Abstract: A method and apparatus for coding a quasi-periodic speech signal. The speech signal is represented by a residual signal generated by filtering the speech signal with a Linear Predictive Coding (LPC) analysis filter. The residual signal is encoded by extracting a prototype period from a current frame of the residual signal. A first set of parameters is calculated which describes how to modify a previous prototype period to approximate the current prototype period. One or more codevectors are selected which, when summed, approximate the error between the current prototype period and the modified previous prototype. A multi-stage codebook is used to encode this error signal. A second set of parameters describe these selected codevectors. The decoder synthesizes an output speech signal by reconstructing a current prototype period based on the first and second set of parameters, and the previous reconstructed prototype period.

Abstract: A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.

Abstract: A method of allocating vocoders during a simultaneous transfer of voice and data frames in a mobile communication system includes allocating a separate vocoder for processing the voice frame and the data frame. When a mobile communication receives a multiple requests for a simultaneous transfer of voice and data frames, the data frames are multiplexed and allocated to a single vocoder. Thus, the availability of the vocoder resources can be significantly increased.

Abstract: A method and apparatus for using coding scheme selection patterns in a predictive speech coder to reduce sensitivity to frame error conditions includes a speech coder configured to select from among various predictive coding modes. After a predefined number of speech frames have been predictively coded, the speech coder codes one frame with a nonpredictive coding mode or a mildly predictive coding mode. The predefined number of frames can be determined in advance from the subjective standpoint of a listener. The predefined number of frames may be varied periodically. An average coding bit rate may be maintained for the speech coder by ensuring that an average coding bit rate is maintained for each successive pattern, or group, of predictively coded speech frames including at least one nonpredictively coded or mildly predictively coded speech frame.

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: Compression of speech may be achieved through the adaptive generation of a compressed sound output. A first processing element may be used to characterize a first sound representation such that a first characterization result is produced. A comparison element may be provided to compare a first comparison input that is related to the first sound representation with a second comparison input that is related to the first characterization result. A determination may be made on whether further processing is desirable based on whether the first comparison result satisfies a first predetermined threshold criteria. Additionally, a second processing element may be included to characterize a second sound representation and to produce a second characterization result only if the first comparison result satisfies the first predetermined threshold. A compressed sound output is generated whose contents are determined based on at least the first comparison result.

Abstract: A method and system for reducing prediction error impulses using a gain average calculator, an impulse detector, a signal classifier decision means and a gain compensator wherein the compensated scaling of a quantizer is determined in a process of encoding/decoding a VBD type transmission by using a vectorial linear non-adaptive predicting type algorithm.

Abstract: A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a pluralitty of seeds This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM) thereby considerably reducing the memory capacity.

Abstract: The present invention intends to enhance a sound quality of a sound source generating portion in a CELP type voice encoding device and a CELP type voice decoding device. A pitch peak position of an adaptive code vector is obtained by a pitch peak position calculator 12, a window for emphasizing an amplitude of the pitch peak position is prepared by an amplitude emphasizing window generator 13, and an amplitude of a noise code vector corresponding to the pitch peak position is emphasized by an amplitude emphasizing window unit 16. Alternatively, pulse search positions are determined in such a manner that they become dense in a pitch peak position vicinity and coarse in the other portions. Based on the determined search positions, a pulse position searching is performed. Alternatively, the pitch peak position and pitch cycle information in the immediately previous sub-frame and the pitch cycle information in the present sub-frame are used to backward adapt and switch a sound source constitution.

Abstract: A data processing system for use in arrays includes a digital signal processor, a search accelerator unit and memory unit, the memory unit having a group storage locations that store the data entries of the matrix. The locations in the matrix are identified by the indices of the location. The access of the matrix by the digital processing unit typically includes an access to a series of locations at periodic intervals along a row or diagonal of the matrix. The series of data entries can include a sequence of non-neighboring matrix data entries. The search accelerator unit includes at least one pointer unit. The pointer unit in the search accelerator unit receives beginning array indices identifying the array entry. The pointer unit increments the array indices to provide the sequence of data entry indices for the matrix. The data entry array indices are converted to a series of memory location addresses.

