Abstract: Coding of an audio signal (x) represented by a respective set of sampled signal values (x(t)) for each of a plurality of sequential time segments is disclosed. The sampled signal values are analyzed to determine one or more sinusoidal components for each of the plurality of sequential segments. The sinusoidal components are linked across a plurality of sequential segments to provide sinusoidal tracks, where each track comprises a number of frames. An encoded signal (AS) is generated, including sinusoidal codes (Cs) comprising a representation level (r) for each frame or including sinusoidal codes (Cs) where some of these codes comprise a phase (?), a frequency (?) and a quantization table (Q) for a given frame when the given frame is designated as a random-access frame. The invention allows random access in a track while avoiding long adaptation of the quantization accuracy in a quantizer and/or the need for a large bit stream while still maintaining improved audio quality.

Abstract: The present invention relates to a method and device for improving concealment of frame erasure caused by frames of an encoded sound signal erased during transmission from an encoder (106) to a decoder (110), and for accelerating recovery of the decoder after non erased frames of the encoded sound signal have been received. For that purpose, concealment/recovery parameters are determined in the encoder or decoder. When determined in the encoder (106), the concealment/recovery parameters are transmitted to the decoder (110). In the decoder, erasure frame concealment and decoder recovery is conducted in response to the concealment/recovery parameters. The concealment/recovery parameters may be selected from the group consisting of: a signal classification parameter, an energy information parameter and a phase information parameter.

Abstract: A voice processing apparatus, which processes a first voice signal, includes: an acoustic analysis part which analyzes a feature quantity of an input second voice signal; a reference range calculation part which calculates a reference range based on the feature quantity; a comparing part which compares the feature quantity and the reference range and outputs a comparison result; and a voice processing part which processes and outputs the input first voice signal based on the comparison result.

Abstract: A system and method for providing voice communications with desired characteristics based upon the intended recipient of a voice communication. An apparatus includes a list of dial strings associated with parties having desired voice communication characteristics. A dial string entered by a user and associated with an intended recipient is compared to a list of preferred dial strings to determine the characteristics of an encoded voice signal to be sent to the recipient. The apparatus can include a vocoder having different bit rate modes and a bit rate mode is selected based upon the dial string entered by a user. Dial strings can be stored at the device or on a network. The apparatus can include a mode selector to select a desired vocoder mode to generate an encoded voice signal.

Abstract: For the transmission of a secondary information flow between a transmitter and a receiver, the secondary information flow is inserted at a parametric vocoder of the transmitter which generates a main information flow. The main information flow is a speech data flow encoding a speech signal and is transmitted from the transmitter to the receiver. Bits from the secondary information flow are inserted into only some of the frames of the main information flow, these frames being selected by a frame mask which is known to the transmitter and the receiver, and/or into a determined frame of the main information flow, by imposing a constraint on only some of the bits of the frame, these bits being selected by a bit mask known to the emitter and the receiver.

Abstract: The present invention overcomes problems of tandem coding method such as degradation of speech quality, increased system latency and computations. An apparatus for trans-coding between code excited linear prediction (CELP) type codecs with different bandwidths, includes: a format parameter translating unit for generating output formant parameters by translating formant parameters from input CELP format to output CELP format; a formant parameter quantizing unit for receiving the output format formant parameters and quantizing the output format formant filter coefficients; an excited parameter translating unit for generating output excitation parameters by translating excitation parameters from input CELP format to output CELP format; and an excitation quantizing unit for receiving the output format excitation parameters and quantizing the output format excitation parameters.

Abstract: A method and apparatus are provided for determining the quality of a speech transmission, including temporal clipping, delay and jitter, using a carefully constructed test signal (300) and digital signal processing techniques. The test signal that is to be transmitted through a speech transmission system (100) is created (700). Then the test signal is transmitted through the speech transmission system such that the speech transmission system creates an output signal that corresponds to the input signal, as modified by the speech transmission system (702). The test signal includes multiple segments (500) of speech signals interleaved with periods of silence. The periods of silence vary in duration according to a predefined pattern. Each segment of speech signals includes multiple predefined speech samples or symbols (400, 402, 404, 406, 408, 410, 412, 414) interleaved with a plurality of silence gaps. The speech samples have a common period of duration, but the silence gaps do not.

