Abstract: The pitch extracting part generates a pitch waveform signal in a manner making the time interval of the pitch of the input audio sound data to be the same. After the number of samples in each region is made to be the same by the re-sampling part, the pitch waveform signal is changed into a subband data that express a time-varying-strength of a basic frequency composition and a higher harmonic composition by the subband analyzing part. The subband data are superimposed by a modulation wave composition that expresses attaching data of an attaching object by the data attaching part and is regarded as a bit stream to output through a nonlinear quantizing. A portion expressing the higher harmonic composition that is made corresponding to the audio sound expressed by this audio sound data in the subband data are deleted by the encoding part.

Abstract: A coding method is adapted to select different codebook search algorithms according to varied types of input signals. An encoder using the coding method is also provided. As appropriate search algorithms may be selected according to all possible structural features of the input signals, certain types of signals for which satisfactory results may be obtained through simple computations may match with search algorithms suitable for these signal types and having low computation complexities, so as to achieve better performance with fewer system resources. Meanwhile, other types of signals that need complicated computations may be processed by more sophisticated search algorithms, thereby ensuring the coding quality.

Abstract: Layered (embedded) code-excited linear prediction (CELP) speech encoders/decoders with adaptive plus algebraic codebooks applied in each layer with fixed codebook pulses of one layer used in higher layers. Pulse weightings emphasize lower layer pulses relative to the higher layer pulses.

Abstract: An apparatus for concealing a highband error in a spilt-band wideband voice codec in accordance with the present invention is disclosed. The apparatus includes: a lowband LPC coefficient extracting unit for extracting a lowband linear predictive coding (LPC) coefficient from a lowband voice signal passed by a lowband decoding unit; a highband excitation signal generating unit for generating a highband excitation signal based on the lowband voice signal and the lowband LPC coefficient; a highband LPC coefficient generating unit for generating a highband LPC coefficient based on the lowband LPC coefficient; a highband voice synthesizing unit for synthesizing a highband voice signal based on the highband excitation signal and the highband LPC coefficient; and a high pass filtering unit for removing a lowband component of the synthesized highband voice signal by the highband voice synthesis unit and generating the synthesized highband voice signal.

Abstract: A method for performing a search of a codebook is provided. The codebook includes a plurality of tracks each having a plurality of even pulse positions. The method includes partitioning a codevector having a plurality of pulses into a first subset of pulses and a second subset of pulses. Each pulse is assignable to a pulse position in the codevector, and each pulse is associated with a shift bit for indicating an odd position. The method also includes performing a first search for determining a first set of possible pulse positions for the pulses in the codevector. The method further includes performing a second search for determining a second set of possible pulse positions for the pulses in the codevector. In addition, the method includes forming the codevector using the first and second sets of possible pulse positions.

Abstract: Provided is a voice encoding device which performs voice encoding by a fixed code book effectively using a bit. In the voice encoding device, a position/polarity calculation unit (205) in a search loop (204) calculates a pulse position and polarity by using values of yH and HH. Moreover, a correlation value/sound source power calculation unit (206) extracts the value of the pulse position calculated by the position/polarity calculation unit (205) using yH and HH and calculates the correlation value and the sound source power. A search loop (207) successively calculates a position, polarity, a correlation value, and a sound source power of other pulses by using the pulse position and the polarity calculated by the position/polarity calculation unit (205) and the correlation value and the sound source power calculated by the correlation value/sound source power calculation unit (206).

Abstract: Embodiments of methods and means for correcting auto-correlated wireless signal samples are provided. Such embodiments include isolating and subtracting an interference vector from auto-correlated signal samples so that a corrected signal sample data set is derived. The corrected signal samples are then used in detecting and identifying symbols within the original wireless signal. Reliable and expeditious wireless communications can be achieved in accordance with the present embodiments.

