Abstract: Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

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: A method of transmitting an input audio signal is disclosed. A current spectral magnitude of the input audio signal is quantized. A quantization error of a previous spectral magnitude is fed back to influence quantization of the current spectral magnitude. The feeding back includes adaptively modifying a quantization criterion to form a modified quantization criterion. A current quantization error is minimized by using the modified quantization criterion. A quantized spectral envelope is formed based on the minimizing and the quantized spectral envelope is transmitted.

Abstract: Some methods may involve receiving a frame of encoded audio data that includes transform coefficient data. The transform coefficient data may include exponent data and mantissa data. The mantissa data may include mantissa values that were encoded with uniform or non-uniform boundaries of quantization intervals. The mantissa values may be reconstructed based, at least in part, on exponent profile data. Based on the exponent profile data, statistics regarding the pre-quantization mantissas values may be inferred. The exponent profile data may include exponent differential data. Some such exponent differential data may be exponent difference pairs, though more than two exponent differential data points may be evaluated in alternative methods. At each frequency bin, mantissa value reconstruction may be conditioned on the exponent differential data, e.g., on the exponent difference pairs.

Abstract: A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.

Abstract: A method and device for coding an input sound signal in at least one lower layer and at least one upper layer of an embedded codec comprises, in the at least one lower layer, coding the input sound signal to produce coding parameters, wherein coding the input sound signal comprises producing a synthesized sound signal. An error signal is computed as a difference between the input sound signal and the synthesized sound signal and a spectral mask is calculated as a function of a minima of a spectrum related to the input sound signal. In the at least one upper layer, the error signal is coded to produce coding coefficients, the spectral mask is applied to the coding coefficients, and the masked coding coefficients are quantized. Applying the spectral mask to the coding coefficients reduces the quantization noise produced upon quantizing the coding coefficients.

Abstract: A method of encoding one or more parent blocks of values, the number of values being the length of each block, the method comprising for each parent block: (a) determining a first sum of values in the parent block; (b) splitting the parent block into smaller subblocks; (c) for at least one of the subblocks, determining a second sum of the values in the subblock, selecting a likelihood table from the plurality of likelihood tables based on said first sum of values in the parent block and encoding the second sum using the likelihood table; (d) designating each subblock a parent block; (e) carrying out steps (a), (b), (c) and (d) until at least one parent block reaches a predetermined condition.

Abstract: A method of transceiving an audio signal is disclosed. An input audio signal is provided. It is determined whether an energy attack signal exists within the input audio signal and a decision flag is set if the energy attack signal exists. A temporal location of the energy attack point in the input audio signal is detected. Energy variations before and after the temporal location of an energy attack point are determined. The energy variations to produce quantized energy variations and a peak area energy of the input audio signal to produce a quantized peak area energy are quantized. The decision flag, the temporal location of the energy attack point, the quantized energy variations, and the quantized peak energy are transmitted.

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 apparatus and method for processing an audio signal including extracting noise filling flag information indicating whether noise filling is used to a plurality of frames; extracting coding scheme information indicating whether a current frame included in the plurality of frames is coded in either a frequency domain or a time domain; when the noise filling flag information indicates that the noise filling is used for the plurality of frames and the coding scheme information indicates that the current frame is coded in the frequency domain, extracting noise level information for the current frame; when a noise level value corresponding to the noise level information meets a predetermined level, extracting noise offset information for the current frame; and, when the noise offset information is extracted, performs the noise-filling for the current frame based on the noise level value and the noise offset information.

Abstract: Provided is an audio encoding device which can detect an optimal pitch pulse when using pitch pulse information as redundant information.

Abstract: A signal encoding device for encoding an input time-series signal includes: partitioning means; low-frequency encoding means; high-frequency gain information generating means; low-frequency reference value information generating means; high-frequency gain difference information generating means; high-frequency gain difference information encoding means; and multiplexing means.

Abstract: In one embodiment, the present invention comprises a vocoder having at least one input and at least one output, an encoder comprising a filter having at least one input operably connected to the input of the vocoder and at least one output, a decoder comprising a synthesizer having at least one input operably connected to the at least one output of the encoder, and at least one output operably connected to the at least one output of the vocoder, wherein the decoder comprises a memory and the decoder is adapted to execute instructions stored in the memory comprising phase matching and time-warping a speech frame.

