Abstract: An apparatus and method for encoding and decoding a signal for high frequency bandwidth extension are provided. An encoding apparatus may down-sample a time domain input signal, may core-encode the down-sampled time domain input signal, may transform the core-encoded time domain input signal to a frequency domain input signal, and may perform bandwidth extension encoding using a basic signal of the frequency domain input signal.

Abstract: The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

Abstract: A system for collecting and storing performance data for an engine is provided. The system includes one or more sensors configured to generate sensor data signals representative of one or more engine data performance parameters. The system further includes a data sampling component, a data quantizing component, a data storage sampling rate component, a data encoding component and a data storage component. The data sampling component is configured to sample the sensor data signals at a data sampling rate. The data quantizing component is configured to generate quantized data samples corresponding to the sampled sensor data signals. The data storage sampling rate component is configured to determine a data storage sampling rate for the quantized data samples, based on an analysis of at least a subset of the quantized data samples.

Abstract: When performing audio communication by using different encoding/decoding methods, a code obtained by encoding audio by a certain method is converted into a code decodable by another method with a high audio quality and a small calculation amount. In a code conversion device for converting a first code string into a second code string, an audio decoding circuit acquires a first linear prediction coefficient and excitation signal information from the first code string and drives the filter having the first linear prediction coefficient by the excitation signal obtained from the excitation signal information, thereby creating a first audio signal. A fixed codebook code generation circuit uses the fixed codebook information and minimizes the distance between the second audio signal generated from the information obtained from the second code string and the first audio signal, thereby obtaining the fixed codebook information in the second code string.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: In accordance with an embodiment, a method of decoding an encoded audio bitstream at a decoder includes receiving the audio bitstream, decoding a low band bitstream of the audio bitstream to get low band coefficients in a frequency domain, and copying a plurality of the low band coefficients to a high frequency band location to generate high band coefficients. The method further includes processing the high band coefficients to form processed high band coefficients. Processing includes modifying an energy envelope of the high band coefficients by multiplying modification gains to flatten or smooth the high band coefficients, and applying a received spectral envelope decoded from the received audio bitstream to the high band coefficients. The low band coefficients and the processed high band coefficients are then inverse-transformed to the time domain to obtain a time domain output signal.

Abstract: In accordance with an embodiment, A method of encoding an audio bitstream at an encoder includes encoding an original low band signal at the encoder by using a closed loop analysis-by-synthesis approach to obtain a coded low band signal, encoding an original high band signal at the encoder by using an open loop energy matching approach to obtain coded high band energy envelopes, comparing an energy of the coded low band signal with an energy of a corresponding original low band signal for a subframe, and generating an indication flag that indicates whether an energy envelope perceptual correction is needed for the subframe based on comparing the energy.

Abstract: According to one embodiment, a sound quality control device includes: a time domain analysis module configured to perform a time-domain analysis on an audio-input signal; a frequency domain analysis module configured to perform a frequency-domain analysis on a frequency-domain signal; a first calculation module configured to calculate first speech/music scores based on the analysis results; a compensation filtering processing module configured to generate a filtered signal; a second calculation module configured to calculate second speech/music scores based on the filtered signal; a score correction module configured to generate one of corrected speech/music scores based on a difference between the first speech/music score and the second speech/music score; and a sound quality control module configured to control a sound quality of the audio-input signal based on the one of the corrected speech/music scores.

Abstract: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.

Abstract: Provided is a communication apparatus for direct communication between networks of different types. The communication apparatus includes a transmission data selector determining whether or not data input from a first communication network is speech data, a data processor digitizing and packetizing the data transferred from the transmission data selector, and a modem for converting the digitized and packetized data into analog data and then directly transmitting the analog data to a second communication network different from the first communication network through a speech channel.

Abstract: The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

Abstract: Perceptual audio codecs make use of filter banks and MDCT in order to achieve a compact representation of the audio signal, by removing redundancy and irrelevancy from the original audio signal. During quasi-stationary parts of the audio signal a high frequency resolution of the filter bank is advantageous in order to achieve a high coding gain, but this high frequency resolution is coupled to a coarse temporal resolution that becomes a problem during transient signal parts by producing audible pre-echo effects. The invention achieves improved coding/decoding quality by applying on top of the output of a first filter bank a second non-uniform filter bank, i.e. a cascaded MDCT. The inventive codec uses switching to an additional extension filter bank (or multi-resolution filter bank) in order to re-group the time-frequency representation during transient or fast changing audio signal sections.

