Abstract: The invention proposed a Dual-Pulse Excitation Model; wherein two pulses of each pair of pulses are always adjacent each other. Only one position index for each pair of pulses needs to be sent to the decoder, which saves bits to code all pulse positions. The magnitudes of each pair of pulses have limited number of patterns. Because the two pulses are adjacent each other, each pair of pulses with different magnitudes can produce different high-pass and/or low-pass effect. Since the magnitudes have enough variation, it is possible to assign the candidate positions of each pair of pulses within a small range in order to save the searching complexity.

Abstract: A method and system for improving speech quality may include estimating at least one component of a distorted portion of a speech signal from at least one component of an undistorted portion of the speech signal and reinforcing the component of the distorted portion based on the estimating. The components may include the pitch, spectral envelope and spectral energy of the speech signal. The undistorted portion of the speech signal may be delayed and the components of the distorted portion may be interpolated from the components of a delayed undistorted portion and a current undistorted portion of the speech signal. The components of the distorted portion of the speech signal may be extrapolated from a current undistorted portion of the speech signal. Components of the distorted portion of the speech signal may be estimated from frequency bands other than the frequency band affected by the distortion.

Abstract: A layered code-excited linear prediction (CELP) encoder, an Adaptive Multirate Wideband (AMR-WB) encoder and methods of CELP encoding and decoding. In one embodiment, the encoder includes: (1) a core layer subencoder and (2) at least one enhancement layer subencoder, at least one of the core layer subencoder and the enhancement layer subencoder having first and second adaptive codebooks and configured to retrieve a pitch lag estimate from the second adaptive codebook and perform a closed-loop search of the first adaptive codebook based on the pitch lag estimate.

Abstract: A wideband speech coding apparatus which causes an input speech signal to be represented by spectrum parameters and an excitation signal. The apparatus includes a coding unit configured to select a plurality of pulses from given pulse position candidates, and to code the excitation signal with the selected pulses; an identification unit configured to identify whether the input speech signal is a wideband speech signal or a narrowband speech signal; and a control unit configured to control the coding unit to select a pulse position candidate having a time resolution which is set in advance in accordance with the wideband speech signal, when the identification unit identifies that the input speech signal is the wideband speech signal, and to control the coding unit to lower the time resolution of the pulse position candidate, when the identification unit identifies that the input speech signal is the narrowband speech signal.

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 encoder comprises a memory and the encoder is adapted to execute instructions stored in the memory comprising classifying speech segments and encoding speech segments, and the decoder comprises a memory and the decoder is adapted to execute instructions stored in the memory comprising time-warping a residual speech signal to an expanded or compressed version of the residual speech signal.

Abstract: A speech decoding method which generates an excitation signal and a synthesis filter from coded data and which obtains a speech signal based on the excitation signal and the synthesis filter. The method includes acquiring identification information used for determining whether the speech signal to be decoded is a narrowband signal or a wideband signal; and modifying the excitation signal based on the identification information by controlling strength or presence of emphasis of pitch periodicity with respect to the excitation signal generated from the coded data, so as to generate the speech signal by use of the modified excitation signal and the synthesis filter.

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: A behavior control system of a robot for learning a phoneme sequence includes a sound inputting device inputting a phoneme sequence, a sound signal learning unit operable to convert the phoneme sequence into a sound synthesis parameter and to learn or evaluate a relationship between a sound synthesis parameter of a phoneme sequence that is generated by the robot and a sound synthesis parameter used for sound imitation, and a sound synthesizer operable to generate a phoneme sequence based on the sound synthesis parameter obtained by the sound signal learning unit.

Abstract: Disclosed herein are methods, systems, and devices for improved audio, video, and data conferencing. The present invention provides a conferencing system comprising a plurality of endpoints communicating data including audio data and control data according to a communication protocol. A local conference endpoint may control or be controlled by a remote conference endpoint. Data comprising control signals may be exchanged between the local endpoint and remote endpoint via various communication protocols. In other embodiments, the present invention provides for improved bridge architecture for controlling functions of conference endpoints including controlling functions of the bridge.

Abstract: According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator.