Abstract: A distributed voice recognition system includes a digital signal processor (DSP), a nonvolatile storage medium, and a microprocessor. The DSP is configured to extract parameters from digitized input speech samples and provide the extracted parameters to the microprocessor. The nonvolatile storage medium contains a database of speech templates. The microprocessor is configured to read the contents of the nonvolatile storage medium, compare the parameters with the contents, and select a speech template based upon the comparison. The nonvolatile storage medium may be a flash memory. The DSP may be a vocoder. If the DSP is a vocoder, the parameters may be diagnostic data generated by the vocoder. The distributed voice recognition system may reside on an application specific integrated circuit (ASIC).

Abstract: A speech encoding comb codebook structure for providing good quality reproduced low bit-rate speech signals in a speech encoding system. The codebook structure requires minimal training, if any, and allows for reduced complexity and memory requirements. The codebook includes a first and at least one additional sub-codebooks, each having a plurality of code-vectors. The codebook may be randomly populated. All even elements may be set to zero in a first codebook, and all odd elements may be set to zero on a second codebook. The resulting comb codebook includes code-vector combination of the code-vectors from the sub-codebooks. In certain embodiments, the code-vectors of the sub-codebooks may contain zero valued elements. In other embodiments where the code-vectors of the sub-codebooks contain only non-zero elements, zero valued elements may be inserted in between the non-zero elements of the sub-codebooks during the forming of the resultant comb codebook.

Abstract: An apparatus and method for determining a speech-encoding rate in a variable rate vocoder are disclosed. A set of thresholds are computed based on background noise energy and its variation. A signal energy value of an input signal is computed, and a rate decision is made based on comparisons of the computed signal energy value with the computed thresholds. In one embodiment, a preliminary rate and a hangover interval are first computed based on the comparisons. The preliminary rate decision is then modified to take into account hangover constraints, a long term prediction gain and minimum and maximum rate constraints.

Abstract: A speech signal is encoded using code excited linear prediction for use in transmitting the speech signal to a receiver. The speech signal is sampled. A current sample of the speech signal is predicted based on in part a previous sample. An innovation sequence is determined based on in part a prediction error between the predicted current sample and the current sample of the speech signal. A code from each of a plurality of codebooks is selected. A combination of the selected codes is the determined innovation sequence. An index of the selected codes is identified and transmitted to the receiver. The transmitted index enables reconstruction of the speech signal at the receiver.

Abstract: A coding parameter control circuit 31 computes frame length from bit rate and coding delay, and provides the computed frame length data to a CELP coding circuit 32. On the basis of the computed frame length, the coding parameter control circuit 32 selects control parameters from a table, in which a plurality of control parameters for controlling the operation of the CELP coding circuit are set, on the basis of the bit rate, and provides the selected control parameters to the CELP coding circuit. The coding parameter control circuit provides the sub-frame length, and bit number distributed to the multi-pulse signal to the multi-pulse signal generation parameter setting circuit 33. The multi-pulse signal coding parameter setting circuit 33 computes pulse number of multi-pulse excitation signal, pulse position candidates of each pulse and candidate positions thereof from the sub-frame length and bit number of multi-pulse signal.

Abstract: A method and system of vocoding comprising filtering an input signal resulting in an excitation signal having at least one signal pulse translating the location of the signal pulse into one of a plurality of valid track locations in a plurality of signal pulse location references. Data is placed into an invalid track location in the signal pulse location references. The excitation signal having the signal pulse location references is transmitted for receipt by a receiving vocoder.

Abstract: Digital audio is generated and coded using a multi-state dynamical system such as cellular automata. The rules of evolution of the dynamical system and the initial configuration are the key control parameters determining the characteristics of the generated audio. The present invention may be utilized as the basis of an audio synthesizer and as an efficient means to compress audio data.

