Abstract: A filter apparatus for filtering a time domain input signal to obtain a time domain output signal, which is a representation of the time domain input signal filtered using a filter characteristic having an non-uniform amplitude/frequency characteristic, comprises a complex analysis filter bank for generating a plurality of complex subband signals from the time domain input signals, a plurality of intermediate filters, wherein at least one of the intermediate filters of the plurality of the intermediate filters has a non-uniform amplitude/frequency characteristic, wherein the plurality of intermediate filters have a shorter impulse response compared to an impulse response of a filter having the filter characteristic, and wherein the non-uniform amplitude/frequency characteristics of the plurality of intermediate filters together represent the non-uniform filter characteristic, and a complex synthesis filter bank for synthesizing the output of the intermediate filters to obtain the time domain output signal.

Abstract: The present invention relates to creating a phonetic index of phonemes from an audio segment that includes speech content from multiple sources. The phonemes in the phonetic index are directly or indirectly associated with the corresponding source of the speech from which the phonemes were derived. By associating the phonemes with a corresponding source, the phonetic index of speech content from multiple sources may be searched based on phonetic content as well as the corresponding source.

Abstract: According to one embodiment, an apparatus for supporting reading of a document includes a model storage unit, a document acquisition unit, a feature information extraction, and an utterance style estimation unit. The model storage unit is configured to store a model which has trained a correspondence relationship between first feature information and an utterance style. The first feature information is extracted from a plurality of sentences in a training document. The document acquisition unit is configured to acquire a document to be read. The feature information extraction unit is configured to extract second feature information from each sentence in the document to be read. The utterance style estimation unit is configured to compare the second feature information of a plurality of sentences in the document to be read with the model, and to estimate an utterance style of the each sentence of the document to be read.

Abstract: A scalable encoding device capable of reducing an encoding rate to reduce a circuit scale while preventing sound quality deterioration of a decoded signal. An extension layer is coarsely divided into a system for processing a first channel and a system for processing a second channel. A sound source predictor for processing the first channel predicts a drive sound source signal of the first channel from a drive sound source signal of a monaural signal, and outputs the predicted drive sound source signal through a multiplier to a first CELP encoder. A sound source predictor for processing the second channel predicts the drive sound source signal of the second channel from the drive sound source signal of the monaural signal and the output from the first CELP encoder, and outputs the predicted drive sound source signal through a multiplier to a second CELP encoder.

Abstract: Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.

Abstract: In one embodiment, a method of generating a highband excitation signal includes harmonically extending the spectrum of a signal that is based on a lowband excitation signal; calculating a time-domain envelope of a signal that is based on the lowband excitation signal; and modulating a noise signal according to the time-domain envelope. The method also includes combining (A) a harmonically extended signal based on a result of the harmonically extending and (B) a modulated noise signal based on a result of the modulating. In this method, the highband excitation signal is based on a result of the combining.

Abstract: A microphone-array-based speech recognition system using a blind source separation (BBS) and a target speech extraction method in the system are provided. The speech recognition system performs an independent component analysis (ICA) to separate mixed signals input through a plurality of microphone into sound-source signals, extracts one target speech spoken for speech recognition from the separated sound-source signals by using a Gaussian mixture model (GMM) or a hidden Markov Model (HMM), and automatically recognizes a desired speech from the extracted target speech. Accordingly, it is possible to obtain a high speech recognition rate even in a noise environment.

Abstract: A method of communicating speech comprising time-warping a residual low band speech signal to an expanded or compressed version of the residual low band speech signal, time-warping a high band speech signal to an expanded or compressed version of the high band speech signal, and merging the time-warped low band and high band speech signals to give an entire time-warped speech signal. In the low band, the residual low band speech signal is synthesized after time-warping of the residual low band signal while in the high band, an unwarped high band signal is synthesized before time-warping of the high band speech signal. The method may further comprise classifying speech segments and encoding the speech segments. The encoding of the speech segments may be one of code-excited linear prediction, noise-excited linear prediction or ? frame (silence) coding.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

Abstract: A down-sampler 101 down-samples the sampling rate of an input signal from sampling rate FH to sampling rate FL. A base layer coder 102 encodes the sampling rate FL acoustic signal. A local decoder 103 decodes coding information output from base layer coder 102. An up-sampler 104 raises the sampling rate of the decoded signal to FH. A subtracter 106 subtracts the decoded signal from the sampling rate FH acoustic signal. An enhancement layer coder 107 encodes the signal output from subtracter 106 using a decoding result parameter output from local decoder 103.