Abstract: A system and method for detection of rate determination algorithm errors in variable rate communications system receivers. The disclosed embodiments prevent rate determination algorithm errors from causing audible artifacts such as screeches or beeps. The disclosed system and method detects frames with incorrectly determined data rates and performs frame erasure processing and/or memory state clean up to prevent propagation of distortion across multiple frames. Frames with incorrectly determined data rates are detected by checking illegal rate transitions, reserved bits, validating unused filter type bit combinations and analyzing relationships between fixed code-book gains and linear prediction coefficient gains.

Abstract: A code excited linear prediction speech decoder includes an adaptive codebook configured to generate an adaptive code vector. The decoder also includes a random codebook configured to generate a random code vector. The decoder also includes a synthesis filter that receives a signal based on said adaptive code vector and said random code vector, and that is configured to perform linear prediction coefficient synthesis on said signal. The random codebook includes a pulse vector providing system configured to provide a pulse vector having a signed unit pulse. The random codebook also includes a comparing system configured to compare a value of adaptive codebook gain with a preset threshold value. The random codebook further includes a selecting system configured to select a dispersion pattern from a plurality of dispersion patterns stored in a memory in accordance with a result of said comparison.

Abstract: A decoder for code excited LP encoded frames with both adaptive and fixed codebooks; erased frame concealment uses repetitive excitation plus a smoothing of pitch gain in the next good frame, plus multilevel voicing classification with multiple thresholds of correlations determining linear interpolated adaptive and fixed codebook excitation contributions.

Abstract: Square sum calculator 603 calculates a square sum of evolution in smoothed quantized LSP parameter for each order. A first dynamic parameter is thereby obtained. Square sum calculator 605 calculates a square sum using a square value of each order. The square sum is a second dynamic parameter. Maximum value calculator 606 selects a maximum value from among square values for each order. The maximum value is a third dynamic parameter. The first to third dynamic parameters are output to mode determiner 607, which determines a speech mode by judging the parameters with respective thresholds to output mode information.

Abstract: A stochastic codebook associates a pulse position of a predetermined channel with a pulse position of another channel, searches for a pulse position by means of a predetermined algorithm, and outputs a code combining a found pulse position with a polarity code to an excitation vector creation section as a stochastic excitation vector code. By this means, it is possible to secure variations so that there are no positions where there is no pulse at all while achieving a reduction of the number of bits used when coding stochastic codebook pulses in order to attain a lower bit rate.

Abstract: The invention relates to compressive transcoding between pulse coders using multipulse dictionaries in which each pulse occupies a position marked by an index. For each current pulse position supplied by a first coder, a neighborhood (Vge, Vde) is formed around that position. As a function of the pulse positions accepted by the second coder, pulse positions are selected in an ensemble constituted by a union of the neighborhoods. The second coder finally receives this selection (sj), involving a number of pulse positions smaller than the total number of pulse positions in the dictionary of the second coder.

Abstract: An electronic circuit includes storage circuitry and a speech coder coupled with the storage circuitry to have a codebook with sets of track location numbers for respective pulses, the speech coder operable to identify a group of track location numbers in the codebook substantially equally spaced from each other by a pitch lag amount, and make a selection from the group of track location numbers of a selected track location number. Other electronic circuits, processes, methods, devices and systems are disclosed and claimed.

Abstract: Provided is a voice encoding device for acquiring a satisfactory sound quality by making sufficient use of a tendency according to the noisiness or noiselessness of an input signal to be encoded. In this voice encoding device, a weight adding unit (206) in a searching loop (204) of a fixed code note searching unit (202) uses a function calculated from a code vector synthesized with a target to be encoded and spectrum enveloping information, as a calculated value to become the searching reference of the code vector stored in a fixed code note, and adds the weight according to the pulse number to form the code vector, to that calculated value.