Abstract: A method for vocoding in an ALL IP network including one or more circuit networks, one or more radio access networks and one or more packet networks includes the steps of: determining if a first vocoding algorithm of a sending terminal is the same as a second vocoding algorithm of a destination terminal; if the first vocoding algorithm is the same as the second vocoding algorithm, bypassing voice data from the sending terminal and transmitting the bypassed voice data to the destination terminal; if the first vocoding algorithm is not the same as the second vocoding algorithm, determining if the sending terminal is a mobile terminal; if the sending terminal is the mobile terminal, at a first radio access network (RAN) gateway coupled to the sending mobile terminal, vocoding the voice data at a data rate of the circuit network to thereby generate first vocoded data and transmitting the first vocoded data to a second RAN gateway coupled to a destination mobile terminal; and at the second RAN gateway, vocoding th

Abstract: Automatic Speech Recognition (ASR) is achieved in wireless communications systems in which reliable ASR feature vector sequences are derived at a base station directly from digitally transmitted speech coder parameters, with no additional processing or signal modification required at the originating handset. No secondary channel need be provided for the transmission of ASR feature vectors. In operating on received speech coder parameters prior to conversion to a voice signal the present system and methods avoid the lossy conversion process and associated voice distortion. Since the received voice parameters are error protected during transmission they are received with greater accuracy. All, or a subset, of speech coding parameters, including, in appropriate cases, spectral envelope parameters, reflection coefficients, LSPs, LSFs, LPCs, LPCCs, and weighted LPCCs may be processed at a receiving base station or forwarded to another location for processing.

Abstract: The present invention relates to a low bit rate speech coding apparatus which performs coding on a speech signal for transmission, for example, in a mobile communication system. Excitation information is coded in multimode using both static and dynamic characteristics of quantized vocal tract parameters. Decoding includes postprocessing in multimode, thereby improving the quality of both unvoiced speech regions and stationary noise regions of the transmitted speech signal.

Abstract: A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.

Abstract: A speech encoding comb codebook structure for providing good quality reproduced low bit-rate speech signals in a speech encoding system. The codebook structure requires minimal training, if any, and allows for reduced complexity and memory requirements. The codebook includes a first and at least one additional sub-codebooks, each having a plurality of code-vectors. The codebook may be randomly populated. All even elements may be set to zero in a first codebook, and all odd elements may be set to zero on a second codebook. The resulting comb codebook includes code-vector combination of the code-vectors from the sub-codebooks. In certain embodiments, the code-vectors of the sub-codebooks may contain zero valued elements. In other embodiments where the code-vectors of the sub-codebooks contain only non-zero elements, zero valued elements may be inserted in between the non-zero elements of the sub-codebooks during the forming of the resultant comb codebook.

Abstract: A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.

Abstract: A method and apparatus is provided for matching a first sequence of patterns representative of a first signal with a second sequence of patterns representative of a second signal. The system uses a plurality of different pruning thresholds (th) to control the propagation of paths which represent possible matchings between a sequence of second signal patterns and a sequence of first signal patterns ending at the current first signal pattern. In particular, the pruning threshold used for a given path during the processing of a current first signal pattern depends upon the position, within the sequence of patterns representing the second signal, of the second signal pattern which is at the end of the given path.

Abstract: A voice signal coding apparatus includes: a voice status detector detecting whether an input signal divided at predetermined frame intervals is a voice or a non-voice signal; a linear predictive analyzer outputting a linear predictive parameter associated with the input signal; a voice sound source predicting circuit; a non-voice sound source predicting circuit including a random signal generator; and a switch controller selecting either the voice or non-voice sound source predicting circuit from the detection result of the voice status detector, wherein the random signal gain is set in accordance with a value obtained by suppressing by a predetermined factor the gain obtained when a non-voice input signal is coded by the voice sound predicting circuit.

Abstract: A method and apparatus for CELP-based to CELP-based vocoder packet translation. The apparatus includes a formant parameter translator and an excitation parameter translator. The formant parameter translator includes a model order converter and a time base converter. The method includes the steps of translating the formant filter coefficients of the input packet from the input CELP format to the output CELP format and translating the pitch and codebook parameters of the input speech packet from the input CELP format to the output CELP format. The step of translating the formant filter coefficients includes the steps of converting the model order of the formant filter coefficients from the model order of the input CELP format to the model order of the output CELP format and converting the time base of the resulting coefficients from the input CELP format time base to the output CELP format time base.