Abstract: An emotion recognition system for assessing human emotional behavior from communication by a speaker includes a processing system configured to receive signals representative of the verbal and/or non-verbal communication. The processing system derives signal features from the received signals. The processing system is further configured to implement at least one intermediate mapping between the signal features and one or more elements of an emotional ontology in order to perform an emotion recognition decision. The emotional ontology provides a gradient representation of the human emotional behavior.

Abstract: A scalable decoder capable of avoiding deterioration in subjective quality of a listener. The scalable decoder for decoding core layer encoding data and extension layer encoding data including an extension layer gain coefficient, wherein a voice analysis section detects variation in power of a core layer decoding voice signal being obtained from the core layer encoding data, a gain attenuation rate calculating section (140) sets the attenuation intensity variable depending on variation in power, and a gain attenuation section (143) attenuates the extension layer gain coefficient in a second period preceding a first period according to a set attenuation intensity when extension layer encoding data in the first period is missing, thus interpolating the extension layer gain coefficient in the first period.

Abstract: A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution.

Abstract: Disclosed is a stereo audio encoding device capable of improving a spatial image of a decoded audio in stereo audio encoding. In this device, an original cross correlation calculation unit (101) calculates a mutual relationship coefficient (C1) between the original L channel signal and the original R channel signal. A stereo audio reconfiguration unit (104) subjects the inputted L channel signal and the R channel signal to encoding and decoding so as to generate an L channel reconfigured signal (L?) and an R channel reconfigured signal (R?). A reconfiguration cross correlation calculation unit (105) calculates a cross correlation coefficient (C2) between the L channel reconfigured signal (L?) and the R channel reconfigured signal (R?). A cross correlation comparison unit (106) calculates and outputs a comparison result &agr; between the cross correlation coefficient (C1) and the cross correlation coefficient (C2).

Abstract: An apparatus and method for processing an encoded signal are discussed. The method includes: if a coding type of an audio signal is a mixed signal coding type, extracting spectral data and a linear predictive coefficient from the audio signal; generating a residual signal for the linear prediction by performing an inverse frequency conversion on the spectral data; reconstructing the audio signal by performing a linear prediction coding on the linear predictive coefficient and the residual signal; and reconstructing a high frequency region signal using an extension base signal corresponding to a partial region of the reconstructed audio signal and band extension information.

Abstract: Disclosed is a method and an apparatus for estimating noise included in a sound signal during sound signal processing. The method includes estimating harmonics components in a frame of an input sound signal; using the estimated harmonics components, computing a Voice Presence Probability (VPP) on the frame of the input sound signal; determining a weight of an equation necessary to estimate a noise spectrum, depending on the computed VPP; and using the determined weight and the equation necessary to estimate a noise spectrum, estimating the noise spectrum, and updating the noise spectrum.

Abstract: An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

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 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: A sound encoder having an improved quantization performance while suppressing an increase of the bit rate to a lowest level. In a second layer encoder, a standard deviation calculator calculates a standard deviation ?c of a first layer decoding spectrum after decoding a scale factor ratio multiplication and outputs the standard deviation ?c to a selector. The selector selects a linear transform function as a function for a nonlinear transform of a residual spectrum according to the standard deviation ?c A nonlinear transform function selects one of prepared nonlinear transform functions #1 to #N according to a result of the selection by the selector, and outputs the selected one to an inverse transformer. The inverse transformer subjects an inverse transform (expansion) to a residual spectrum candidate that is stored in a residual spectrum code book using the nonlinear transform function outputted from the nonlinear transform function and outputs the result to an adder.

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: 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: A voice recognition terminal executes a local voice recognition process and utilizes an external center voice recognition process. The terminal includes: a voice message synthesizing element for synthesizing at least one of a voice message to be output from a speaker according to the external center voice recognition process and a voice message to be output from the speaker according to the local voice recognition process so as to distinguish between characteristics of the voice message to be output from the speaker according to the external center voice recognition process and characteristics of the voice message to be output from the speaker according to the local voice recognition process; and a voice output element for outputting a synthesized voice message from the speaker.

