Abstract: An approach for improving quality of synthesized speech is presented. The input speech or residual is first separated into a voiced portion and a noise portion. The voice portion is coded using CELP methods. The noise portion of the input speech may be estimated at the decoder since it contains minimal voiced speech components. The separation is frequency dependent and is adaptive to the input speech. The separation may be accomplished using a lowpass/highpass filter combination. The information regarding bandwidth of the lowpass/highpass is presented to the decoder to facilitate reproduction of the noise portion of the speech.

Abstract: In one system embodiment, the system is characterized by: a differential amplifier including but not limited to at least one amplifying transistor having an emitter coupled directly to a ground. In one embodiment of a method of making a system, the method is characterized by: operably coupling at least one amplifying transistor of a differential amplifier directly to a ground. In one embodiment of a method of driving a system, the method is characterized by: driving at least one amplifying transistor of a differential amplifier with an emitter-follower feedback loop. In one system embodiment, the system is characterized by: a differential amplifier including but not limited to a first amplifying transistor having a base operably coupled with a first emitter-follower feedback loop.

Abstract: The power control system utilizing a normal average power control loop to control average power and modulation level of an optic signal generated by an optic signal generator. The target average power is adjusted by a small percentage, and the corresponding change in bias current needed to adjust the average power level is monitored. The change in average power divided by the change in bias current is a measure of the optic signal generator's slope efficiency. Modulation current is adjusted up or down based on the change in slope efficiency compared to the slope efficiency stored at the time of module calibration. By adjusting the modulation current, the desired optic signal output from the transmitter may be maintained.

Abstract: There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1?k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

Abstract: A method and apparatus for signal amplification and decision device operation on a signal which has a magnitude, peak value, or other average power that varies over time. In general, a peak detector detects the peak value of a received signal and responsive to the peak value, selectively enables a decision device when the signal is amplified to increase decision device accuracy. The decision may occur before the amplification forces the signal into saturation. Multi-stage amplification in combination with controlled switching or multiplexing may be utilized to selectively amplify the signal and selectively initiate decision device operation. The peak detector value may also control signal amplification levels in a multi-stage amplifier. Responsive to the peak value, a switch, which receives as its input the output from one or more amplifier stages, may be controlled to output the signal after a desired amount of amplification.

Abstract: There is provided a method of decoding speech data generated from a speech signal. The method comprises receiving the speech data having at least one main pulse in a subframe of the speech data; generating a first predicted pulse, based on the at least one main pulse, on one side of the main pulse in the subframe of the speech data, wherein the first predicted pulse has a lower gain than the main pulse; generating a second predicted pulse, as a mirror image of the first predicted pulse on a reverse time scale, on the other side of the main pulse in the subframe of the speech data; reconstructing the speech signal using the at least one main pulse, the first predicted pulse and the second predicted pulse.

Abstract: There is provided a method of encoding an input speech signal. The method comprises identifying a fixed codebook vector from a fixed codebook; identifying an adaptive codebook vector from a adaptive codebook; calculating an adaptive codebook gain; reducing the adaptive codebook gain by an amount; optimally selecting a fixed codebook gain based on the adaptive codebook gain while both the fixed codebook vector and the adaptive codebook vector remain fixed; and converting the input speech signal into an encoded speech using the fixed codebook gain, the adaptive codebook gain, the fixed codebook vector and the adaptive codebook vector. The amount of reducing the adaptive codebook gain may be varied.

Abstract: In one aspect, a device includes an encoder configured to receive a speech sample and generate an encoded voice packet from the speech sample, the encoded voice packet having a packet size and a plurality of bytes. The device also includes an encryption unit configured to receive a voice block and generate an encrypted voice block, the voice block having a block size, wherein the packet size is not divisible by the block size and yields a remainder. The device further includes a packet block manager configured to divide the encoded voice packet into a plurality of the voice blocks and provide the plurality of the voice blocks to the encryption unit, the packet block manager further configured to create a remainder voice block including remainder bytes of the encoded voice packet and additional bytes from the encrypted voice block and provide the remainder voice block to the encryption unit.

Abstract: There is provided a pitch lag predictor for use by a speech decoder to generate a predicted pitch lag parameter. The pitch lag predictor comprises a summation calculator configured to generate a first summation based on a plurality of previous pitch lag parameters, and a second summation based on a plurality of previous pitch lag parameters and a position of each of the plurality of previous pitch lag parameters with respect to the predicted pitch lag parameter; a coefficient calculator configured to generate a first coefficient using a first equation based on the first summation and the second summation, and a second coefficient using a second equation based on the first summation and the second summation, wherein the first equation is different than the second equation; and a predictor configured to generate the predicted pitch lag parameter based on the first coefficient and the second coefficient.

