder uses a combination of an overlap windowing transform and block transform that have reversible implementations to provide a reversible, integer-integer form of a lapped transform. The reversible lapped transform permits both lossy and lossless transform domain coding of an audio signal having variable subframe sizes.

Abstract: There is provided an audio encoding device capable of effectively encoding stereo audio in audio encoding having monaural-stereo scalable configuration. In this device, a correlation degree comparison unit (304) calculates correlation in a first channel (correlation degree between the past signal and the current signal in the first channel) from the first channel audio signal and calculates correlation in a second channel (correlation degree between the past signal and the current signal in the second channel) from the second channel audio signal. The correlation in the first channel is compared to the correlation in the second channel. A channel having the greater correlation is selected.

Abstract: There is provided an audio encoding device capable of effectively encoding a stereo audio even when a correlation between channels of the stereo audio is small. In the device, a monaural signal generation unit (110) generates a monaural signal by using a first channel signal and a second channel signal contained in the stereo signal. An encoding channel selection unit (120) selects one of the first channel signal and the second channel signal. An encoding unit including a monaural signal encoding unit (112), a first channel encoding unit (122), a second channel encoding unit (124), and a switching unit (126) encodes the generated monaural signal to obtain core-layer encoded data and encodes the selected channel signal to obtain extended layer encoded data corresponding to the core-layer encoded data.

Abstract: An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoder encodes an input signal and generates first encoded information, a first layer decoder decodes the first encoded information and generates a first decoded signal, a characteristic judger analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoder changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information.

Abstract: Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.

Abstract: This invention relates to a method, a computer program product, apparatuses and a system for extracting coded parameter set from an encoded audio/speech stream, said audio/speech stream being distributed to a sequence of packets, and generating a time scaled encoded audio/speech stream in the parameter coded domain using said extracted coded parameter set.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: A frame is received that has the wideband audio signal. The low band audio signal is encoded to generate an encoded low band signal. The high band signal is analyzed to determine whether the high band signal is perceptually relevant to the low band signal. If the high band signal is not perceptually relevant to the low band signal, the low band signal is encoded and provided in a frame to the decoder without including parameters corresponding to characteristics of the high band signal. If the high band signal is perceptually relevant, the high band signal is encoded to generate an encoded high band signal. The resultant frame that is sent to the decoder will include a combination of the encoded low band signal and the encoded high band signal.

Abstract: Provided are a method and apparatus for encoding the frequency of a continuation sinusoidal signal and a method and apparatus for decoding the same. In the encoding method, a continuation sinusoidal signal successive to a sinusoidal signal in a previous section is extracted from a current section; a frequency of the continuation sinusoidal signal at the boundary between the current and previous sections is changed to a first frequency, based on representative frequencies of the continuation sinusoidal signal and at least one sinusoidal signal that belongs to a section adjacent to the current section and is successive to the continuation sinusoidal signal; and the first frequency is encoded.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.

Abstract: An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: Disclosed is a code conversion method to convert a first code sequence conforming to a first speech coding scheme into a second code sequence conforming to a second speech coding scheme. The method includes the following steps. The first step discriminates whether the first code sequence corresponds to a speech part or to a non-speech part, and generates a numerical value that indicates the discrimination result as a control flag. The second step converts the first code sequence into the second code sequence and outputs said second code sequence, when the value of the control flag corresponds to the speech part. The third step outputs the second code sequence that corresponds to the value of the control flag, when the value of the control flag corresponds to the non-speech part.

Abstract: A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

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: Methods and apparatus to audio watermarking and watermark detection and extracted are described herein. An example method includes receiving a media content signal, sampling the media content signal to generate samples, storing the samples in a buffer, determining a first sequence of samples in the buffer, determining a second sequence of samples in the buffer, wherein the second sequence of samples is of substantially equal length as the first sequence of samples, calculating an average of the first sequence of samples and the second sequence of samples to generate an average sequence of samples, extracting an identifier from the average sequence of samples, and storing the identifier in a tangible memory.

Abstract: A multi-channel audio encoder (10) for encoding a multi-channel audio signal (101), e.g. a 5.1 channel audio signal, into a spatial down-mix (102), e.g. a stereo signal, and associated parameters (104, 105). The encoder (10) comprises first and second units (110, 120). The first unit (110) encodes the multi-channel audio signal (101) into the spatial down-mix (102) and parameters (104). These parameters (104) enable a multi-channel decoder (20) to reconstruct the multi-channel audio signal (203) from the spatial down-mix (102). The second unit (120) generates, from the spatial down-mix (102), parameters (105) that enable the decoder to reconstruct the spatial down-mix (202) from an alternative down-mix (103), e.g. a so-called artistic down-mix that has been manually mixed in a sound studio. In this way, the decoder (20) can efficiently deal with a situation in which an alternative down-mix (103) is received instead of the regular spatial, down-mix (102).

Abstract: A method for minimizing message collision in a device is presented. Two or more overlapping real-time streaming simplex audio messages are received. One of the audio messages is forwarded to be reproduced while the other is stored. Forwarding of the delayed audio message is delayed such that overlapping reproduction of the audio messages is minimized. Reproduction is delayed until a predetermined clock time expires or is dependent on one or more of: the length of the second audio message or the amount of overlap of the first and second audio messages.

Abstract: An apparatus and method for encoding and decoding using mutual information between a high band signal and a low band signal to increase a coding efficiency in a portable terminal are provided. The apparatus includes a bandwidth extender for extracting auxiliary information relating to a characteristic of a high band signal using the high band signal and a low band signal and an encoder for encoding residual high band signal obtained by subtracting auxiliary information acquired from the low band signal from auxiliary information acquired from the high band signal.

Abstract: A method, a device, and a system for transcoding between two embedded codecs are disclosed. The method includes: delaying a first encoded stream in input streams for integer number of frames, where the first encoded stream includes a stream of at least one extension layer in the input streams obtained after input signals are encoded by using a first codec; and using the first codec to decode other encoded streams in the input streams to obtain the first decoded signal; and performing delay aligning and adjusting to obtain an adjusted signal so as to reduce the transcoding complexity and enhance quality of the transcoded signals.

Abstract: Aspects of a method and system for multi-band amplitude estimation and gain control in an audio CODEC are provided. In this regard, an audio signal may be filtered and delayed to generate one or more sub-band signals, a gain may be applied to each sub-band signal to generate one or more level adjusted sub-band signals, and the one or more level adjusted signals may be added to a delayed version of the audio signal. The gain applied to a particular one of the one or more sub-band signals may be controlled based on a detected amplitude of a summed signal derived by summing the particular one of the one or more sub-band signals and a corresponding one of the one or more level-adjusted sub-band signals.