Abstract: A vector quantization apparatus performs coding of a linear predictive coding coefficient by multi-stage vector quantization. A first codebook and a second codebook store code vectors, and a storing section stores scalars. A first quantizing section extracts a first code vector stored in the first codebook and performs first stage quantization for quantizing a target vector using the first code vector. A second quantizing section extracts a second code vector stored in the second codebook, calculates a third code vector by multiplying the second code vector and one of the scalars stored in the storing section, performs distance calculation using the target vector, the first code vector and the third code vector, and performs second stage quantization for quantizing the target vector using a result of the distance calculation. Each scalar stored in the storing section is associated with at least one of the vectors stored in the first codebook.

Abstract: A system for encoding speech includes a speech encoder (106, FIG. 1), a speech recognizer (110), and a difference encoder (108). When the speech recognizer (110) recognizes a word, phoneme or feature within an input speech signal (122), the difference encoder (108) calculates the differences between speech parameters (140, 142) derived by the speech encoder (106) and speech parameters (146, 148) derived by the speech recognizer (110). The difference encoder (108) quantizes the differences (128), which replace corresponding encoder-derived parameters to be transmitted over a channel (130). In one embodiment, the difference encoder representation (128) of the speech parameters consumes fewer bits than the encoder-derived representation (124). Accordingly, the resulting bandwidth consumed by a single channel can be decreased.

Abstract: This invention proposes a more efficient way to quantize temporal envelope shaping of high band signal by benefiting from energy relationship between low band signal and high band signal; if low band signal is well coded or it is coded with time domain codec such as CELP, temporal envelope shaping information of low band signal can be used to predict temporal envelope shaping of high band signal; the temporal envelope shaping prediction can bring significant saving of bits to precisely quantize temporal envelope shaping of high band signal. This prediction approach can be combined with other specific approach to further increase the efficiency and save mores bits.

Abstract: An encoding device includes: a frequency region converter which converts an inputted audio signal into a frequency region; a band selector which selects a quantization object band from a plurality of sub bands obtained by dividing the frequency region; and a shape quantizer which quantizes the shape of the frequency region parameter of the quantization object band. When a prediction encoding presence/absence determiner determines that the number of common sub bands between the quantization object band and the quantization object band selected in the past is not smaller than a predetermined value, a gain quantizer performs prediction encoding on the gain of the frequency region parameter of the quantization object band. When the number of common sub bands is smaller than the predetermined value, the gain quantizer non-predictively encodes the gain of the frequency region parameter of the quantization object band.

Abstract: An advanced audio coding (AAC) encoder quantization architecture is described. The architecture includes an efficient, low computation complexity approach for estimating scalefactors in which a base scalefactor estimate is adjusted by a delta scalefactor estimate that is based, in part, on global scalefactor adjustments applied to the previously quantized/encoded frame. Using such feedback, the AAC encoder quantization architecture is able to produce scalefactor estimates that are very close to the actual scalefactor applied by the subsequent quantization and encoding process. The architecture further includes a frequency hole avoidance approach that reduces a magnitude of an estimated scalefactor to avoid generating frequency holes in quantized SFBs.

Abstract: In a device configurable to encode speech performing an closed loop re-decision may comprise representing a speech signal by amplitude components and phase components for a current frame and a past frame. In a first closed loop stage, a first set of compressed components and a first set of uncompressed components for a current frame may be generated. A first set of features may be generated by comparing current and past frame amplitude and/or phase components. In a second closed loop stage, a second set of compressed components for the current frame may be generated by compressing the first set of compressed components and compressing the first set of uncompressed components. Generation of a second set of features may be based on the second set of compressed components from the current frame and a combination of amplitude and/or phase components from the past frame.

Abstract: A system and method is described for compensating for the effects of a corrupted Continuously Variable Delta Slope Modulation (CVSD) decoder memory state on a decoded audio signal. In accordance with the system and method, a first estimated step size associated with a first frame of the decoded audio signal is calculated and a second estimated step size associated with a replacement frame generated to conceal bit errors in the first frame of the decoded audio signal is calculated. At least a second frame of the decoded audio signal is then modified based on the first estimated step size and the second estimated step size.

Abstract: A system and method for encoding and decoding speech signals that includes a specially-designed Code Excited Linear Prediction (CELP) encoder and a vector quantization (VQ) based Noise Feedback Coding (NFC) decoder or that includes a specially-designed VQ-based NFC encoder and a CELP decoder. The VQ based NFC decoder may be a VQ based two-stage NFC (TSNFC) decoder. The specially-designed VQ-based NFC encoder may be a specially-designed VQ based TSNFC encoder. In each system, the encoder receives an input speech signal and encodes it to generate an encoded bit stream. The decoder receives the encoded bit stream and decodes it to generate an output speech signal. A system and method is also described in which a single decoder receives and decodes both CELP-encoded audio signals as well as VQ-based NFC-encoded audio signals.