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: Provided is an audio encoding device capable of improving performance of an adaptive codebook and improving quality of a decoded audio. In this audio encoding device, an adaptive codebook cuts out a vector specified by a comparator from adaptive code vectors stored in an internal buffer and outputs it to a filter and a switch. The filter performs a predetermined filtering process on the adaptive sound source signal and outputs the obtained adaptive code vector to the switch. According to an instruction from the comparator, the switch outputs the adaptive code vector directly output from the adaptive codebook to a adjuster when the adaptive codebook is searched and outputs the adaptive code vector output from the filter after being subjected to the filtering process to the gain adjuster when a fixed sound source is searched after the adaptive sound source search.

Abstract: A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal.

Abstract: A quantizer according to an embodiment is configured to quantize a smoothed value of an input value (e.g., a vector of line spectral frequencies) to produce a corresponding output value, where the smoothed value is based on a scale factor and a quantization error of a previous output value.

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 system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

Abstract: A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal.

Abstract: A pitch search method and device for digitally encoding a wideband signal, in particular but not exclusively a speech signal, in view of transmitting, or storing, and synthesizing this wideband sound signal. The new method and device which achieve efficient modeling of the harmonic structure of the speech spectrum uses several forms of low pass filters applied to a pitch codevector, the one yielding higher prediction gain (i.e. the lowest pitch prediction error) is selected and the associated pitch codebook parameters are forwarded.

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: The present invention provides a method, a computer-software-product and an apparatus for enabling a determination of speech related audio data within a record of digital audio data. The method comprises steps for extracting audio features from the record of digital audio data, for classifying one or more subsections of the record of digital audio data, and for marking at least a part of the record of digital audio data classified as speech. The classification of the digital audio data record is performed on the basis of the extracted audio features and with respect to at least one predetermined audio class.

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 pulse allocating method capable of coding stereophonic voice signals efficiently. In the fixed code note retrievals of this pulse allocating method, for individual subframes, the stereophonic voice signals are compared to judge similarity between channels, and are judged on their characteristics. On the basis of the similarity between the channels and the characteristics of the stereophonic signals, the pulse numbers to be allocated to the individual channels are determined. Pulse retrievals are executed to determine the pulse positions for the individual channels, so that the pulses determined are coded.

Abstract: Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

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

Abstract: A hardware device for analyzing an audio signal comprises a calculator for calculating a neural activity pattern over time resulting at nerve fibers of an ear model based on the audio signal and a processor for processing the neural activity pattern to obtain a sequence of time information as an analysis representation describing a temporal position of consecutive trajectories, wherein a trajectory includes activity impulses on different nerve fibers based on the same event in the audio signal. A two-dimensional representation of the neural activity pattern is gradually distorted over time, and it is recognized when an approximately straight line is contained in the distorted two-dimensional representation of the neural activity pattern over time. Accordingly, a time information belonging to the trajectory is provided.

Abstract: An audio encoding device for correcting a component having insufficient encoding capability in a core layer by an extended layer. A core layer encoder encodes an audio signal. An extended layer encoder encodes an encoding residual of the core layer encoder. A characteristic correction inverse filter arranged at a pre-stage of an LPC synthesis filter subjects the component having insufficient encoding capability in the core layer to an inverse characteristic correction process, and a characteristic correction filter arranged at a post-stage of the LPC synthesis filter performs a process for characteristic correction of the synthesis signal inputted from the LPC synthesis filter.

Abstract: The present invention provides a frame erasure concealment device and method that is based on reestimating gain parameters for a code excited linear prediction (CELP) coder. During operation, when a frame in a stream of received data is detected as being erased, the coding parameters, especially an adaptive codebook gain gp and a fixed codebook gain gc, of the erased and subsequent frames can be reestimated by a gain matching procedure. By using this technique with the IS-641 speech coder, it has been found that the present invention improves the speech quality under various channel conditions, compared with a conventional extrapolation-based concealment algorithm.

Abstract: An implementation of the present invention comprises a voice encoder and decoder method and system that uses voice excitation, eliminating the voice/unvoiced pitch tracking, and the first formant up to 2400 Hertz for synchronous and up to 1600 Hertz for asynchronous, does not use pulse code modulation encoding, but uses the zero crossings only of the first formant, frequency dividing by two and sampling at the formant frequency. The resulting combination uses half or less of the bit rate for excitation and the remainder for short-term spectrum analysis. The spectrum could be updated each 20 milliseconds using 49 bits for the spectrum frame and 49 bits for excitation and one frame bit for synchronous Asynchronous operation could be update at 21.25 milliseconds using 49 bits for the spectrum information and 34 bits for excitation with one bit for frame synchronization.