Abstract: A data processing apparatus able to start decoding at a timing earlier than the conventional timing and able to reduce the storage capacity required for a storing means for storing the encoded data until a decoding side decodes the input encoded data in comparison with the conventional storage capacity, which apparatus selects frame data from frame data f(5) having the last decoding order to frame data f(0) having the first decoding order for processing for calculating a delay time min_delay and calculates the delay time min_delay. It calculates the delay time min_delay indicating the delay time from when the decoding side starts to receive input of the frame data to when the data is decoded based on the specified size and the bit rate of the input of the frame data to the decoding side for each of the frame data for processing.

Abstract: A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution.

Abstract: According to the present invention, it is possible to calculate appropriate chirp factor and noise component amount with a little processing amount. Input subband signal is segmented into a plurality of ranges by a range segmentation unit 101. The range segmentation is performed for energy value calculation, chirp factor calculation, noise component calculation, and tone component calculation, respectively, and determined range segmentation information ei, bi, qi, and hi are outputted. Respective processing for the energy calculation, the chirp factor calculation, the tone component calculation, and the noise component calculation are performed sequentially for the respective corresponding ranges. By using linear prediction processing, it is possible to obtain an parameter having higher accuracy with a little operation amount.

Abstract: Speech signal classification and encoding systems and methods are disclosed herein. The signal classification is done in three steps each of them discriminating a specific signal class. First, a voice activity detector (VAD) discriminates between active and inactive speech frames. If an inactive speech frame is detected (background noise signal) then the classification chain ends and the frame is encoded with comfort noise generation (CNG). If an active speech frame is detected, the frame is subjected to a second classifier dedicated to discriminate unvoiced frames. If the classifier classifies the frame as unvoiced speech signal, the classification chain ends, and the frame is encoded using a coding method optimized for unvoiced signals. Otherwise, the speech frame is passed through to the “stable voiced” classification module. If the frame is classified as stable voiced frame, then the frame is encoded using a coding method optimized for stable voiced signals.

Abstract: The present invention provides a speech signal compression device which allows a storage capacity of data representing speech to be efficiently compressed. In the present invention, a computer C1 operates with respect to speech data to be compressed into speech data for each phoneme on the basis of phoneme labeling data, to unify the time length of a unit pitch section for each of the divided speech data into the same value, thereby creating a pitch waveform and creating a sub-band data representing variation in time of spectrum components of the pitch waveform signal. Also, this sub-band data is compressed so as to match a condition designated by a table for compression, and the compressed data is further encoded in entropy to output the entropy coded data.

Abstract: A method and apparatus include a voice activity detection module configured to detect silent frames, and a codec mode selection module configured to determine a codec mode. The voice activity detection module includes a receiver configured to receive a frame, a first determiner configured to determine a first set of parameters from the frame, and a providing unit configured to provide the first set of parameters to the codec mode selection module. The codec mode selection module includes a second determiner configured to determine a second set of parameters in dependence on the first set of parameters, and a selector configured to select a codec mode in dependence on the second set of parameters.

Abstract: A method and system for decoding WCDMA AMR speech data using redundancy may include generating at least one bit-sequence for at least one of a plurality of channels that comprises received WCDMA speech data. The bit-sequence may be generated by using a decoding algorithm and may be decrypted to recover the data that may have been encrypted before being transmitted. At least one bit-sequence may be selected for each of the channels by using redundancy, such as, for example, CRC, in the received WCDMA speech data. The redundancy in the received WCDMA speech data may be, for example, CRC. The bit-sequence for each of the channels may be combined to form at least one speech stream. A speech stream may be selected based on speech constraints, which may comprise gain continuity and/or pitch continuity. The selected speech stream may be communicated to a voice decoder.

Abstract: In a sinusoidal audio encoder a number of sinusoids are estimated per audio segment. A sinusoid is represented y frequency, amplitude and phase. Normally, phase is quantised independent of frequency The invention uses a frequency dependent quantisation of phase, and in particular the low frequencies are quantised using smaller quantisation intervals than at higher frequencies. Thus, the unwrapped phases of the lower frequencies are quantised more accurately, possibly with a smaller quantisation range, than the phases of the higher frequencies. The invention gives a significant improvement in decoded signal quality, especially for low bit-rate quantisers.

