Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: In a Noise Feedback Coding (NFC) system having a corresponding ZERO-STATE filter structure, the first ZERO-STATE filter structure including multiple filters, a method of producing a ZERO-STATE response error vector. The method includes: (a) transforming the first ZERO-STATE filter structure to a second ZERO-STATE filter structure including only an all-zero filter, the all-zero filter having a filter response substantially equivalent to a filter response of the ZERO-STATE filter structure including multiple filters; and (b) filtering a VQ codevector with the all-zero filter to produce the ZERO-STATE response error vector corresponding to the VQ codevector.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: An extended signal coding system that accommodates substantially music-like signals within a signal while maintaining a high perceptual quality in a reproduced signal during discontinued transmission (DTX) operation. The extended signal coding system contains internal circuitry that performs detection and classification of the speech signal, depending on numerous characteristics of the signal, to ensure the high perceptual quality in the reproduced signal. In certain embodiments of the invention, the signal is a speech signal, and the speech signal has a substantially music-like signal contained therein, and the extended signal coding system overrides any voice activity detection (VAD) decision that is used to determine which among a plurality of source coding modes are to be employed using a voice activity detection (VAD) correction/supervision circuitry. This is particularly relevant for discontinued transmission (DTX) operation.

Abstract: In a cellular telephone system where a digital cellular telephone is connected to a regular telephone through the public switched telephone network (PSTN), a speech encoder/decoder is used with an A/&mgr;-Law encoder/decoder causing annoying audible noise at very low levels because of the quantization characteristics of the A/&mgr;-Law encoder/decoder. This noise is eliminated by adding a digital constant to the output of the speech coder, shifting the low level signal away from zero. The resulting DC level added to the speech signal is inaudible to the PSTN telephone user and does not degrade speech quality. Alternatively, the constant added to the output of the speech coder is confined to a small value added to the speech coder output to move the entire speech coder output during the silence period, between speech periods, above zero or below zero.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: In a Noise Feedback Coding (NFC) system having a corresponding ZERO-STATE filter structure, the first ZERO-STATE filter structure including multiple filters, a method of producing a ZERO-STATE response error vector. The method includes: (a) transforming the first ZERO-STATE filter structure to a second ZERO-STATE filter structure including only an all-zero filter, the all-zero filter having a filter response substantially equivalent to a filter response of the ZERO-STATE filter structure including multiple filters; and (b) filtering a VQ codevector with the all-zero filter to produce the ZERO-STATE response error vector corresponding to the VQ codevector.

Abstract: In a Noise Feedback Coding (NFC) system operable in a ZERO-STATE condition and a ZERO-INPUT condition, the NFC system including at least one filter having a filter memory, a method of updating the filter memory. The method comprises: (a) producing a ZERO-STATE contribution to the filter memory when the NFC system is in the ZERO-STATE condition; (b) producing a ZERO-INPUT contribution to the filter memory when the NFC system is in the ZERO-INPUT condition; and (c) updating the filter memory as a function of both the ZERO-STATE contribution and the ZERO-INPUT contribution.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

Abstract: A filter controller processes a decoded speech (DS) signal. The DS signal has a spectral envelope including a first plurality of formant peaks having different respective amplitudes. The controller produces, from the DS signal, a spectrally-flattened DS signal that is a time-domain signal. The spectrally-flattened time-domain DS signal has a spectral envelope including a second plurality of formant peaks. Each of the second plurality of formant peaks approximately coincides in frequency with a respective one of the first plurality of formant peaks. Also, the second plurality of formant peaks have approximately equal respective amplitudes. Next, the controller derives, from the spectrally-flattened time-domain DS signal, a set of filter coefficients representative of a filter response that is to be used to filter the DS signal.

Abstract: A method to eliminate discontinuities in an adaptively filtered signal includes filtering a beginning portion of a current signal frame using a past set of filter coefficients, thereby producing a first filtered frame portion. The method also includes filtering the beginning portion of the current signal frame using a current set of filter coefficients, thereby producing a second filtered frame portion. The method also includes modifying the second filtered frame portion with the first filtered frame portion so as to smooth a possible filtered signal discontinuity between the second filtered frame portion and a past filtered frame produced using the past filter coefficients.

Abstract: A method of performing an excitation Vector Quantization (VQ) in a Noise Feedback Coding environment involves reorganizing a calculation of an energy of an error vector for each of a plurality of candidate excitation vectors of a codebook. The energy of the error vector is a cost function that is minimized during a search of the codebook for a best candidate excitation VQ vector. The reorganization includes expanding a Mean Squared Error (MSE) term of the error vector, excluding an energy term that is invariant to the candidate excitation vector, and pre-computing energy terms of ZERO-STATE responses of the candidate excitation vectors that are invariant to sub-vectors of a subframe. Another method searches a signed codebook. Both methods use correlation techniques.

Abstract: A quantizer for quantization of a vector comprises a codevector generator that generates a set of candidate codevectors and a memory for storing an illegal space definition representing illegal vectors. The quantizer also includes a legal status tester that determines legal candidate codevectors among the set of candidate codevectors using the illegal space definition, and a codevector selector that determines a best legal candidate codevector among the one or more legal candidate codevectors. The vector includes parameters relating to a speech and/or audio signal, such as Line Spectral Frequencies (LSFs).

Abstract: A method of searching a signed codebook to quantize a vector includes weighting a shape codevector in a set of shape codevectors with a weighting function for a Weighted Mean Square Error (WMSE) criteria, to produce a weighted shape codevector. The method further includes correlating the weighted shape codevector with the vector to produce a weighted correlation term. The method also includes determining, based on a sign of the weighted correlation term, a preferred one of a positive and a negative signed codevector associated with the shape codevector. The method further includes determining whether one of the signed codevectors does not belong to an illegal space defining illegal vectors.

Abstract: A sub-quantizer for sub-quantization of a vector includes a sub-codevector generator that generates a set of candidate sub-codevectors, and transformation logic that transforms each candidate sub-codevector into a corresponding codevector. A memory stores an illegal space definition representing illegal vectors. A legal status tester determines legal codevectors among the codevectors based on the illegal space definition. An error calculator generates error terms corresponding to the candidate sub-codevector, and a sub-codevector selector determines a best one of the sub-codevectors corresponding to a legal codevector and a best error term. The vector includes parameters relating to a speech and/or audio signal, such as Line Spectral Frequencies (LSFs).