Abstract: A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided.

Abstract: A system determines a voicing measure as a measure of the degree of signal periodicity and uses the determined voicing measure to quantize the spectral magnitude of the slowly evolving waveform (SEW) and the modeling of the SEW and rapidly evolving waveform (REW) phase spectra.

Abstract: A low bit rate voice codec based on Frequency Domain Interpolation (FDI) technology is designed to operate at multiple rates of 4.0, 2.4, and 1.2 Kbps. At 4 Kbps, the codec uses a 20 ms frame size and a 20 ms lookahead for purposes of voice activity detection (VAD), noise reduction, linear prediction (LP) analysis, and open loop pitch analysis. The LP parameters are encoded using backward predictive hybrid scalar-vector quantizers in the line spectral frequency (LSF) domain after adaptive bandwidth broadening to minimize excessive peakiness in the LP spectrum. Prototype Waveforms (PW) are extracted every subframe or 2.5 ms from the LP residual and subsequently aligned and normalized. The PW gains are encoded separately using a backward predictive vector quantizer (VQ). The normalized and aligned PWs are separated into a magnitude component and a phase component. The phase component is encoded implicitly using PW correlations and a voicing measure which are jointly quantized using a VQ.

Abstract: Encoding of prototype waveform components applicable to telecommunication systems provides improved voice quality enabling a dual-channel mode of operation which permits more users to communicate over the same physical channel. A prototype word (PW) gain is vector quantized using a vector quantizer (VQ) that explicitly populates a codebook by representative steady state and transient vectors of PW gain for tracking the abrupt variations in speech levels during onsets and other non-stationary events, while maintaining the accuracy of the speech level during stationary conditions.

Abstract: Encoding of prototype waveform components applicable to GeoMobile and Telephony Earth Station (TES) providing improved voice quality enabling a dual-channel mode of operation which permits more users to communicate over the same physical channel. A prototype word (PW) gain is vector quantized using a vector quantizer (VQ) that explicitly populates the codebook by representative steady state and transient vectors of PW gain for tracking the abrupt variations in speech levels during onsets and other non-stationary events, while maintaining the accuracy of the speech level during stationary conditions. The rapidly evolving waveform (REW) and slowly evolving waveform (SEW) component vectors are converted to magnitude-phase. The variable dimension SEW magnitude vector is quantized using a hierarchical approach, i.e., a fixed dimension SEW mean vector computed by a sub-band averaging of SEW magnitude spectrum, and only the REW magnitude is explicitly encoded.

Abstract: A communication network, comprising data transmission and data receiving systems and operations, for carrying reliable voice communications, particularly for mobile communications applications involving land-mobile, aeronautical and maritime systems. Because the quality of the links in such systems may vary, the present invention combines a nested codec with a channel coder at the transmission side of the link and a decoder at the reception side. The nested codec will transmit at a constant rate, which is the highest bit rate possible in the system, but according to channel conditions, will reconstruct voice at different bit rates. Thus, for example, transmission will always occur at a rate of 3600, but the reconstruction of the transmitted signal will be at 2400 or 1200 bps, depending on the quality of the signal. Notably, the lower two rates are subsets of the higher rate, thus defining their "nested" characteristic.

Abstract: Before transmitting signals to a receiver, the signals are subjected to adaptive prediction to generate a residual signal for transmission, and the residual signal is then transformed into frequency domain coefficients, the coefficients are grouped together to form vectors, and the vectors are then quantized.

Abstract: A method of detecting and correcting received values of a pitch period estimate of a speech signal for use in a speech coder or the like. An average is calculated of the nonzero values of received pitch period estimate since the previous reset. If a current pitch period estimate is within a range of 0.75 to 1.25 times the average, it is assumed correct, while if not, a correction process is carried out. If correction is required successively for more than a preset number of times, which will most likely occur when the speaker changes, the average is discarded and a new average calculated.