Abstract: A multipulse interpolative coder for transition speech frames includes an extractor configured to represent a first frame of transitional speech samples by a subset of the samples of the frame. The coder also includes an interpolator configured to interpolate the subset of samples and a subset of samples extracted from an earlier-received frame to synthesize other samples of the first frame that are not included in the subset. The subset of samples is further simplified by selecting a set of pulses from the subset and assigning zero values to unselected pulses. In the alternative, a portion of the unselected pulses may be quantized. The set of pulses may be the pulses having the greatest absolute amplitudes in the subset. In the alternative, the set of pulses may be the most perceptually significant pulses of the subset.

Abstract: Silence description coding for multi-rate speech coding systems that employ discontinued transmission. Speech coding systems include multi-rate speech codecs having an encoder and a decoder. The silence description coding is performed in either the encoder or the decoder of the multi-rate speech codec. It may also be performed in a distributed manner wherein it is performed partially in the encoder and partially in the decoder. The silence description coding is performed on a speech signal having a substantially non-speech-like characteristic. Voice activity detection classifies the speech signal as being either substantially speech-like or substantially non-speech-like. The silence description coding is selected from a plurality of coding modes. In certain embodiments of the invention, the silence description coding is a source coding mode that operates at a bit rate that fits within a bit rate budget as determined by all of the available source coding modes within the plurality of coding modes.

Abstract: A communication apparatus making use of speech vector coding scheme stores a plurality of predetermined code vectors and registered keyword data. A speech encoder codes an input speech keyword to produce coded keyword data by referring to the predetermined code vectors, after detecting a matching level between the registered keyword data and the coded keyword data, it is determined whether the coded keyword data is true or false by comparing the matching level with a predetermined criterion.

Abstract: A sound compression system adaptively switches codebooks in and out based on a calculation carried out with the output of the codebook. The system uses three separate codebooks: adaptive vector quantization codebook, real pitch codebook, and noise codebook. The perceptually-weighted filter is generated adaptively using the predictive coefficients from the current sub-frame.

Abstract: It is an objective of the present invention to provide an optimized method of selection of the encoding mode that provides rate efficient coding of the input speech. It is a second objective of the present invention to identify and provide a means for generating a set of parameters ideally suited for this operational mode selection. Third, it is an objective of the present invention to provide identification of two separate conditions that allow low rate coding with minimal sacrifice to quality. The two conditions are the coding of unvoiced speech and the coding of temporally masked speech. It is a fourth objective of the present invention to provide a method for dynamically adjusting the average output data rate of the speech coder with minimal impact on speech quality.

Abstract: A speech communication system using a code excited linear prediction speech decoder. The decoder using a first codebook containing a first digital value sequence selected from the set of binary values {0, 1}. The decoder also using a second codebook containing a second digital value sequence having values selected from the set of binary values {−1, 0}. The first digital value sequence and the second digital value sequence are combined to become a third digital value sequence having a set of ternary values from the set of {−1, 0, 1}.

Abstract: The spectral or pitch parameters of a speech signal are quantized, and impulse responses thereof are predicted by using a filter. An orthogonal transform is made of the speech signal, or a signal derived therefrom, or of the impulse responses or signals derived therefrom. The result of the orthogonal transform is entirely or partly quantized to obtain a plurality of pulses. More preferably, these pulses are retrieved recurrently by also using codevectors retrieved from a codebook or collectively quantizing their senses or amplitudes. This method optimizes speech signal coding.

Abstract: The present invention intends to enhance a sound quality of a sound source generating portion in a CELP type voice encoding device and a CELP type voice decoding device. A pitch peak position of an adaptive code vector is obtained by a pitch peak position calculator 12, a window for emphasizing an amplitude of the pitch peak position is prepared by an amplitude emphasizing window generator 13, and an amplitude of a noise code vector corresponding to the pitch peak position is emphasized by an amplitude emphasizing window unit 16. Alternatively, pulse search positions are determined in such a manner that they become dense in a pitch peak position vicinity and coarse in the other portions. Based on the determined search positions, a pulse position searching is performed. Alternatively, the pitch peak position and pitch cycle information in the immediately previous sub-frame and the pitch cycle information in the present sub-frame are used to backward adapt and switch a sound source constitution.