Abstract: A target vector is coded by multi-stage vector quantization. A first stage of the coding of the target vector uses a first code vector stored in a first codebook. A scalar associated with a code of each first code vector is stored. A third code vector is determined by multiplying a second code vector stored in a second codebook and the scalar together, performing distance calculation using the target vector, the first code vector and the third code vector, and performing a second stage of the coding of the target vector using a result of the distance calculation.

Abstract: A decoding apparatus includes a standard predicted image generating unit which generates a standard decoded image and a standard predicted image of standard image quality; a non-standard decoded image generating unit which generates a non-standard decoded image different in image quality based on a prediction error information; a non-standard predicted image generating unit which generates a non-standard predicted image different in image quality based on the non-standard decoded image; and a correction value calculating unit which calculates correction values corresponding to differences between the standard and non-standard predicted images. The non-standard decoded image generating unit includes a predicted image reconstructing unit which corrects the non-standard predicted image, and a decoding unit which inverse-quantizes the prediction error information and adds the inverse-quantized prediction error information and the corrected non-standard predicted image to generate the non-standard decoded image.

Abstract: Methods of spectral partitioning which may be implemented in an encoder are described. The methods comprise determining an estimate of bit requirements for each of a plurality of spectral sub-bands. These estimates are then used to group the sub-bands into two or more regions by minimizing a cost function. This cost function is based on the estimates of bit requirements for each sub-band and the estimates may include estimates of code bit requirements and/or additional code bit requirements for each sub-band. These estimates may be determined in many different ways and a number of methods are described.

Abstract: A method for stereo audio perceptual encoding of an input signal includes masking threshold estimation and bit allocation. The masking threshold estimation and bit allocation are performed once every two encoding processes. Another method for stereo audio perceptual encoding of an input signal includes performing a time-to-frequency transformation, performing a quantization, performing a bitstream formatting to produce an output stream, and performing a psychoacoustics analysis. The psychoacoustics analysis includes masking threshold estimation on a first of every two successive frames of the input signal.

Abstract: A sound encoding device enabling the amount of delay to be kept small and the distortion between frames to be mitigated. In the sound encoding device, a window multiplication part (211) of a long analysis section (21) multiplies a long analysis frame signal of analysis length M1 by an analysis window, the resultant signal multiplied by the analysis window is outputted to an MDCT section (212), and the MDCT section (212) performs MDCT of the input signal to obtain the transform coefficients of the long analysis frame and outputs it to a transform coefficient encoding section (30). The window multiplication part (221) of a short analysis section (22) multiplies a short analysis frame signal of analysis length M2 (M2<M1) by an analysis window and the resultant signal multiplied by the analysis window is outputted to the MDCT section (222).

Abstract: A post filter and a decoder enabling improvement of the sound quality of a decoded signal even when the sound quality of the decoded signal is different from the bands are disclosed. A frequency converting section determines a decoded spectrum. A power spectrum computing section computes the power spectrum from the decoded spectrum. A correction band determining section determines the band in which the power spectrum is corrected according to layer information. A power spectrum correcting section corrects the power spectrum in the corrected band in such a way that the variation along the frequency axis is suppressed. An inverse converting section subjects the corrected power spectrum to inverse conversion to determine an autocorrelation function. An LPC analyzing section determines an LPC coefficient of the determined autocorrelation function.

Abstract: A module (402) for binary coding of a signal envelope, comprising coding module (502) for coding a variable length first mode. The coding module for coding the first mode incorporates an envelope saturation detector and said coding module (402) further includes a second coding module (503) for coding a second mode in parallel with the coding module (502) for coding the first mode and a mode selector (504) adapted to select one of the two coding modes as a function of a code length criterion and of the result from the envelope saturation detector.

Abstract: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames and computing a set of model parameters for the frames. The set of model parameters includes at least a first parameter conveying pitch information. The voicing state of a frame is determined and the first parameter conveying pitch information is modified to designate the determined voicing state of the frame, if the determined voicing state of the frame is equal to one of a set of reserved voicing states. The model parameters are quantized to generate quantizer bits which are used to produce the bit stream.