Abstract: A “STAC Codec” provides audio transcoding and decoding by processing an encoded audio signal using a backward-adaptive run-length Golomb-Rice (RLGR) decoder to recover transform coefficients of the encoded audio signal. The transform coefficients are then either transcoded in the transform domain to lossy or other formats, or decoded to the time domain by applying an inverse integer-reversible modulated lapped transform (MLT) to the recovered transform coefficients to recover an uncompressed time domain representation compressed audio signal. In additional embodiments, an inter-block spectral estimation and inverse data sorting strategy is used in recovering the transform coefficients from the encoded audio signal.

Abstract: A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.

Abstract: The invention relates to an electronic device, which includes a voice user interface (VUI), speech-recognition devices (SR) for implementing the voice user interface (VUI), and memory (MEM), in which language-configuration data is arranged for the user interface (VUI, UI), including several language packages (LP1-LP9), in which packages (LP1-LP9) several languages (L1-L34) are grouped, of which at least some of the languages (L1-L34) may belong to several language packages (LP1-LP9), and at least one language package (LP1) is arranged to be selected for use in the user interface (VUI, UI). In the invention, the language package (LP1-LP9) is selected by the device.

Abstract: A method for supporting an encoding of an audio signal is shown, wherein at least a first and a second coder mode are available for encoding a section of the audio signal. The first coder mode enables a coding based on two different coding models. A selection of a coding model is enabled by a selection rule which is based on signal characteristics which have been determined for a certain analysis window. In order to avoid a misclassification of a section after a switch to the first coder mode, it is proposed that the selection rule is activated only when sufficient sections for the analysis window have been received. The invention relates equally to a module in which this method is implemented, to a device and a system comprising such a module and to a software program product including a software code for realizing the proposed method.

Abstract: Method and decoder for processing of audio signals. The method and decoder relate to deriving a processed vector {circumflex over (d)} by applying a post-filter directly on a vector d comprising quantized MDCT domain coefficients of a time segment of an audio signal. The post-filter is configured to have a transfer function H which is a compressed version of the envelope of the vector d. A signal waveform is reconstructed by performing an inverse MDCT transform on the processed vector {circumflex over (d)}.

Abstract: An object of the present invention is to achieve high coding efficiency for a companded signal sequence and reduce the amount of codes. A coding method according to the present invention includes an analysis step and a signal sequence transformation step. The analysis step is to check whether or not there is a number that is included in a particular range but does not occur in a second signal sequence (a number sequence that indicates the magnitude (magnitude relationship) of original signals) and output information that indicates the number that does not occur.

Abstract: A voice band enhancement apparatus is used that includes a frequency transform unit to perform frequency transform on an input signal to calculate a spectrum, a mapping function calculating unit to calculate, by use of the spectrum, a mapping function for generating high-range components from low-range components of the spectrum, a wide-band spectrum generating unit to generate, in a higher range than a band of the spectrum, a high-range spectrum based on the mapping function and to integrate the generated high-range spectrum and the spectrum calculated by the frequency transform unit, thereby generating a wide-band spectrum wider than the band of the spectrum calculated by the frequency transform unit, and an inverse frequency transform unit to perform inverse frequency transform on the wide-band spectrum to calculate an output signal.

Abstract: Methods of using individually distinctive patterns of voice characteristics to identify a speaker include computing the reassigned spectrogram of each of at least two voice samples, pruning each reassigned spectrogram to remove noise and other computational artifacts, and comparing (either visually or with the aid of a processor) the strongest points to determine whether the voice samples belong to the same speaker.

Abstract: A sampling rate conversion apparatus and a method thereof are provided which increase the sampling rate of a discrete audio signal sampled at a predetermined sampling rate by using a fractal interpolation function (FIF). An audio signal portion formed by a predetermined number of sampling data items is divided into a plurality of interpolation intervals. On the audio signal portion, mapping points are determined. The number of the mapping points is in accordance with the degree of increase in the sampling rate. For the respective interpolation intervals, mapping parameters for performing mapping using the FIF on the mapping points are calculated. In all of the interpolation intervals, the mapping using the FIF is performed on the mapping points with the use of the mapping parameters according to the respective interpolation intervals. Thereby, new sampling data items are generated.