Abstract: A method of deriving a codebook including deriving a first codeword Pl in Euclidean coordinates, transforming the first codeword into eigen-coordinates, applying a Hochwald construction to the first codeword in eigen-coordinates to derive a plurality of codewords in eigen-coordinates and transforming the plurality of codewords in eigen-coordinates into a plurality of codewords in Euclidean coordinates to form the codebook.

Abstract: The object of this invention is converting a code that has been obtained by encoding speech by one particular system is converted to code that can be decoded by another system with high speech quality, and moreover, with a low computational load in transmitting speech signal between different systems. This invention comprising an adaptive codebook (ACB) delay search range control circuit (1250 in FIG. 7) for calculating a search range control value from first adaptive codebook delay that is stored and held and said second adaptive codebook delay that is stored and held, and an adaptive codebook encoding circuit (1220 in FIG.

Abstract: A system and method for providing voice communications with desired characteristics based upon the intended recipient of a voice communication. An apparatus includes a list of dial strings associated with parties having desired voice communication characteristics. A dial string entered by a user and associated with an intended recipient is compared to a list of preferred dial strings to determine the characteristics of an encoded voice signal to be sent to the recipient. The apparatus can include a vocoder having different bit rate modes and a bit rate mode is selected based upon the dial string entered by a user. Dial strings can be stored at the device or on a network. The apparatus can include a mode selector to select a desired vocoder mode to generate an encoded voice signal.

Abstract: A method and system for identifying explanatory text for a referenced web page based on a reference to the referenced web page contained in a repeated pattern of a referencing web page is provided. An anchor explanatory text (“AET”) system uses the hierarchical organization of the web page to identify a repeated pattern of hierarchical elements that contain references to other display pages. After the AET system identifies a repeated pattern, it identifies the dominant reference or anchor within each occurrence of the pattern. The AET system uses the explanatory text surrounding a dominant anchor as a description of the referenced web page.

Abstract: Tandem-free vocoder operations (TFO) between non-compatible communication systems may be enabled through hardware modifications at communication elements within each system. In one aspect, each infrastructure entity in System 1 comprises an intra-system TFO Frame Generator G1, an intra-system TFO Frame Extractor E1, and a TFO Frame Extractor E2 of System 2, which is non-compatible to System 1. Each infrastructure entity in System 2 comprises an intra-system TFO Frame Generator G2, an intra-system TFO Frame Extractor E2, and a TFO Frame Extractor E1 of System 1.

Abstract: Method and apparatus for enhancing a coded audio signal comprising indices which represent audio signal parameters which comprise at least a first parameter representing a first characteristic of speech are disclosed. A current first parameter value is determined from an index corresponding to at least the first parameter. The current first parameter value is adjusted in order to achieve an enhanced first characteristic, thereby obtaining an enhanced first parameter value. A new index value is determined from a table relating index values to at least first parameter values, such that a new first parameter value corresponding to the new index value substantially matches the enhanced first parameter value.

Abstract: A method of encoding a frame in a communication network using multiple codec modes, wherein the frame encoded by each codec mode is represented by multiple parameters. The method includes at least one stage, wherein the stage includes the steps of selecting one group from multiple groups of codec modes, wherein each group includes at least one codec mode and is arranged to have a common parameter characteristic. The method further includes encoding the frame with one of the codec modes from the selected group in dependence on the common parameter characteristic.

Abstract: Provided is a residual signal coding/decoding apparatus and method. The residual signal coding apparatus includes a transformer, an LPC coefficient extractor, an LPC coefficient quantizer, an LP analysis filter, a band splitter, a pulse searcher, and a pulse quantizer. The transformer transforms time-domain residual signals into a frequency domain to output transform coefficients. The LPC coefficient extractor extracts LPC coefficients from the transform coefficients. The LPC coefficient quantizer quantizes the LPC coefficients to output quantized LPC coefficients and corresponding indices. The LP analysis filter performs an LP analysis on the transform coefficients to output LP residual transform coefficients. The band splitter splits the LP residual transform coefficients into bands to output the LP residual transform coefficients. The pulse searcher searches the LP residual transform coefficients for the respective bands to select optimal pulses and output parameters of the optimal pulses.

Abstract: A conferencing system is provided that utilizes both time domain signal mixing and direct signal fast transcoding. An exemplary embodiment of the present invention utilizes both time domain signal mixing and direct signal fast transcoding to process a bit-stream from a same channel during a conference.