Abstract: An information coding apparatus includes a predictive signal generator that generates a predictive signal; a predictive residual signal generator that generates a predictive residual signal; a quantizer that quantizes a quantization input signal generated based on the predictive residual signal; a quantization error signal generator that generates a quantization error signal; a feedback signal generator that generates a feedback signal for controlling the frequency characteristic of the quantization noise after decoding based on the quantization error signal; and a quantization input signal generator that generates the quantization input signal.

Abstract: A down sampler 13 down samples a digital signal in the sampling frequency thereof from 96 kHz to 48 kHz on a frame-by-frame basis. The converted signal is compression encoded and output as a main code Im. An up sampler 16 converts a partial signal corresponding to the main code Im to a signal having the original sampling frequency 96 kHz, for example. An error signal between the up sampled signal and an input digital signal is generated. An array converting and encoding unit 18 array converts bits of sample chains of the error signal, thereby outputting an error code Pe. On a decoding side, a high fidelity reproduced signal is obtained based on the main code Im and the error code Pe, or a reproduced signal is obtained based on the main code Im only.

Abstract: A method and apparatus for expanding a bandwidth of an input narrowband voice signal is provided. The narrowband voice signal is analyzed separately for each frame, and a Degree of Voicing (DV) and a Degree of Stationary (DS) are calculated depending on the analysis. A Degree of Difficulty of Bandwidth Expansion (DDBWE) of the narrowband voice signal is calculated based on DV and DS. Bandwidth expansion is controlled according to DDBWE.

Abstract: A transform coding apparatus includes an input scale factor calculating section that calculates an input scale factor having a predetermined number of scale factors associated with an input spectrum as an element, and a codebook that stores a plurality of scale factor candidates having a predetermined number of elements and outputs one scale factor candidate. The transform coding apparatus also includes an error calculating section that calculates an error on a per element basis, a weighted error calculating section that determines a weight on a per element basis and calculates a sum of products of the error and the weight to calculate a weighted error, and a searching section that searches for a scale factor candidate that minimizes the weighted error in the codebook.

Abstract: An encoding device reduces the encoding distortion as compared to the conventional technique and obtains a preferable sound quality for auditory sense. In the encoding device, a shape quantization unit quantizes the shape of an input spectrum with a small number of pulse positions and polarities. The shape quantization unit sets a pulse amplitude width to be searched later upon search of the pulse position to a value not greater than the pulse amplitude width which has been searched previously. A gain quantization unit calculates a gain of a pulse searched by the shape quantization unit for each of bands.

Abstract: A method of encoding one or more parent blocks of values, the number of values being the length of each block, the method comprising for each parent block: (a) determining a first sum of values in the parent block; (b) splitting the parent block into smaller subblocks; (c) for at least one of the subblocks, determining a second sum of the values in the subblock, selecting a likelihood table from the plurality of likelihood tables based on said first sum of values in the parent block and encoding the second sum using the likelihood table; (d) designating each subblock a parent block; (e) carrying out steps (a), (b), (c) and (d) until at least one parent block reaches a predetermined condition.

Abstract: Provided are a method and apparatus for coding and decoding an amplitude of partials, in which a step phenomenon can be prevented in the result of coding the amplitude of continuation partial partials in a parametric codec, thereby improving reproduced sound quality. The method of coding the amplitude of partials includes obtaining an inversely quantized amplitude of partials of a previous frame, determining a quantization level based on a function for the inversely quantized amplitude of the partials of the previous frame, and quantizing an amplitude of partials of a current frame based on the determined quantization level.

Abstract: Systems and methods are provided for ultra-low latency decompression for a general-purpose audio input signal. In accordance with one implementation, a computer-implemented method is provided that includes decoding, by a processor, an input bit stream into quantization indices and residue quantization indices; applying an inverse quantization algorithm to the quantization indices to generate signal coefficients; applying an inverse transform to the signal coefficients to generate a time-domain reconstructed signal waveform; applying a stochastic noise synthesis algorithm to the residue quantization indices to generate a time-domain reconstructed residue waveform; combining, by the processor, the reconstructed signal waveform and the reconstructed residue waveform as a reconstructed signal waveform block; and generating an output signal by applying a boundary synthesis algorithm to the reconstructed signal waveform blocks.

Abstract: The invention relates to a method for generating a vector quantization dictionary for a signal, of the type comprising a statistical analysis of driving vectors representing the signal determining a finite set of code vectors which are proxies for the said driving vectors, wherein the method also comprises modifying the said finite set of code vectors to impose a minimum distance between the modified code vectors two by two, these modified code vectors forming the said dictionary.

