Abstract: An optical logic circuit is disclosed. The optical logic circuit is configured on a substrate of a first material. The optical logic circuit also has an optical layer which overlays the substrate layer and is at least partially configured of a second material. The optical layer is patterned to provide a plurality of optical pathways. At least one of the optical pathways transmits an optical bias and at least one optical pathway is configured to provide an optical input and at least one optical pathway is configured to provide an optical output. The optical pathways are configured to provide a Boolean logic output based on the at least one optical input.

Abstract: A method and apparatus configured to transmit module data between optic modules over a primary communication channel, such as an optic fiber configured to carry network data or outgoing data. A control center may send the module data, such as any type of DDMI data, over the optic fiber to control one or more aspects of the optic module system. The optic module may also be configured to send module data regarding any aspect of module status or operation, to a control center, via the optic fiber. Use of the primary communication channel, normally reserved for only network data, allow for module to module communication or module to control center communication without need of cumbersome two wire interface or supplement channels. Optic module data communication may occur concurrent with network data transmission. Optic module data back-up may occur via the optic channel to provide rapid re-load of important optic module data.

Abstract: According to one aspect, a method of switching a first gateway from a voice mode to a facsimile mode comprises: configuring the first gateway to the voice mode for communication with a second gateway over a packet network, receiving a plurality of data packets from the second gateway over the packet network, analyzing one or more of the plurality of data packets, such as UDP packets, and configuring the first gateway to the facsimile mode if the analyzing determines that the one or more of the plurality of data packets carry facsimile data packets. The analyzing may include calculating a length of the UDP payload in accordance with UDPTL packet structure, and deciding the UDP payload includes a UDPTL packet if the calculated length is equal to UDP payload length, as indicated in the UDP header.

Abstract: There is provided a method of detecting an infinite echo return loss (ERL) in an echo cancellation system while in a finite ERL mode. The method comprises determining a running mean attenuation by the echo cancellation system, determining a ratio of an echo signal to a near-end noise ratio (ENR), defining an infinite ERL threshold (THinfinite) as a function of the ENR, and switching to an infinite ERL mode as a function of the running mean attenuation and the THinfinite. The running mean attenuation can be enhanced echo return loss (ERLE), and the higher the ENR the higher the THinfinite and the lower the ENR the lower the THinfinite. The switching can further be a function of an energy distribution, where the switching switches to the infinite ERL mode based on a non-localized energy distribution over an echo path delay for a predetermined period of time.

Abstract: There is provided a method of detecting and reporting poor voice quality for use by a gateway device. The method comprises facilitating a connection between a telephone and a remote telephone via a network, and detecting a poor voice quality indictor during the connection. The method further comprises capturing, for a pre-determined period of time, telephone voice data being exchanged between the gateway and the telephone, network voice data being exchanged between the gateway and the network, and gateway parameters. The method also comprises packetizing the telephone voice data, the network voice data and the gateway parameters into a plurality packets having a network address of a network storage, and transmitting the plurality packets destined for the network storage via the network. In one aspect, the poor voice quality indictor may be generated by a user of the telephone in response to a poor voice quality of the connection.

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.