Abstract: In a device configurable to encode speech performing an open loop re-decision may comprise representing a speech signal by amplitude components and phase components for a current frame and a past frame. During the current frame, there may be an extraction of uncompressed amplitude components and uncompressed phase components. The amplitude components and the phase components from the past frame may then be retrieved. A set of features may be generated based on the uncompressed amplitude components from the current frame, the uncompressed phase components from the current frame, the amplitude components from the past frame, and the phase components from the past frame. The set of features may be checked as part of the open loop re-decision, and determining a final encoding decision based on the checking may be performed. The final encoding decision may be an encoding mode and/or encoding rate.

Abstract: According to an aspect of an embodiment, a method for regenerating an audio signal including a low frequency component and a high frequency component by decoding a coded data including a first coded data and a second coded data, the method comprising the steps of: generating the low frequency component; generating the high frequency component; determining whether the low frequency component has transient characteristics or not; generating a low frequency correction component by removing a stationary component when the audio signal has the transient characteristics; generating a corrected high frequency component by correcting the high-frequency component on the basis of the duration of the low frequency correction component when the audio signal has the transient characteristics; and regenerating the audio signal by synthesizing the low frequency component with the corrected high-frequency component.

Abstract: A speech processing apparatus includes a spectrum envelope extracting unit which extracts the spectrum envelope of an input speech signal, a spectrum envelope deforming unit which applies deformation to the spectrum envelope to generate a deformed spectrum envelope, a spectrum fine structure extracting unit which extracts the spectrum fine structure of the input speech signal, a deformed spectrum generating unit which generates a deformed spectrum by combining the deformed spectrum envelope with the spectrum fine structure, and a speech generating unit which generates an output speech signal on the basis of the deformed spectrum. This apparatus emits a disrupting sound based on the output speech signal to prevent a third party from eavesdropping on a conversation.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware.

Abstract: A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

Abstract: A scalable encoding device prevents sound quality deterioration of a decoded signal, reduces the encoding rate, and reduces the circuit size. The scalable encoding device includes a first layer encoder for generating a monaural signal by using a plurality of channel signals (L channel signal and R channel signal) constituting a stereo signal and encoding the monaural signal to generate a sound source parameter. The scalable encoding device also includes a second layer encoder for generating a first conversion signal by using the channel signal and the monaural signal, generating a synthesis signal by using the sound source parameter and the first conversion signal, and generating a second conversion coefficient index by using the synthesis signal and the first conversion signal.

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

Abstract: A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain.

Abstract: A temporal processing apparatus includes: a splitter splitting an audio signal, included in the sub-band domain, into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit generating a downmix signal by downmixing the direct signals; BPFs respectively generating a bandpass downmix signal and bandpass diffuse signals; normalization processing units respectively generating a normalized downmix signal and normalized diffuse signals; a scale computation processing unit computing, on a predetermined time slot basis, a scale factor indicating the magnitude of energy of the normalized downmix signal with respect to energy of the normalized diffuse signals; a calculating unit generating scale diffuse signals; a HPF generating high-pass diffuse signals; an adding unit generating addition signals; and a synthesis filter bank performing synthesis filter processing on the addition signals and transforming the addition signals into the time domains.

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: A scalable encoding apparatus capable of reducing the bit rates of encoded parameters and also capable of efficiently encoding audio signals in which a plurality of harmonic structures are coexistent. In the apparatus, an MDCT analyzer MDCT analyzes an audio signal for converting/encoding processes. A pitch frequency converter determines an inverse of a pitch period to calculate a pitch frequency. A selector selects spectra located at frequencies that are integral multiples of the pitch frequency, and a second layer encoder encodes the selected spectra.

Abstract: An audio decoding system performs frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The audio decoding system employs two different FLC methods: one designed to perform well for music, and the other designed to perform well for speech. When a frame is deemed lost, the audio decoding system analyzes a previously-decoded audio signal corresponding to previously-decoded frames of an audio bit-stream. Based on the results of the analysis, the lost frame is classified as either speech or music. Using this classification, other signal analysis, and knowledge of the employed FLC methods, the audio decoding system selects the appropriate FLC method which then performs FLC on the lost frame.

Abstract: Packet loss concealment (PLC) systems and methods are described that use time-warping to merge a concealment signal generated to replace one or more bad frames of an audio signal with a received signal representing one or more subsequent good frames of the audio signal in a manner that avoids signal discontinuity and audible artifacts resulting therefrom. Prediction-based PLC systems and methods are also described that use time-warping to conceal the loss of one or more frames containing a transition region in a manner that will not result in an audible artifact.