Abstract: Disclosed is an encoding device that improves the quality of a decoded signal in a hierarchical coding (scalable coding) method, wherein a band to be quantized is selected for every level (layer). The encoding device (101) is equipped with a second layer encoding unit (205) that selects a first band to be quantized of a first input signal from among a plurality of sub-bands, and that generates second layer encoding information containing first band information of said band; a second layer decoding unit (206) that generates a first decoded signal using the second layer encoding information; an addition unit (207) that generates a second input signal using the first input signal and the first decoded signal; and a third layer encoding unit (208) that selects a second band to be quantized of the second input signal using the first decoded signal, and that generates third layer encoding information.

Abstract: A system, method and computer program product for classification of an analog electrical signal using statistical models of training data. A technique is described to quantize the analog electrical signal in a manner which maximizes the compression of the signal while simultaneously minimizing the diminution in the ability to classify the compressed signal. These goals are achieved by utilizing a quantizer designed to minimize the loss in a power of the log-likelihood ratio. A further technique is described to enhance the quantization process by optimally allocating a number of bits for each dimension of the quantized feature vector subject to a maximum number of bits available across all dimensions.

Abstract: Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. Each frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied, and (iii) window information. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes. Subband samples are then generated by dequantizing the decoded quantization indexes, and a sequence of different window functions that were applied within a single frame of the audio data is identified based on the window information.

Abstract: Provided are, among other things, systems, methods and techniques for merging entropy codebook application ranges within an audio signal. According to one embodiment, an audio signal is obtained, the audio signal including quantization indexes, identification of segments of said quantization indexes, and indexes of entropy codebooks that have been assigned to such segments, with a single entropy codebook index having been assigned to each said segment; potential merging operations in which specified segments potentially would be merged with each other are identified; bit penalties are estimated for the potential merging operations; then, the potential merging operation having the lowest estimated the penalty is performed.

Abstract: A sound decoding device is capable of improving the lost frame compensation performance and improving quality of the decoded sound. A rise frame sound source compensation unit generates a compensation sound source signal when the current frame is a lost frame and a rise frame. An average sound source pattern update unit updates the average sound source pattern held in an average sound source pattern holding unit over a plurality of frames. When a frame is lost, an LPC synthesis unit performs LPC synthesis on a decoded sound source signal by using the compensation sound source signal inputted via a switching unit and a decoded LPC parameter from an LPC decoding unit and outputs the compensation decoded sound signal.

Abstract: A multi-channel audio decoder reconstructs multi-channel audio of more than two physical channels from a reduced set of coded channels based on correlation parameters that specify a full power cross-correlation matrix of the physical channels, or merely preserve a partial correlation matrix (such as power of the physical channels, and some subset of cross-correlations between the physical channels, or cross-correlations of the physical channels with coded or virtual channels).

Abstract: Provided are a fixed codebook search method based on iteration-free global pulse replacement in a speech codec, and a Code-Excited Linear-Prediction (CELP)-based speech codec using the method.

Abstract: Disclosed is a vector quantisation device which can reduce the computational complexity of an audio codec without reducing the audio quality. A vector quantisation device (112) searches a codebook using code vectors, with which the impulse response of an audibility weighted synthesis filter is convolved and which configure the codebook, and target vectors. A filtering unit (201) applies a filter exhibiting a low pass and/or a high pass characteristic to the impulse response. If the filter has a high pass characteristic, a compaction unit (202) then compacts the degree of the post-filtering impulse response. A convolution unit (203) convolves the post-filtering impulse response with each of the code vectors. If the filter has a low pass characteristic, a search unit (204) thins out elements of the plurality of code vectors with which the impulse response has been convolved, and elements of the target vectors.

Abstract: Provided are a method, apparatus, and medium for encoding/decoding a high frequency band signal by using a low frequency band signal corresponding to an audio signal or a speech signal. Accordingly, since the high frequency band signal is encoded and decoded by using the low frequency band signal, encoding and decoding can be carried out with a small data size while avoiding deterioration of sound quality.

Abstract: There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).

Abstract: A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.

Abstract: Methods and apparatus for iteratively encoding a portion of a signal are described in which the portion of the signal is quantised and an output bit count is estimated based on the sum of logarithms to base n of values of each sample in the plurality of quantised samples and the total number of samples. The output bit count corresponds to an estimate of the output bit count for the portion of the signal once encoded using a code, such as a Huffman code.