Abstract: A dispersed vector generator used for a speech encoder or a speech decoder includes a pulse vector provider that provides a pulse vector having a signed unit pulse on one element of a vector axis. A dispersion pattern determiner determines a dispersion pattern of a set of waveforms defined before a start of encoding or decoding. A dispersed vector generator convolutes the pulse vector and the determined dispersion pattern to generate a dispersed vector. A length of the waveforms is shorter than a length of a sub-frame.

Abstract: An apparatus sets a pitch cycle search object in pitch cycle search processing for searching for a pitch cycle included in a linear predictive residual on a per subframe basis. A pitch cycle indicator of the apparatus sequentially output spitch cycle candidates within a predetermined pitch cycle search range at integral accuracy. A memory stores an integral component of a pitch cycle selected in pitch cycle search processing of a previous subframe. An adaptive sound source vector generator sets, as the pitch cycle search object in pitch cycle search processing in a processing subframe section, a group of candidates comprising a group of integral-accuracy pitch cycle candidates output from the pitch cycle indicator and a group of fractional-accuracy pitch cycle search candidates that cover a pitch cycle near an integral component of the pitch cycle read from the previous subframe integral pitch cycle memory using fractional accuracy.

Abstract: An embodiment of the invention improves upon the International Telecommunication Union's ITU-T G.729 Annex B comfort noise generation algorithm by reducing the computational complexity of the comfort noise generation algorithm. The computational complexity is reduced by reusing pre-computed random Gaussian noise samples for each non active voice frame versus calculating new random Gaussian noise samples for each non active voice frame as described by Annex B.

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

Abstract: Provided are a flexible bit rate code vector generation method and a wideband vocoder employing the same. This invention implements a flexible bit rate by getting three code vectors which are composed of 24, 16, and 8 pulses, at a time in a search process, through improvement of an algebraic codebook search process in a wideband AMR-WB vocoder. The method includes the steps of: performing a preprocess, wherein the preprocess divides a sub-frame by tracks and decides a pulse position having a maximum value in each track; among a plurality of pulses to be searched, fixing a same number of pulses as the tracks to the position with the maximum value of each track sequentially, and searching optimal positions having a minimum error with a target signal by combining two pulses in two consecutive tracks for the remaining pulses; and creating a code vector with flexible bit rate.

Abstract: A method for transcoding a bit stream encoded according to a linear predictive coding (LPC) standard to a bit stream encoded according to a mixed-excitation linear prediction (MELP) standard, including: decoding a bit stream into a first set of vocoder parameters compatible with the LPC standard; transforming the first set of vocoder parameters into a second set of vocoder parameters compatible with the MELP standard without converting the first set of vocoder parameters to an analog or digital waveform representation; and encoding the second set of vocoder parameters into a bit stream compatible with the MELP vocoder standard.

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 (101) that generates a pulse excitation vector; a convolution operation section (151) that convolutes an impulse response of an 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 (152) that generates a Toeplitz-type convolution matrix by means of the second impulse response vector; and a convolution operation section (153) that convolutes the matrix generated by matrix generating section (152) into the pulse excitation vector generated by algebraic codebook (101).

Abstract: A method of processing a communication includes determining an estimated excitation energy component of a subframe of a coded frame. A filter energy component of the subframe is also estimated. Determining an estimated energy of the subframe is based upon the estimated excitation energy component and the estimated filter energy component. This technique allows for estimating frame energy of a communication such as a voice communication without having to fully decode the communication.

Abstract: When an error is detected in coded data in the current frame, data separation section 201 separates the data into coding parameters first. Then, mode information decoding section 202 outputs decoding mode information in the previous frame and uses this as the mode information of the current frame. Furthermore, using the lag parameter code and gain parameter code of the current frame obtained at data separation section 201 and the mode information, lag parameter decoding section 204 and gain parameter decoding section 205 adaptively calculate a lag parameter and gain parameter to be used in the current frame according to the mode information.