Abstract: A method and apparatus for eighth-rate random number generation for speech coders includes a random number generator configured to generate values of a first random variable. A lookup table is used to store values of a second random variable. The lookup table is addressed with the values of the first random variable. The second random variable is an inverse transform of a cumulative distribution function of the first random variable. An codec encodes input silence frames with the values of the first and second random variables, and regenerates the silence frames with the values of the first and second random variables. The speech coder may be an enhanced variable rate coder, and the silence frames may be encoded at eighth rate. The random variables are advantageously Gaussian random variables with values that are uniformly distributed between zero and one.

Abstract: A differential pulse-code modulation coder obtains an improved signal-to-noise ratio, with only a small increase in bit rate, by repetitive coding. In one aspect, the coder divides the input signal into frames, codes each frame repeatedly using different prediction coefficients or different quantizing step functions, and selects the coefficient or step function that produces the least quantization error. In another aspect, the coder repeats the coding of individual samples located in the outermost steps of the quantizing step function.

Abstract: Methods and apparatus implementing a technique for producing an audio output customized to a listener's hearing impairment through a digital telephone. A user initially sets user parameters to represent the user's hearing spectrum. In receiving a call, the digital telephone receives an input signal. The digital telephone adjusts the input signal according to the user parameters and generates an output signal based upon the adjusted input signal.

Abstract: A digital input symbol is transmitted to a receiver by determining one or more formant frequencies that correspond to the digital input symbol. In one embodiment, a pre-programmed addressable memory is used to map the set of possible digital input symbols onto a set of corresponding speech units, each comprising a superposition of one or more formant frequencies. A signal is then generated having the speech units. The signal is supplied for transmission over a voice channel. This may include supplying the signal to a voice coder prior to transmission. In another aspect of the invention, a forward error correction code (FEC) is determined for the digital input symbol, and the one or more speech units are modified as a function of the forward error correction code. In this way, the FEC may also be transmitted with the encoded input symbol. The modification may affect any of a number of attributes of the speech units, including a volume attribute and a pitch attribute.

Abstract: A transmission system wherein a speech signal is represented by a plurality of prediction parameters updated once per frame. Each frame comprises a plurality of sub-frames in which an excitation signal generated by a fixed codebook and an adaptive codebook is updated. In order to enhance the reconstructed speech quality the prediction coefficients are interpolated at the decoder by an LPC coefficient interpolator to obtain interpolated prediction coefficients for each sub-frame. According to the present invention the interpolation of the prediction coefficients is not based on the prediction coefficients used for transmission, such as reflection coefficients or Log Area Ratios, but on Line Spectral Frequencies. This reduces degradation of speech quality due to interpolation.

Abstract: A speech coder (400) for coding an information signal varies the codebook configuration based on parameters inherent in the information signal. The speech coder (400) requires no additional overhead for sending of mode parameters while allowing subframe resolution. The configurations vary not only for voicing level, but also for pitch period since different physiological traits yield different codebook configurations. A dispersion matrix (406) within the speech coder (400) facilitates a codebook search which is performed on vectors whose length can be less than a subframe length. Additionally, use of the dispersion matrix (406) allows the addition of random events for very slightly voiced speech which incurs little computational overhead but produces a rich excitation.

Abstract: To facilitate the reversion of a communication to tandem operation (200), a first transcoder (20), having previously changed back to tandem operation (206), inverts bits (208) of a double-encoded frame format that correspond to synchronisation bits in a non-tandem, single-encoded frame format to generate errors, in relation to the synchronisation bits, at a second transcoder (34). Upon detection (210) of a predetermined number of errors in the synchronisation bits during a predetermined time, the second transcoder (34) reverts to tandem operation, as shown in the flow diagram of FIG. 2.