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: First encoded voice bits are transcoded into second encoded voice bits by dividing the first encoded voice bits into one or more received frames, with each received frame containing multiple ones of the first encoded voice bits. First parameter bits for at least one of the received frames are generated by applying error control decoding to one or more of the encoded voice bits contained in the received frame, speech parameters are computed from the first parameter bits, and the speech parameters are quantized to produce second parameter bits. Finally, a transmission frame is formed by applying error control encoding to one or more of the second parameter bits, and the transmission frame is included in the second encoded voice bits.

Abstract: A fixed codebook searching apparatus which slightly suppresses an increase in the operation amount, even if the filter applied to the excitation pulse has the characteristic that it cannot be represented by a lower triangular matrix and realizes a quasi-optimal fixed codebook search. This fixed codebook searching apparatus is provided with an algebraic codebook that generates a pulse excitation vector; a convolution operation section that convolutes an impulse response of auditory weighted synthesis filter into an impulse response vector that has a value at negative times, to generate a second impulse response vector that has a value at second negative times; a matrix generating section that generates a Toeplitz-type convolution matrix by means of the second impulse response vector; and a convolution operation section that convolutes the matrix generated by matrix generating section into the pulse excitation vector generated by algebraic codebook.

Abstract: An enhancement system extracts pitch from a processed speech signal. The system estimates the pitch of voiced speech by deriving filter coefficients of an adaptive filter and using the obtained filter coefficients to derive pitch. The pitch estimation may be enhanced by using various techniques to condition the input speech signal, such as spectral modification of the background noise and the speech signal, and/or reduction of the tonal noise from the speech signal.

Abstract: A fixed codebook searching apparatus which slightly suppresses an increase in the operation amount, even if the filter applied to the excitation pulse has the characteristic that it cannot be represented by a lower triangular matrix and realizes a quasi-optimal fixed codebook search. This fixed codebook searching apparatus is provided with an algebraic codebook that generates a pulse excitation vector; a convolution operation section that convolutes an impulse response of auditory weighted synthesis filter into an impulse response vector that has a value at negative times, to generate a second impulse response vector that has a value at second negative times; a matrix generating section that generates a Toeplitz-type convolution matrix by means of the second impulse response vector; and a convolution operation section that convolutes the matrix generated by matrix generating section into the pulse excitation vector generated by algebraic codebook.

Abstract: A fixed codebook search method includes: initializing a counter; searching for pulses and calculating the value of a cost function Qk; initializing the counter if the Qk value increases; increasing the value of the counter if the Qk value does not increase; judging whether the value of the counter is greater than the threshold value; continuing the search process if the value of the counter is not greater than the threshold value; and ending the whole search process if the value of the counter is greater than the threshold value. The present invention reduces the search count and improves the search efficiency.

Abstract: The invention relates to an audio encoder and decoder and methods for audio encoding and decoding. In a preferred encoder embodiment an audio signal is encoded by deterministic encoder means to form a first encoded signal part. A spectrum of the audio signal is determined and represented by an excitation pattern, i.e. spectral values corresponding to human auditory filters, as a second encoded signal part. A masking curve is also extracted based on the excitation pattern, thus improving encoding efficiency in terms of bit rate. In a preferred decoder the first encoded signal part is decoded by deterministic decoder means. A noise generator uses the decoded first signal part together with the second signal part, i.e. the excitation pattern for the original audio signal, to generate a noise signal. The noise signal is then added to the first decoded signal part to form an output audio signal. At the decoder side the masking curve is also extracted based on the second encoded signal part, i.e.

Abstract: A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal.

Abstract: There is disclosed a speech processing device in which prediction taps for finding prediction values of the speech of high sound quality are extracted from the synthesized sound obtained on affording linear prediction coefficients and residual signals, generated from a preset code, to a speech synthesis filter, speech of high sound quality being higher in sound quality than the synthesized sound, and in which the prediction taps are used along with preset tap coefficients to perform preset predictive calculations to find the prediction values of the speech of high sound quality. The speech of high sound quality is higher in sound quality than the synthesized sound.