Abstract: A code excited linear prediction speech decoder is provided. An adaptive codebook generates an adaptive code vector. A random codebook generates a random code vector. A synthesis filter receives a signal based on the adaptive code vector and the random code vector, and performs linear prediction coefficient synthesis on the signal. The random codebook includes a pulse vector provider that provides a pulse vector having a signed unit pulse, a comparator that compares a value of adaptive codebook gain with a preset threshold value, a selector that selects a dispersion pattern from a plurality of dispersion patterns stored in a memory in accordance with a result of the comparison, and a generator that generates the dispersed vector by convoluting the pulse vector and the selected dispersion pattern.

Abstract: A method and apparatus for the variable rate coding of a speech signal. An input speech signal is classified and an appropriate coding mode is selected based on this classification. For each classification, the coding mode that achieves the lowest bit rate with an acceptable quality of speech reproduction is selected. Low average bit rates are achieved by only employing high fidelity modes (i.e., high bit rate, broadly applicable to different types of speech) during portions of the speech where this fidelity is required for acceptable output. Lower bit rate modes are used during portions of speech where these modes produce acceptable output. Input speech signal is classified into active and inactive regions. Active regions are further classified into voiced, unvoiced, and transient regions. Various coding modes are applied to active speech, depending upon the required level of fidelity. Coding modes may be utilized according to the strengths and weaknesses of each particular mode.

Abstract: Provided are a combined, fixed codebook searching method and apparatus used in a code excited linear prediction (CELP) speech codec.

Abstract: A low-bit-rate coding technique for unvoiced segments of speech, without loss of quality compared to the conventional Code Excited Linear Prediction (CELP) method operating at a much higher bit rate. A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter. These gains are then quantized and applied to a randomly generated sparse excitation. The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal. Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance.

Abstract: A pulse allocating method capable of coding stereophonic voice signals efficiently. In the fixed code note retrievals (ST21 to ST25) of this pulse allocating method, for individual subframes, the stereophonic voice signals are compared (ST21) to judge similarity between channels, and are judged (ST22) 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 (ST23). Pulse retrievals are executed (ST24) to determine the pulse positions for the individual channels, so that the pulses determined at ST24 are coded (ST25).

Abstract: A method and apparatus for identifying a speaker within a captured audio signal from a collection of known speakers. The method and apparatus receive or generate voice representations for each known speakers and tag the representations according to meta data related to the known speaker or to the voice. The representations are grouped into one or more groups according to the indices. When a voice to be recognized is introduced, characteristics are determined according to which the groups are prioritized, so that the representations participating only in part of the groups are matched against the o voice to be identified, thus reducing identification time and improving the statistical significance.

Abstract: There is provided a method of decoding speech data generated from a speech signal. The method comprises receiving the speech data having at least one main pulse in a subframe of the speech data; generating a first predicted pulse, based on the at least one main pulse, on one side of the main pulse in the subframe of the speech data, wherein the first predicted pulse has a lower gain than the main pulse; generating a second predicted pulse, as a mirror image of the first predicted pulse on a reverse time scale, on the other side of the main pulse in the subframe of the speech data; reconstructing the speech signal using the at least one main pulse, the first predicted pulse and the second predicted pulse.

Abstract: Methods, systems, and devices for speech transduction are disclosed. One aspect of the invention involves a computer-implemented method in which a computer receives far-field acoustic data acquired by one or more microphones. The far-field acoustic data are analyzed. The far-field acoustic data are modified to reduce characteristics of the far-field acoustic data that are incompatible with human speech characteristics of near-field acoustic data.

Abstract: A first determiner 121 tentatively determines whether the current processing unit represents a stationary noise period, based on stationary properties of a decoded signal. Based on the tentative determination result and a determination result of the periodicity of the decoded signal, a second determiner 124 determines whether the current processing unit represents a stationary noise period, thereby distinguishing a decoded signal including a stationary speech signal such as a stationary vowel from stationary noise and correctly identifying the stationary noise period.