Abstract: Provided are a methods and apparatuses for encoding/decoding an audio signal to efficiently encode/decode a harmonic envelope. The method of encoding an audio signal includes performing harmonic analysis with respect to an input signal to determine harmonic parameters with respect to harmonic signals; correlating the amplitudes of the harmonic signals to signals in a time domain instead of signals in a frequency domain; applying a time-frequency transformation operation to the amplitudes in the time domain to generate time-frequency transformed values in the frequency domain; and encoding the time-frequency transformed values. When expressing a harmonic envelope, the amplitudes of the harmonic signals are regarded as signals in the time domain so as to perform a time-frequency transformation and only a part from among the transformed values is selected to be encoded. Therefore, sound quality is not affected and coding efficiency greatly improves.

Abstract: The present invention relates to methods and devices for encoding and decoding digital audio signals, e.g. a speech signal. An audio coder and a decoder are provided wherein a modeller adds a first distribution model obtained from model parameters of past segments of the digital audio signal and a fixed distribution model, each of the models being multiplied by a weighting coefficient, for obtaining a combined distribution model. The weighting coefficients are selected to minimize a code length of a current segment of the digital audio signal. As the combined distribution model is a sum of several distribution models, wherein at least some of the models is based on the model parameters, flexibility is introduced in the signal model used to encode the digital audio signal. Thus, an audio coder and decoder providing a low bit rate in average, low bit rate variations and low error propagation are provided.

Abstract: A speech signal compression and/or decompression method, medium, and apparatus in which the speech signal is transformed into the frequency domain for quantizing and dequantizing information of frequency coefficients. The speech signal compression apparatus includes a transform unit to transform a speech signal into the frequency domain and obtain frequency coefficients, a magnitude quantization unit to transform magnitudes of the frequency coefficients, quantize the transformed magnitudes and obtain magnitude quantization indices, a sign quantization unit to quantize signs of the frequency coefficients and obtain sign quantization indices, and a packetizing unit to generate the magnitude and sign quantization indices as a speech packet.

Abstract: A method for decoding audio frames includes producing a first frame of coded audio samples, producing at least a portion of a second frame of coded audio samples, generating audio gap filler samples based on parameters representative of a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples, and forming a sequence including the audio gap filler samples and the portion of the second frame of coded audio samples.

Abstract: An adaptive encoding method includes splitting an input signal into a low-frequency band signal and a high-frequency band signal; performing forward adaptive linear prediction on the low-frequency band signal and thus filtering the low-frequency band signal; selectively performing backward adaptive linear prediction or long-term prediction on the filtered low-frequency band signal according to the analysis result of the low-frequency band signal; transforming the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, into a signal in a frequency domain and quantizing the signal; and encoding the high-frequency band signal using the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, or the quantized signal. Therefore, compression efficiency of both speech and music signals can be enhanced, and a robust compression method can be provided for various audio contents at a low bit rate.

Abstract: Methods and systems are provided for gathering research data that includes information pertaining to audio signals received on a portable device, such as a cell phone. Frequency domain data is received or produced, a signature is extracted from the frequency domain data and an ancillary code is read from the frequency domain data.

Abstract: A sound pickup apparatus includes: a microphone array including at least three microphones, wherein a first pair of microphones in which two of the at least three microphones are aligned on a first axis, and a second pair of microphones in which two of the at least three microphones are aligned on a second axis; a first null signal generator which outputs a first null signal based on a differential output of the first pair of microphones; a second null signal generator which outputs a second null signal based on a differential output of the second pair of microphones; and a combiner which generates a target signal based on the first null signal and the second null signal, the target signal having a directional characteristic in which the lowest sensitivity is formed in a direction to a line along which the first null surface meets the second null surface.

