Interdigital Technology Corporation

International Mobile Machines Corporation

Hughes Electronics Corporation

Interdigital Technology Corpoartion

Interdigital Patents Corporation

Daniel Lin

Brian Michael McCarthy, Sr

Scott David Kurtz

James M Kresse

Kumar Swaminathan

A method and apparatus for processing speech in a wireless communication system uses CELP speech encoded signals. A decoder receives encoded speech including a code index, a code index gain, a pitch lag, a pitch gain, and a line spectral pair (LSP) index. An innovation codevector and an adaptive codevector are determined and scaled. An excitation sequence is generated. Reconstructed speech is then output based on the excitation sequence and LSP index.

Method and apparatus for receiving an encoded speech signal based on codebooks

A method and apparatus for processing speech in a wireless communication system uses code excited linear prediction (CELP) speech encoded signals. A speech input receives samples of a speech signal and a codebook analysis block selects an index of a code from one or more codebooks. A prediction error between a predicted current sample and a current sample of the speech samples is determined. An innovation sequence is determined based on the prediction error and an index is selected based on the innovation sequence. The index is transmitted to the receiver to enable reconstruction of the speech signal at the receiver.

Method and apparatus for transmitting an encoded speech signal

The present invention is a synthetic speech encoding device that produces a synthetic speech signal which closely matches an actual speech signal. The actual speech signal is digitized, and excitation pulses are selected by minimizing the error between the actual and synthetic speech signals. The preferred pattern of excitation pulses needed to produce the synthetic speech signal is obtained by using an excitation pattern containing a multiplicity of weighted pulses at timed positions. The selection of the location and amplitude of each excitation pulse is obtained by minimizing an error criterion between the synthetic speech signal and the actual speech signal. The error criterion function incorporates a perceptual weighting filter which shapes the error spectrum.

Method and device for encoding speech using open-loop pitch analysis

A method and apparatus for processing speech in a wireless communication system uses CELP speech encoded signals. A speech input receives samples of a speech signal and a codebook analysis block for selects an index of a code from at least one of a plurality of codebooks. A weighted synthesis filter is used in the generation of a prediction error between a predicted current sample and a current sample of the speech samples. The index is transmitted to the receiver to enable reconstruction of the speech signal at the receiver.

A method for processing speech in a spread spectrum communication system uses CELP speech encoded signals. A speech input receives samples of a speech signal and a codebook analysis block for selects an index of a code from each of a plurality of codebooks. A weighted synthesis filter is used in the generation of a prediction error between a predicted current sample and a current sample of the speech samples. The index is transmitted to the receiver to enable reconstruction of the speech signal at the receiver.

Receiver for encoding speech signal using a weighted synthesis filter

The present invention is a method for determining linear predictive coding filter parameters for encoding a voice signal. The method includes sampling a voice signal, grouping the samples into a plurality of frames, generating a plurality of reflection coefficients for each frame of samples, quantizing the reflection coefficients, generating spectral coefficients from the quantized reflection coefficients, selecting a quantized reflection coefficient having the smallest log-spectral distance between a quantized spectrum, and an unquantized spectrum and, converting the selected quantized reflection coefficient to linear predictive coding (LPC) filter coefficient.

Determining linear predictive coding filter parameters for encoding a voice signal

Linear predictive coding (LPC) filter parameters are determined for use in encoding a voice signal. Samples of a speech signal using a z-transform function are pre-emphasized. The pre-emphasized samples are analyzed to produce LPC reflection coefficients. The LPC reflection coefficients are quantized by a voiced quantizer and by an unvoiced quantizer producing sets of quantized reflection coefficients. Each set is converted into respective spectral coefficients. The set which produces a smaller lag-spectral distance is determined. The determined set is selected to encode the voice signal.

A receiver is used in decoding a received encoded signal. The received encoded speech signal is encoded using excitation linear prediction. The receiver receives the encoded speech signal. The encoded speech signal comprises a code, a pitch lag and a line spectral pair index. An innovation sequence is produced by selecting a code from each of a plurality of codebooks based on the code index. A line spectral pair quantization of a speech signal is determined using the line spectral pair index. A pitch lag is determined using the pitch lag index. A speech signal is reconstructed using the produced innovation sequence, the determined line spectral pair quantization and pitch lag.

Receiver for receiving a linear predictive coded speech signal

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.

Encoding a speech signal using code excited linear prediction using a plurality of codebooks

A version of a speech signal and an output of a pitch synthesis filter and a linear predictive all-pole (LPC) filter is received. A system impulse response is produced based on in part the received pitch synthesis filter and LPC output. An excitation pulse location is determined so that the determined location minimizes an error between the speech signal version and the system impulse response. The speech signal is encoded with a representation of the determined location.

Multiple impulse excitation speech encoder and decoder

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 &lcub;0, 1&rcub;. The decoder also using a second codebook containing a second digital value sequence having values selected from the set of binary values &lcub;&minus;1, 0&rcub;. 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 &lcub;&minus;1, 0, 1&rcub;.

Efficient codebook structure for code excited linear prediction coding

To perform pitch analysis for encoding a speech signal, a speech signal is sampled. The sampled speech signal is spectrally whitened to produce a spectral residual signal. Samples of the spectral residual signal are collected and the collected samples are autocorrelated. Maximum values of the correlated result are determined. Gain values are determined based on at least in part the maximum values of the correlated result. The gain values are quantized using a codebook to produce a codebook index and an associated frame delay. The codebook index and the frame delay represent a pitch of the speech signal to facilitate encoding the speech signal.

The generation of multipulse excitation codes by digitizing an original speech, partitioning the digitized signal into a number of samples, pre-emphasizing the samples, producing linear predictive reflection coefficients from said samples, quantizing these reflection coefficients, converting the quantized reflection coefficients to spectral coefficients and subjecting the spectral coefficients to pitch analysis to obtain a spectral residual signal.

A line spectral frequency (LSF) vector quantizer, having particular application in digital cellular networks (DCN), is provided for code excited linear predictive (CELP) speech encoders. The LSF vector quantizer is efficient in terms of bits employed, robust and effective in terms of performance across speakers and handsets, moderate in terms of complexity, and accommodates effective and simple built-in transmission error detection schemes. The LSF vector quantizer employs a minimum number of bits, is of moderate complexity and incorporates built-in error detection capability in order to combat transmission errors. The LSF vector quantizer classifies unquantized line spectral frequencies into four categories, employing different vector quantization tables for each category. Each quantization table is optimized for particular types of vectors. For each category, three split vector codebooks are used with a simplified error measure to find three candidate split quantized vectors. The three sets of three split vectors are combined to produce as many as 27 vectors from each category. The quantizer then makes a final selection of optimal category using a more complex error measure to achieve the robust performance across speakers and handsets. Split vector quantization follows a two stage constrained search procedure that results in an ordered set of quantized line spectral frequencies that is 'close' to the unquantized set with moderate complexity within each category. Effective and simple transmission error detection schemes at the receiver are made possible by the split nature of the vector quantization and the constrained search procedure. Only twenty-six bits are required to encode ten line spectral frequencies.

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Daniel Lin