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: An automated voice message or caller prioritization system that extracts words, prosody, and/or metadata from a voice input. The data extracted is classified with a statistical classifier into groups of interest. These groups could indicate the likelihood that a call is urgent versus nonurgent, from someone the user knows well versus someone that the user only knows casually or not at all, from someone using a mobile phone versus a landline, or a business call versus a personal calls. The system then can determine an action based on results of the groups, including the display of likely category labels on the message. Call handling and display actions can be defined by user preferences.

Abstract: A transcoder for use between speech codecs using different Code-Excited Linear Prediction (CELP) type and a method therefor are disclosed. The transcoder includes a decoding unit of an input CELP codec, a transcoding filter, a transcoding filter design unit, and an encoding unit of an output CELP codec. By substituting a post-filter and a perceptual weighting filter of a prior art with one transcoding filter, the calculation amount of the transcoder is reduced, and speech quality decoded at a receiving end is improved.

Abstract: The present invention relates to a data processing apparatus capable of obtaining high-quality sound data. A tap generation section 121 generates a prediction tap used for a process in a prediction section 125 by extracting decoded speech data in a predetermined positional relationship with subject data of interest within the decoded speech data such that coded data is decoded by a CELP method and by extracting an I code located in a subframe according to a position of the subject data in the subject subframe. Similarly to the tap generation section 122, a tap generation section 122 generates a class tap used for a process in a classification section 123. The classification section 123 performs classification on the basis of the class tap, and a coefficient memory 124 outputs a tap coefficient corresponding to the classification result. The prediction section 125 performs a linear prediction computation by using the prediction tap and the tap coefficient and outputs high-quality decoded speech data.

Abstract: A device and a method for estimating an open-loop pitch in a general speech CODEC are disclosed. The open-loop pitch estimation device includes an autocorrelation function calculation unit which calculates a normalized autocorrelation function from a perceptual weighing filtered speech signal, a maximum autocorrelation function and lag estimation unit which estimates a maximum autocorrelation function and candidates for the maximum autocorrelation function, a pitch candidate decision unit which decides candidates for a pitch by using the ratio of the estimated maximum autocorrelation function to the candidates for the estimated maximum autocorrelation function, and lags of which values are smaller than a predetermined threshold value, and a pitch estimation unit which estimates a pitch between the candidates for a pitch and the lags corresponding to the estimated maximum autocorrelation function by using a pitch of a previous frame of the speech signal.

Abstract: A speech encoding apparatus calculates encoding distortion of a noise-like fixed code vector and multiplies the encoding distortion by a fixed weight corresponding to the noise-like degree of the noise-like fixed code vector, calculates encoding distortion of a non-noise-like fixed code vector and multiplies the encoding distortion by a fixed weight corresponding to the non-noise-like fixed code vector, and selects the fixed excitation code associated with multiplication result with a smaller value.

Abstract: A gain-constrained noise suppression for speech more precisely estimates noise, including during speech, to reduce musical noise artifacts introduced from noise suppression. The noise suppression operates by applying a spectral gain G(m, k) to each short-time spectrum value S(m, k) of a speech signal, where m is the frame number and k is the spectrum index. The spectrum values are grouped into frequency bins, and a noise characteristic estimated for each bin classified as a “noise bin.” An energy parameter is smoothed in both the time domain and the frequency domain to improve noise estimation per bin. The gain factors G(m, k) are calculated based on the current signal spectrum and the noise estimation, then smoothed before being applied to the signal spectral values S(m, k).

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: An audio encoding apparatus and the like are disclosed which can improve the sound quality of encoded audio signals even in a case of scalable CELP encoding the audio signals in sections that vary with time. In this apparatus, an enhancement layer extended adaptive codebook generating part (102) generates an extended adaptive codebook (d_enh_ext [i]) from both one frame of core layer drive sound source signals (exc_core[n]) received from a core layer CELP encoding part (101) and past enhancement layer drive sound source signals (exc_enh[n]) received from an adder (106), and further inputs the generated extended adaptive codebook (d_enh_ext [i]) to an enhancement layer extended adaptive codebook (103) for each of sub-frames. That is, the enhancement layer extended adaptive codebook generating part (102) updates the extended adaptive codebook (d_enh_ext[i]) for each of the sub-frames.