Abstract: A “STAC Codec” provides lossless audio compression and decompression by processing an audio signal using integer-reversible modulated lapped transforms (MLT) to produce transform coefficients. Transform coefficients are then encoded using a backward-adaptive run-length Golomb-Rice (RLGR) encoder to produce losslessly compressed audio signals. In additional embodiments, further compression gains are achieved via an inter-block spectral estimation and data sorting strategy. Further, compression in the transform domain allows the bitstream to be partially decoded, using the corresponding RLGR decoder, to reconstruct the frequency-domain coefficients. These frequency-domain coefficients are then directly used to speed up various transform-domain based applications such as transcoding media to lossy or other formats, search, identification, visualization, watermarking, etc.

Abstract: Methods, and corresponding codec-containing devices are provided that have source coding schemes for encoding a component of an excitation. In some cases, the source coding scheme is an enumerative source coding scheme, while in other cases the source coding scheme is an arithmetic source coding scheme. In some cases, the source coding schemes are applied to encode a fixed codebook component of the excitation for a codec employing codebook excited linear prediction, for example an AMR-WB (Adaptive Multi-Rate-Wideband) speech codec.

Abstract: An encoder generating a decoded signal with an improved quality by scalable encoding by canceling the characteristic inherent to the encoder and causing degradation of quality of the decoded signal.

Abstract: The encoding apparatus includes an MDCT unit which transforms an inputted audio signal into a frequency parameter, for every predetermined time-frequency transformation frame length, and an MDCT coefficient encoding unit which encodes the frequency parameter. The encoding apparatus also includes a pitch cycle detection unit which detects a pitch cycle of the audio signal, a framing unit which frames the audio signal based on the detected pitch cycle, and a waveform modification unit which performs waveform modification on the audio signal framed based on the pitch cycle, in conformance with the time-frequency transformation frame length, and outputs the waveform-modified audio signal to the MDCT unit. A multiplex unit multiplexes the frequency parameter encoded by MDCT coefficient encoding unit and the pitch cycle, and outputs the multiplexed result as a bitstream.

Abstract: Methods and apparatus for audio and speech processing including generating a plurality of frames, each of the frames comprising a plurality of transform coefficients, and allocating bits to the transform coefficients in each of the frames such that at least two of the transform coefficients in the same frame have different bit allocations and the total number of the bits allocated to the transform coefficients in at least two of the frames is equal.

Abstract: An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal.

Abstract: Disclosed is a spectral smoothing device with a structure whereby smoothing is performed after a nonlinear conversion has been performed for a spectrum calculated from an audio signal, and with which the amount of processing calculation is significantly reduced while maintaining excellent audio quality. With this spectral smoothing device, a sub band division unit (102) divides an input spectrum into multiple sub bands; a representative value calculation unit (103) calculates a representative value for each sub band using an arithmetic mean and a geometric mean; with respect to each representative value, a nonlinear conversion unit (104) performs a nonlinear conversion the characteristic of which is further emphasized as the value increases; and a smoothing unit (105) that smoothes the representative value which has undergone the nonlinear conversion for each sub band, at the frequency domain.

Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. At least two transforms are applied on an audio signal, each with different transform periods for better resolutions at both low and high frequencies. The transform coefficients are selected and combined such that the data rate remains similar as a single transform. The transform coefficients may be coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

Abstract: Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.

Abstract: The invention includes several techniques and tools, which can be used in combination or separately. For example, an audio encoder can encode information directly using coding processes that include a windowed overlapped transform, a selective multi-channel transform, scalar quantization and entropy encoding. The audio encoder can also encode information parametrically according to a parametric compression mode that accounts for audibility of distortion according to an auditory model. A corresponding audio decoder can decode first information directly and second information according to the parametric mode.

Abstract: A data communication system for communicating one or more payload streamed data signals and an auxiliary data signal, the auxiliary data signal being arranged as one or more data packets according to a data packet protocol, each packet having a packet destination address. The system includes at least two data handling nodes, a transmitting one of the data handling nodes arranged to transmit data to a receiving one of the data handling nodes, and a transmission data formatter associated with the transmitting node for formatting the packets of the auxiliary data signal into a streamed data signal format and for multiplexing the payload streamed data signals and the formatted auxiliary data signal into a bitstream for transmission. The system further includes a received data reformatter associated with the receiving node for demultiplexing the input streamed data signals and the formatted auxiliary data signal and for reformatting the auxiliary data signal into packets according to the data packet protocol.

Abstract: Methods and systems are provided for gathering research data. Frequency domain data is received or produced, a signature is extracted from the frequency domain data and an ancillary code is read from the frequency domain data.