Abstract: A technique is described herein for updating a state of a decoder configured to decode a series of frames representing an encoded audio signal. In accordance with the technique, an output audio signal associated with a lost frame in the series of frames is synthesized. The decoder state is set to align with the synthesized output audio signal at a frame boundary. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with a first received frame after the lost frame in the series of frames, wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoder state is then reset based on the time lag.

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

Abstract: A contour for a syllable (or other speech segment) in a voice undergoing conversion is transformed. The transform of that contour is then used to identify one or more source syllable transforms in a codebook. Information regarding the context and/or linguistic features of the contour being converted can also be compared to similar information in the codebook when identifying an appropriate source transform. Once a codebook source transform is selected, an inverse transformation is performed on a corresponding codebook target transform to yield an output contour. The corresponding codebook target transform represents a target voice version of the same syllable represented by the selected codebook source transform. The output contour may be further processed to improve conversion quality.

Abstract: An embodiment includes a method that includes receiving data through a non-voice input. The method also includes translating the data into one or more numeric values. The method includes encoding the one or more numeric values into an audio stream, wherein the audio stream is to be transmitted over a transmission medium that is in use for voice communication.

Abstract: A downsampler 101 converts input data having a sampling rate 2*FH to a sampling rate 2*FL which is lower than the sampling rate 2*FH. A base layer coder 102 encodes the input data having the sampling rate 2*FL in predetermined base frame units. A local decoder 103 decodes a first coded code. An upsampler 104 increases the sampling rate of the decoded signal to 2*FH. A subtractor 106 subtracts the decoded signal from the input signal and regards the subtraction result as a residual signal. A frame divider 107 divides the residual signal into enhancement frames having a shorter time length than that of the base frame. An enhancement layer coder 108 encodes the residual signal divided into the enhancement frames and outputs a second coded code obtained by this coding to a multiplexer 109.

Abstract: The invention relates to a process for operating a voice dialog system and a voice dialog system which can be controlled over a telecommunications link by a communications terminal, a speech element transmitted by the communications terminal being received by a receiving unit of the voice dialog system and being analyzed for statement content in a processing unit, the speech element being filed in a memory assigned to the processing unit and after the telecommunications link is broken being analyzed by the processing unit.

Abstract: Provided is a method and apparatus for encoding a scalable wideband audio signal, the method including: filtering a voiced signal by performing linear prediction on the voiced signal and modulating the filtered signal; encoding the modulated signal in the time domain, and outputting a core layer encoding result of the voiced signal; subtracting a signal obtained by decoding the core layer encoding result from the modulated signal and outputting an error signal; and encoding the error signal and outputting an enhancement layer encoding result of the voiced signal.

Abstract: A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution.

Abstract: In a method and apparatus to conceal an error in an audio signal, when the current frame has no error and a past frame input prior to the current frame has an error, a parameter for the past frame is generated using a parameter for the current frame and a parameter of a frame out of frames input prior to the past frame and a previously stored parameter is updated with the generated parameter, thereby concealing an error of an audio signal without additional delay and preventing degradation in sound quality in a frame that is input after a frame having an error.

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.

Abstract: A speech coding apparatus comprises a repetition period pre-selecting unit for generating a plurality of candidates for the repetition period of a driving excitation source by multiplying the repetition period of an adaptive excitation source by a plurality of constant numbers, respectively, and for pre-selecting a predetermined number of candidates from all the candidates generated. A driving excitation source coding unit provides both excitation source location information and excitation source polarity information that minimize a coding distortion, for each of the predetermined number of candidates, and provides an evaluation value associated with the minimum coding distortion for each of the predetermined number of candidates.

Abstract: A method of speech encoding comprises generating a first synthesized speech signal from a first excitation signal, weighting the first synthesized speech signal using a first error weighting filter to generate a first weighted speech signal, generating a second synthesized speech signal from a second excitation signal, weighting the second synthesized speech signal using a second error weighting filter to generate a second weighted speech signal, and generating an error signal using the first weighted speech signal and the second weighted speech signal, wherein the first error weighting filter is different from the second error weighting filter. The method may further generate the error signal by weighting the speech signal using a third error weighting filter to generate a third weighted speech signal, and subtracting the first weighted speech signal and the second weighted speech signal from the third weighted speech signal to generate the error signal.