Abstract: This invention relates to a method for improving performance of voice coder. A target signal is calculated for a window, and the optimal candidate codebooks and optimal candidate codebook gains from the target signal for the window are searched for all codebook indices and all codebook optimal gains. Target signals for a second subframe are then calculated from the target signal for the window, optimal candidate codebooks and optimal candidate codebook gains for the first subframe. The optimal candidate codebooks and optimal candidate codebook gains for the second subframe from the target signal for the second subframe are searched and the optimal candidate codebooks and optimal candidate codebook gains for the first subframe are searched.

Abstract: A method of quantizing harmonic amplitudes (FIG. 3), used in a speech encoder (10). The method compares variable dimension input vectors to fixed dimension codebook vectors, by first sampling each codebook vector so that it is converted to a vector having the same dimension as the input vector (FIG. 3, step 35). The resulting codebook vector is compared to the input vector (step 37). The difference (error) is weighted in favor of low frequency harmonics. Also, the weighting favors formant amplitudes so that they are quantized more accurately than formant nulls (FIG. 3, step 38; FIG. 5).

Abstract: The device and method of the invention receives a digital speech signal, which is processed by an Acoustic Processor to produce a Mel-Cepstrum Vector and Pitch. This is recalibrated and encoded. The encoded signal is transmitted over a narrow-band Channel, then decoded, split and recalibrated. From the split signals, one signal feeds a Statistical Processor which produces Recognized Text. Another signal feeds a Regenerator, which produces Regenerated Speech.

Abstract: Vocoder bypass is provided using in-band signaling. In preferred embodiments of the present invention, three signaling channels are arranged for transmission within the compressed speech. Each of the signaling channels are communicated at a preferred rate to permit fast, reliable detection of conditions indicating vocoder bypass mode of operation, to negotiate suitable vocoder type if necessary, and to synchronize and communicate compressed speech in a vocoder bypass mode of operation.

Abstract: Vocoder bypass is provided using a combination of out-of-band and in-band signaling. In preferred embodiments of the present invention, two signaling channels are arranged for transmission within the compressed speech. Each of the signaling channels are communicated at a preferred rate to permit fast, reliable detection of conditions indicating vocoder bypass mode of operation and to synchronize and communicate compressed speech in a vocoder bypass mode of operation.

Abstract: A coding system delivers a global data stream consisting of primary coded subband data streams from a primary subband coder bank, coding an input signal data stream, and secondary coded subband data streams from a secondary subband coder bank. The coding delay of the primary coder bank is smaller than that of the secondary coder bank. A filter bank receives the input signal data and generates signal streams in a plurality of subbands, which are coded by the respective coder of the primary subband coder bank, forming the primary streams. A bank of decoders receive and decode the respective coded primary subbank streams, which decoded subband signals are subtracted by a bank of subtractors from the corresponding original subband signals, which difference streams are input to the respective coder in a secondary subband coder bank. The secondary coder generates coded secondary subband data streams. A multiplexer interlaces the primary and the secondary coded subband data streams into a single global data stream.

Abstract: Adaptive speech coding includes receiving an original speech signal, performing on the original speech signal a current coding operation, and adapting the current coding operation in response to information used in the current coding operation. Adaptive speech decoding includes receiving coded information, performing a current decoding operation on the coded information, and adapting the current decoding operation in response to information used in the current decoding operation.

Abstract: In a CELP system, a coder and a decoder have identical codebooks, and the amount of data to be transmitted is compressed by transmission and reception of codebook indexes. Past excitation signals are stored in a memory and used as an adaptive codebook to improve the speech quality. The coder and the decoder each comprise memory means for storing index data for at least one frame, and means for generating an adaptive codebook afresh by initialization to zero for each frame when generating an excitation signal according to stored indexes.

Abstract: A method and system for controlling speech encoding in a communication system utilizes feedback information such as packet modification control data (154) sent from a communication link output controller, such as a network arbitor (142). The network arbitor (142) sends the packet modification control data (154) to a selected vocoder (146) to change the filter states of the selected vocoder (146) when the network arbitor (142) modifies an output speech packet communicated of a communication link (20), to facilitate improved convergence of a speech encoder and a speech decoder, such as a mobile subscriber unit.