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: Provided is an encoder which can decode a high-quality stereo signal while keeping the amount of information in the bit allocation information to a minimum when a scalable coding technique is used for a stereo signal. In the encoder, a principal component analysis (PCA) converter (101) PCA converts the left signal and the right signal of the stereo signal and generates the main signal of the first layer and the sub-signal of the first layer. In the first layer to the M-th layer (where M is a natural number, 2 or greater), an adaptive residual encoder (102-m) (where m is a natural number from 1 to M) compares the importance of the main signal of the m-th layer and the importance of the sub-signal of the m-th layer, selects the signal having the higher importance, encodes the selected signal, and generates the encoded data of the m-th layer.

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: Methods and units are shown for supporting a switching from a first coding scheme to a Modified Discrete Cosine Transform (MDCT) based coding scheme calculating a forward or inverse MDCT with a window (h(n)) of a first type for a respective coding frame, which satisfies constraints of perfect reconstruction. To avoid discontinuities during the switching, it is proposed that for a transient frame immediately after a switching, a sequence of windows (h0(n),h1(n),h2(n)) is provided for the forward and the inverse MDCTs. The windows of the window sequence are shorter than windows of the first type. The window sequence splits the spectrum of a respective first coding frame into nearly uncorrelated spectral components when used as basis for forward MDCTs, and the second half of the last window (h2(n)) of the sequence of windows is identical to the second half of a window of the first type.

Abstract: Prevalence detection is advantageously applied to the result of specific spectral discrimination to adaptively determine prevalent frequencies existing within an audio signal containing speech. Prevalent frequencies in this audio signal so isolated are attenuated in a highly selective manner, thus reducing the masking potential of pervasive resonances and obfuscative energy within the speech itself over low energy language-imparting speech elements.

Abstract: A transient noise removal system removes or dampens undesired transients from speech. When the transient noise removal system receives a speech frame, the system performs a wavelet transform analysis. The speech frame may be represented by one or more wavelet coefficients across one or more wavelet levels. For a given wavelet level, the transient noise-removal system may determine a wavelet threshold. The transient noise removal system may compare the threshold corresponding to a wavelet level to the wavelet coefficients within that level. The transient noise removal system may attenuate each wavelet coefficient based on a comparison to a threshold.

Abstract: The present invention is a detection system of a segment including specific sound signal which detects a segment in a stored sound signal similar to a reference sound signal, including: a reference signal spectrogram division portion which divides a reference signal spectrogram into spectrograms of small-regions; a small-region reference signal spectrogram coding portion which encodes the small-region reference signal spectrogram to a reference signal small-region code; a small-region stored signal spectrogram coding portion which encodes a small-region stored signal spectrogram to a stored signal small-region code; a similar small-region spectrogram detection portion which detects a small-region spectrogram similar to the small-region reference signal spectrograms based on a degree of similarity of a code; and a degree of segment similarity calculation portion which uses a degree of small-region similarity and calculates a degree of similarity between the segment of the stored signal and the reference signal

Abstract: The invention relates to a method for supporting an encoding of an audio signal, wherein at least one section of the audio signal is to be encoded with a coding model that allows the use of different coding frame lengths. In order to enable a simple selection of the respectively best suited coding frame length, it is proposed that at least one control parameter is determined based on signal characteristics of the audio signal. The control parameter is then used for limiting the options of possible coding frame lengths for the at least one section. The invention relates equally to a module 10,11 in which this method is implemented, to a device 1 and a system comprising such a module 10,11, and to a software program product including a software code for realizing the proposed method.

Abstract: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.

Abstract: Coding systems that provide a perceptually improved approximation of the short-term characteristics of speech signals compared to typical coding techniques such as linear predictive analysis while maintaining enhanced coding efficiency. The invention advantageously employs a non-linear transformation and/or a spectral warping process to enhance particular short-term spectral characteristic information for respective voiced intervals of a speech signal. The non-linear transformed and/or warped spectral characteristic information is then coded, such as by linear predictive analysis to produce a corresponding coded speech signal. The use of the non-linear transformation and/or spectral warping operation of the particular spectral information advantageously causes more coding resources to be used for those spectral components that contribute greater to the perceptible quality of the corresponding synthesized speech.