Abstract: There is provided a method of using a processing circuitry for selecting a preferential pitch lag value from a plurality of pitch lag values, including a first pitch lag value and a second pitch lag value, for coding an input speech signal. The method comprises determining a first timing relationship between a previous pitch lag value and at least one of the plurality of pitch lag values; determining a second timing relationship between the first pitch lag value and the second pitch lag value; favoring one of the first pitch lag value and the second pitch lag value based on the first timing relationship and the second timing relationship to select one of the first pitch lag value and the second pitch lag value as the preferential pitch lag value; and converting the input speech signal into an encoded speech using the preferential pitch lag value.

Abstract: There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1?k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

Abstract: Various systems and methods are provided for biasing a laser. In one example, a feedback circuit establishes a bias current in a laser. A stabilizing capacitor is coupled to the feedback circuit, the stabilizing capacitor slowing the response of the feedback circuit to a feedback input. A capacitor charging circuit is included in the feedback circuit. The capacitor charging circuit is activated upon a start-up of the feedback circuit to charge the stabilizing capacitor. Thereafter, the capacitor charging circuit is deactivated when a charge on the stabilizing capacitor reaches a predefined charge threshold.

Abstract: A system and method for enhancing the dynamic range of a receiver without degrading the signal to noise ratio of an incoming signal is disclosed. In one embodiment, an enhanced dynamic range receiver for receiving an incoming signal and processing the incoming signal comprises a plurality of VGAs alternately connected in series with a plurality of equalizers to form a VGA-equalizer chain, the VGA-equalizer chain adapted to receive the incoming signal and operable to generate a first analog signal, a first PD adapted to receive the first analog signal and operable to adjust the equalizers' coefficient values, and a second PD adapted to receive a second analog signal from a selected node in the VGA-equalizer chain and operable to adjust the gain of the VGAs.

Abstract: Various systems and methods for buffer memory management are disclosed. In one embodiment a buffer memory includes at least one queue configured to store a number of buffer access tasks. Buffer reclamation logic is executed to free at least one segment of the buffer memory holding an amount of stale data. Buffer reclamation logic is also included that enables the buffer reclamation logic to submit a buffer access task to the buffer memory based upon a total number of the buffer access tasks stored in the at least one queue.

Abstract: An alternate multi-thread pipeline structure and method are provided. A deep pipeline is provided in which two threads of two separate pipeline stages are alternatively presented to the various logic and latch circuits for execution. The execution and latching of the threads alternates from one thread to the other within a single clock cycle. Thus, each thread is executed once per clock cycle and two threads are executed in a single clock cycle.

Abstract: There is provided a method for use by a speech encoder to encode an input speech signal.

Abstract: A modem detection method is provided for use by a first gateway, the first gateway capable of communicating with a second gateway over a packet network. The method comprises setting the first gateway to a voice mode having a first coding scheme, encoding an input signal using the first coding scheme to generate a first encoded input signal, transmitting the first encoded input signal to the second gateway over the packet network, looking for a first tone in the input signal while the first gateway is in the voice mode, wherein the first tone is indicative of a beginning of a character, detecting the first tone in the input signal; setting the first gateway from the voice mode to a modem mode in response to the detecting; and transmitting the input signal to the second gateway according to the modem mode.

Abstract: An adaptive equalization method and system for automatically adjusting an equalizer to compensate a signal's frequency dependent loss is disclosed. The method comprises setting the equalizer's initial coefficient value, receiving an equalized signal, identifying a positive to zero transition of the equalized signal, generating a common mode voltage from the equalized signal, and determining the equalized signal's tail settling voltage. The method further comprises comparing the equalized signal's tail settling voltage to the common mode voltage, determining if the signal is over equalized or under equalized based on the comparison, and adjusting the equalizer's coefficient value based on whether the signal is under equalized or over equalized.

Abstract: An apparatus and method for configuring a gateway device for a modem over packet network session is disclosed. The method for configuring the first gateway device generally comprises: negotiating with a first client device to establish a first link; receiving, from a second gateway device, data indicative of a second link establish between the second gateway device and a second client device; and configuring the first gateway device to transmit data received from the first client device to the second gateway device using a reliable transport and to receive data from the second gateway device using an unreliable transport if the first link is established with error correction and the second link is established without error correction. In some embodiments the error correction provides flow control between the first gateway device and the first client device. The error correction may be based on MNP or V.42 error correction protocol.

Abstract: A method and apparatus is disclosed for optic signal power control to maintain a desired or optimum optic signal power level. During start-up, a default or target value from memory may be utilized to bias or otherwise control operation of an optic signal generator or driver. During operation, pre-stored values may continue to be utilized or an open loop or closed loop control system may be utilized. An open loop control system may incorporate a temperature module or a timer module to account for changes in environment or changes due to aging that may undesirably affect system operation. A closed loop control system may incorporate one or more feedback loops that generate a compensation value to account for detected changes. In one configuration one or more peak values of the actual optic signal, or a portion thereof, are detected and processed to generate the compensation signal.