Abstract: A method and apparatus for performing multiple descriptive source coding in which a plurality of homogeneous encoders are advantageously employed in combination with a corresponding plurality of advantageously substantially identical decoders. In particular, diversity is provided to the multiple encoders by modifying the quantization process in at least one of the encoders such that the modified quantization process is based at least on a quantization error resulting from the quantization process of another one of the encoders. In this manner, diversity among the multiple bit streams is obtained, and in particular, the quality of a reconstructed signal based on a combination of multiple decoded bit streams at the receiver is advantageously superior to that based on any one of the decoded bit streams. In accordance with a first illustrative embodiment of the present invention, two Pulse Code Modulation (PCM) coders are employed.

Abstract: Audio/video programming content is made available to a receiver from a content provider, and meta data is made available to the receiver from a meta data provider. The meta data corresponds to the programming content, and identifies, for each of multiple portions of the programming content, an indicator of a likelihood that the portion is an exciting portion of the content. In one implementation, the meta data includes probabilities that segments of a baseball program are exciting, and is generated by analyzing the audio data of the baseball program for both excited speech and baseball hits. The meta data can then be used to generate a summary for the baseball program.

Abstract: First codebook and second codebook respectively have two subcodebooks, and in respective codebooks, addition sections obtain respective excitation vectors by adding sub-excitation vectors fetched from respective two subcodebooks. Addition section obtains an excitation sample by adding those excitation vectors. According to the aforementioned constitution, it is possible to store sub-excitation vectors with different characteristics in respective sub-codebooks. Therefore, it is possible to correspond to input signals with various characteristics, and achieve excellent sound qualities at the time of decoding.

Abstract: An excitation vector generator includes an input vector providing system that is capable of providing an input vector having at least one pulse, each pulse having a predetermined position and a respective polarity. A fixed waveform storage system is capable of storing at least one fixed waveform. An arranging system is capable of arranging the at least one fixed waveform in accordance with the position and the polarity of the at least one pulse.

Abstract: In order to recover the excitation energy quickly and keep the adaptive excitation contribution percentage in the entire excitation after bit-stream packet loss, the two excitation gains (Gp 305 and Gc 306) can be first transformed into the two other special parameters: one is the entire excitation energy and another is the energy ratio of the adaptive excitation contribution portion relative to the entire excitation energy. Then, the transformed parameters are quantized and sent to decoder. At the decoder side, the quantized parameters are transformed back to the original form of the gains (Gp 305 and Gc 306).

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 of significantly reducing error propagation due to voice packet loss, while still greatly profiting from long-term pitch prediction, is achieved by adaptively limiting the maximum value of the pitch gain for the first pitch cycle within one frame. A speech coding system for encoding a speech signal, wherein said a plurality of speech frames are classified into said a plurality of classes depending on if the first pitch cycle is included in one subframe or several subframes. The pitch gain is set to a value significantly smaller than 1 for the subframes covering first pitch cycle; wherein the pitch gain reduction is compensated by increasing the coded excitation codebook size or adding one more stage of excitation for the subframes covering the first pitch cycle.

Abstract: A system and method of processing sound signals are disclosed. In one embodiment, a speech coder applies a first sound signal enhancement process to a first part of a sound signal and applies a second sound signal enhancement process to a second part of the sound signal. The sound signal is then coded using the enhanced first part of the sound signal and the enhanced first part of the sound signal and the enhanced sound part of the sound signal. Examples of the portions of the sound signal that are separately processed include an excitation signal component and a spectral component of the sound signal.