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: At the speech encoding end, upon generation of an fixed excitation vector, the shape of an excitation vector output from pulse excitation codebook 301 is identified in pulse excitation vector shape identifier 302, a dispersion vector used for excitation vectors of the shape is output from dispersion vector storage 304, and, in dispersion vector convolution processor 303, dispersion vector convolution processing of the excitation vector is performed. In particular, when a pulse excitation vector having a specific shape of high frequency of use is output from pulse excitation codebook 301, pulse excitation vector shape identifier 302 controls dispersion vector storage 304 in such a way that an additional dispersion vector prepared dedicated to the pulse excitation vector is output. By this means, it is possible to provide a technology that improves the quality of decoded speech and that decodes speech more natural and audible to the user.

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: 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: Techniques for implementing vocoders in parallel digital signal processors are described. A preferred approach is implemented in conjunction with the BOPS® Manifold Array (ManArray™) processing architecture so that in an array of N parallel processing elements, N channels of voice communication are processed in parallel. Techniques for forcing vocoder processing of one data-frame to take the same number of cycles are described. Improved throughput and lower clock rates can be achieved.

Abstract: Coding of spectral data by representing certain portions of the spectral data as a scaled version of a code-vector, where the code-vector is chosen from either a fixed predetermined codebook or a codebook taken from a baseband. Various optional features are described for modifying the code-vectors in the codebook according to some rules which allow the code-vector to better represent the data they are modeling. The code-vector modification comprises a linear or non-linear transform of one or more code-vectors, such as, by exponentiation, negation, reversing, or combining elements from plural code-vectors.

Abstract: A digital line transmission unit can carry out switching between speech codecs during the same call to achieve balance between making effective use of a line and a high sound quality without bringing about a feeling of discomfort in a user by the switching. It includes in an encoder a first speech codec 7 with a high sound quality and a high bit rate, a second speech codec 8 with a reasonable sound quality but a low bit rate. It carries out switching between these speech codecs in response to the control information an operation monitoring controller 4 obtains by making a decision as to the traffic volume of the bearer line 111. The switching between the speech codecs is made during a speech pause a speech burst detector 31 in a signal detector 3 detects in an input speech signal.

Abstract: A mixed lossless audio compression has application to a unified lossy and lossless audio compression scheme that combines lossy and lossless audio compression within a same audio signal. The mixed lossless compression codes a transition frame between lossy and lossless coding frames to produce seamless transitions. The mixed lossless coding performs a lapped transform and inverse lapped transform to produce an appropriately windowed and folded pseudo-time domain frame, which can then be losslessly coded. The mixed lossless coding also can be applied for frames that exhibit poor lossy compression performance.

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 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: Transcoding from EVRC to G.729ab with LSP parameters interpolated from EVRC to G.729ab, EVRC pitch used as input to G.729ab closed-loop pitch search, and G.729ab fixed codebook pulses found from a search limited to positions of EVRC fixed codebook pulses together with positions of target-impulse correlation maxima on the subframe tracks or full track search if no EVRC pulses.

Abstract: The invention provides a method of retransmitting a speech packet. In one embodiment, the method includes receiving (S702) at a transmitting device (140) a first negative acknowledgement (NACK) from a receiving device (120). The NAK indicates a corrupted first speech packet transmission. The transmitting device then retrieves (S706) a first speech packet associated with the first NACK and compresses (S714-S720) the speech packet to form a replacement speech packet. Next, a current segment of speech is encoded (S808) to form a current speech packet and current speech packet is combined with the replacements speech packet. The combined speech packet is then transmitted (S814) to the receiving device.

Abstract: An isochronous telecommunication stream comprises a plurality of frames encoded by a variable rate isochronous coder-decoder (codec) at a plurality of code rates of multiple available code rates. A control component removes one or more encoded frames from the plurality of frames of the isochronous telecommunication stream based on one or more code rates of the one or more encoded frames to create a non-isochronous telecommunication stream.

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: A speech communication system provides a speech encoder that generates a set of coded parameters representative of the desired speech signal characteristics. The speech communication system also provides a speech decoder that receives the set of coded parameters to generate reconstructed speech. The speech decoder includes an equalizer that computes a matching set of parameters from the reconstructed speech generated by the speech decoder, undoes the set of characteristics corresponding to the computed set of parameters, and imposes the set of characteristics corresponding to the coded set of parameters, thereby producing equalized reconstructed speech.