Abstract: A system extends the high-frequency spectrum of a narrow band audio signal in the time domain. The system extends the harmonics of vowels by introducing a non linearity in a narrow band signal. Extended consonants are generated by a random-noise generator. The system differentiates the vowels from the consonants by exploiting predetermined features of a speech signal.

Abstract: A fixed codebook search method includes initializing a counter, searching for pulses and calculating the value of a cost function Qk, initializing the counter if the Qk value increases, increasing the value of the counter if the Qk value does not increase, judging whether the value of the counter is greater than the threshold value, continuing the search process if the value of the counter is not greater than the threshold value, and ending the whole search process if the value of the counter is greater than the threshold value.

Abstract: Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.

Abstract: A scalable encoding device for realizing scalable encoding by CELP encoding of a stereo sound signal and improving the encoding efficiency. In this device, an adder and a multiplier obtain an average of a first channel signal CH1 and a second channel signal CH2 as a monaural signal M. A CELP encoder for a monaural signal subjects the monaural signal M to CELP encoding, outputs the obtained encoded parameter to outside, and outputs a synthesized monaural signal M? synthesized by using the encoded parameter to a first channel signal encoder. By using the synthesized monaural signal M? and the second channel signal CH2, the first channel signal encoder subjects the first channel signal CH1 to CELP encoding to minimize the sum of the encoding distortion of the first channel signal CH1 and the encoding distortion of the second channel signal CH2.

Abstract: A system for monitoring conditions associated with an individual in a region includes at least one speech input transducer and speech processing software coupled thereto. Results of the speech processing can initiate communications with a displaced communications device such as a telephone or a computer to provide a source of feedback.

Abstract: Even when a combination of the stegonography technique and prediction encoding is applied to sound encoding, a sound encoder does not cause deterioration in quality of decoded signals. In the device, an encoding section (102) outputs an encoding code (I) to a bit embedding section (104). A function extension encoding section (103) generates an encoding code (J) for information required for extending functions of the sound encoder (100) and outputs it to the bit embedding section (104). The bit embedding section (104) embeds information on the encoding code (J) into a part of bits of the encoding code (I) and outputs the resultant encoding code (I?). A synchronization information generating section (106) generates synchronization information according to the encoding code (I?) after the bit embedding and outputs the synchronization information to the encoding section (102).

Abstract: A method and a device for source coding with a time advanced excitation signal. During an encoding process, a source data signal is first divided into consecutive blocks, then a first set of parameters related to a filter describing properties of a first block covering a first time period is extracted, followed by the extraction of a second set of parameters related to an excitation signal for said filter, where said second set of parameters is determined from and describing properties of both the first block and a second block following the first block within a second time period starting later than said first time period and extending outside said first time period.

Abstract: Disclosed are a vector quantization device and others capable of adaptively adjusting a vector space of a code vector for quantization of a second stage by using a quantization result of a first stage and improving the quantization accuracy.

Abstract: Provided are an apparatus and method for speech segment detection, and a system for speech recognition. The apparatus is equipped with a sound receiver and an image receiver and includes: a lip motion signal detector for detecting a motion region from image frames output from the image receiver, applying lip motion image feature information to the detected motion region, and detecting a lip motion signal; and a speech segment detector for detecting a speech segment using sound frames output from the sound receiver and the lip motion signal detected from the lip motion signal detector. Since lip motion image information is checked in a speech segment detection process, it is possible to prevent dynamic noise from being misrecognized as speech.

Abstract: An apparatus and method for encoding and decoding a voice signal. The apparatus includes an encoder configured to generate an output bitstream signal from an input voice signal. The output bitstream signal is associated with at least a first standard of a first plurality of CELP voice compression standards. Additionally, the apparatus includes a decoder configured to generate an output voice signal from an input bitstream signal. The input bitstream signal is associated with at least a first standard of a second plurality of CELP voice compression standards. The CELP encoder includes a plurality of codec-specific encoder modules. Additionally, the CELP encoder includes a plurality of generic encoder modules. The CELP decoder includes a plurality of codec-specific decoder modules. Additionally, the CELP decoder includes a plurality of generic decoder modules.

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.