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: A method and apparatus for noise suppression is described herein. The channel gain is controlled based on a degree of variability of the background noise. The noise variability estimate is used in conjunction with a variable attenuation concept to produce a family of gain curves that are adaptively suited for a variety of combinations of long-term peak SNR and noise variability. More specifically, a measure of the variability of the background noise is used to provide an optimized threshold that reduces the occurrence of non-stationary background noise entering into the transition region of the gain curve.

Abstract: A method and apparatus for noise suppression is described herein. The channel gain is controlled based on a degree of variability of the background noise. The noise variability estimate is used in conjunction with a variable attenuation concept to produce a family of gain curves that are adaptively suited for a variety of combinations of long-term peak SNR and noise variability. More specifically, a measure of the variability of the background noise is used to provide an optimized threshold that reduces the occurrence of non-stationary background noise entering into the transition region of the gain curve.

Abstract: The invention provides a method of coding an information signal. An information signal is represented by a sequence of pulses. A plurality of pulse parameters are determined based on the sequence of pulses including a non-zero pulse parameter corresponding to a number of non-zero pulse positions in the sequence of pulses. The non-zero pulse parameter is coded using a variable-length codeword.

Abstract: The invention provides a method of coding an information signal. An information signal is represented by a sequence of pulses. A plurality of pulse parameters are determined based on the sequence of pulses including a non-zero pulse parameter corresponding to a number of non-zero pulse positions in the sequence of pulses. The non-zero pulse parameter is coded using a variable-length codeword.

Abstract: In order for the Voice Activity Detector (VAD) decision to overcome the problem of being over-sensitive to fluctuating, non-stationary background noise conditions, a bias factor is used to increase the threshold on which the VAD decision is based. This bias factor is derived from an estimate of the variability of the background noise estimate. The variability estimate is further based on negative values of the instantaneous SNR.

Abstract: Bits are allocated to short-term repetition information for unvoiced input signals. Stated differently, more bits are allocated for pitch information during unvoiced input speech than in the prior art. The improved method and apparatus in an encoder (300) and decoder (700) result in improved consistency of amplitude pulses compared to prior art methods which indicates improved stability due to increased search resolution. Also, the improved method and apparatus result in higher energy compared to prior art methods which indicates that the synthesized waveform matches the target waveform more closely, resulting in a higher fixed codebook (FCB) gain.

Abstract: A noise suppression system implemented in communication system provides an improved level of quality during severe signal-to-noise ratio (SNR) conditions. The noise suppression system, inter alia, incorporates a frequency domain comb-filtering (289) technique which supplements a traditional spectral noise suppression method. The invention includes a real cepstrum generator (285) for an input signal (285) G(k) to produce a likely voiced speech pitch lag component and converting a result to frequency domain to obtain a comb-filter function (290) C(k), applying input signal (291) G(k) to comb-filter function (290) C(k), and equalizing the energies of the corresponding pre and post filtered subbands, to produce a signal (293) G″(k) to be used for noise suppression. This prevents high frequency components from being unnecessarily attenuated, thereby reducing muffling effects of prior art comb-filters.

Abstract: An improved speech coder takes advantage of the fact that any given pulse combination can be uniquely described by the following four properties: number of degenerate pulses, signs of pulses, positions of pulses, and pulse magnitudes. In accordance with the invention, a four stage iterative classification of the pulse combinations, where each stage groups the pulse combinations by one of these four properties, is performed. The process starts with the number of pulses, then determines the total number of possible sign combinations, pulse position combinations, and pulse magnitude combinations. This flexibility allows for the sign combinations to be grouped in the last stage. Since the number of sign combinations is always a power of two, leaving the sign combinations for last along with appropriately ordering the elements in the previous three stages allows the signs to be coded by independent bits, in turn allowing for error protection of those bits.

Abstract: A method and apparatus for improved pitch period (&tgr;) estimation in a compression system is disclosed. The system uses original estimates of integer lag (&tgr;0) and open-loop prediction gain (&bgr;ol) as input to an adaptive filter parameter initialization block (304) which supplies inputs to a plurality of adaptive filter elements (306-308). Adaptive filter elements (306-308) provide information regarding the harmonics of the residual signal (&egr;(n)) to an adaptive filter parameter analysis block (310). Adaptive filter parameter analysis block (310) estimates the fundamental frequency of the residual signal based on the analysis of the harmonics and outputs a pitch period (&tgr;) for eventual use in a delay contour computation.

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: An apparatus (200) for providing a multi-party speech connection for use in a wireless communication system (10). The apparatus (200) comprises a first speech encoder (202) producing a first encoded speech signal, a second speech encoder (20) producing a second encoded speech signal, a conference circuit (22), and a speech decoder (208) responsive to the conference circuit (22). The conference circuit (22) receives the first and second speech encoded signals and produces a multiplexed encoded speech signal. The speech decoder (208) receives the multiplexed encoded speech signal and produces a decoded speech signal.

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 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.

Abstract: A method and system for controlling an encoding rate in a communication system utilizes feedforward rate information (48) and/or rate desirability information (50) sent from each of a plurality of variable rate vocoders (34) to a communication link output controller (24), such as a network arbitor. The communication link output controller (24) then sends a feedback control signal (32) to a selected variable rate vocoder to change the encoding rate of the selected variable rate vocoder to facilitate re-encoding of the speech packet when a bottleneck is detected. In another embodiment, the network arbitor (134) may additionally and independently modify speech packet data when it determines that a bottleneck may occur. The network arbitor (134) also communicates a packet modification control signal (132) for the variable rate vocoder that generated the dropped packet so that the corresponding variable rate vocoder can adjust its filter states to maintain convergence.