Abstract: A two-way conversion transcoder comprising a spectrum parameter calculation circuit that calculates a spectrum parameter for a signal produced by decoding a first code; a coefficient calculation circuit that receives the spectrum parameter and converts it to the coefficients of a band extended signal, a noise generation circuit that outputs a band-limited noise signal, a gain circuit that multiplies the output signal of the noise generation circuit by a gain, a synthesis filter circuit that receives the output signal from the noise generation circuit and the coefficients from the coefficient calculation circuit and outputs a high frequency signal for band extension, a sampling frequency conversion circuit that outputs a signal generated by up-sampling the signal to a predetermined sampling frequency, an adder that adds up a high-frequency signal and the signal to form a band extended signal, and a second encoding circuit that encodes the band extended signal by a second encoding method and outputs the encoded

Abstract: The present invention provides an energy feature extraction method for noisy speech recognition. At first, noisy speech energy of an input noisy speech is computed. Next, the noise energy in the input noisy speech is estimated. Then, the estimated noise energy is subtracted from the noisy speech energy to obtain estimated clean speech energy. Finally, delta operations are performed on the log of the estimated clean speech energy to determine the energy derivative features for the noisy speech.

Abstract: A frame type for a current SBR frame is determined according to a type of end border of a previous frame, as well as presence of a transient in the current SBR frame. A start border is determined according to the end border of the previous SBR frame. For a FIXFIX frame, a low time-resolution setting is used. For a FIXVAR or a VARVAR frame, a search for intermediate borders is conducted in the region between the transient and maximum allowed end border location. The end border is also determined at this stage. If there is excess capacity for more borders, another search is conducted in the region between the transient and the start border. For a VARFIX frame, only one search needs to be conducted, in the whole region partitioned by a variable start border and a fixed end border. All of the above are accomplished with two Forward Search operations and one Backward Search operation.

Abstract: A variable-rate voice transcoder that transcodes a bitstream representing frames of data encoded according to a first compression standard to a bitstream representing frames of data according to a second compression standard; the second compression standard defines a variable-rate voice codec. The method includes unquantizing a bitstream into a first set of parameters compatible with the first compression standard. The first set of parameters in addition to external control commands are then used to determine a frame class and a rate for the second compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with the second compression standard according to the frame-classification and rate determination decision. Lastly, the second set of parameters is packed into a bitstream compatible with the second compression standard.

Abstract: An automated method of providing a pronunciation of a word to a remote device is disclosed. The method includes receiving an input indicative of the word to be pronounced. The method further includes searching a database having a plurality of records. Each of the records has an indication of a textual representation and an associated indication of an audible representation. At least one output is provided to the remote device of an audible representation of the word to be pronounced.

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 decoding apparatus is provided. The decoding apparatus has a first decoding part for decoding a code word obtained by encoding an input signal using a Code-Excited Linear Prediction encoding method. A second decoding part decodes a code word obtained by encoding a signal with an encoding method other than the Code-Excited Linear Prediction encoding method. A rising-transition detection and notification part has a detection part that detects the existence of a rising-transition of amplitude of the input signal based on time variation of a gain of excitation vectors obtained by the first decoding part, and a notification part that notifies the second decoding part that the rising-transition of the amplitude exists.

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: Upon receiving (101) a vocoded voice frame and detecting (102) that the received vocoded voice frame comprises an erased frame, one automatically replaces (103) the erased frame with a valid frame having at least one error condition. In a preferred approach this error condition is one that is known to cause a receiving target platform to invoke a corresponding erasure process with respect to the valid frame when received.

Abstract: An overflow problem of LSF quantization in G.729 Annex B speech encoding which may lead to non-assignment of a codebook index. Preferred embodiments fix the problem with default or limited random variable assignments or flagging the overflow and adjusting the frame encoding such as by limiting spectral components or changing quantization targets.

Abstract: An apparatus and method for mapping CELP parameters between a source codec and a destination codec. The apparatus includes an LSP mapping module, an adaptive codebook mapping module coupled to the LSP mapping module, and a fixed codebook mapping module coupled to the LSP mapping module and the adaptive codebook mapping module. The LSP mapping module includes an LP overflow module and an LSP parameter modification module. The adaptive codebook mapping module includes a first pitch gain codebook. The fixed codebook mapping module includes a first target processing module, a pulse search module, a fixed codebook gain estimation module, a pulse position searching module.