Abstract: A facsimile system is disclosed for transferring a facsimile using a Transmission Control Protocol/Internet Protocol (TCP/IP) network. The facsimile system comprises a first communication processing system and a second communication processing system. The first communication processing system converts the facsimile into application packets that indicate individual application packet lengths. The first communication processing system then converts the application packets into TCP/IP packets and transfers the TCP/IP packets to the TCP/IP network. The second communication processing system receives the transferred TCP/IP packets from the TCP/IP network and converts the transferred TCP/IP packets into transferred application packets. The second communication processing system converts the transferred application packets into the facsimile using the individual application packet lengths.

Abstract: A system and method for training an equalizer structure of a digital data communication system in order to compensate for transmission impairments on the line particularly wherein one transceiver of the two is resource limited. In an exemplary embodiment, a line card provides equalization feedback to a modem whenever changes to the equalization are beneficial. The line card calculates a limited number of tap correction factors at one time, transfers the tap correction factors to the modem, and then trains up a new set of tap correction factors. The modem incorporates the tap correction factors into the taps of the corresponding frequency ranges. The process iterates indefinitely through the transmission resulting in a very high quality equalization.

Abstract: A speech encoder comprises an encoding element for encoding a noise reduced speech signal, and a noise suppression element that takes a noisy speech signal and generates the noise reduced speech signal by maximizing the signal to noise ratio (SNR) of the noisy speech signal without suppressing the voiced speech components of the noisy speech signal. The noise suppression element may use harmonic modeling techniques that maximize the SNR in each sub-band of the noisy speech signal by reconstructing the voiced speech components of the noisy voiced speech signal emphasizing harmonic frequencies within each sub-band. The SNR is further maximized by eliminating noise components between signal peaks at the harmonic frequencies, and eliminating noise at signal peaks at the harmonic frequencies by smoothing harmonic parameters generated by the reconstruction of the voiced speech components of the noisy speech signal.

Abstract: In exemplary embodiments, a dataport in an access server is suitably configured to emulate a modern connection for non-modem calls such as voice calls. Computer applications for administering, for example, voice-over-network services thus communicate with the port via standard modem calls. In accordance with one aspect of the invention, communications between the port hardware and the voice enabled application take place through an encapsulating protocol such as the Point-to-Point-Protocol (PPP)such that the port is addressable and able to receive controls or other instructions from a voice application residing on the access server or from a distributed location.

Abstract: There are provided speech coding methods and systems for estimating a plurality of speech parameters of a speech signal for coding the speech signal using one of a plurality of speech coding algorithms, the plurality of speech parameters includes pitch information, the plurality of speech parameters is calculated using a plurality of thresholds. An example method includes estimating a background noise level in the speech signal to determine a signal to noise ratio (SNR) for the speech signal, adjusting one or more of the plurality of thresholds based on the SNR to generate one or more SNR adjusted thresholds, analyzing the speech signal to extract the pitch information using the one or more SNR adjusted thresholds, and repeating the estimating, the adjusting and the analyzing to code the speech signal using one the plurality of speech coding algorithms.

Abstract: Packet processing circuitry comprises a look-up engine and a processor. The look-up engine transfers a first selector to a CAM and receives a corresponding first result from the CAM. The look-up engine retrieves a first context structure based on the first result. The look-up engine builds a summation block using the first context structure and transfers the summation block. The processor receives and processes the summation block to control handling of the communication packet.

Abstract: An integrated circuit processes a communication packet and comprises a core processor and scheduling circuitry. The core processor executes a software application that directs the core processor to process the communication packet. The scheduling circuitry retrieves first scheduling parameters cached in a context buffer for the packet and executes a first algorithm based on the first scheduling parameters to schedule subsequent transmission of the communication packet.

Abstract: A method and system for performing sequence time domain reflectometry over a communication channel to determine the location of line anomalies in the communication channel is disclosed. In one embodiment, the system generates a sequence signal and transmits the sequence signal over an optical channel. The system receives one or more reflection signals over the optical channel and performs reflection signal processing on the reflection signal. In one embodiment, the optical reflection is transformed to an electrical signal and correlated with the original sequence signal to generate a correlated signal. The time between the start of the reflection signal and a subsequent point of correlation and the rate of propagation reveals a line anomaly location. A circulator, beam splitter, or any other similar device may direct the reflection signal to the apparatus configured to perform reflection signal processing.

Abstract: A method and apparatus is disclosed for adaptively determining threshold values in a decision device, such as a decision slicer. In one embodiment, a slicer receives one or more updated threshold values during operation to adaptively accommodate changes in the received signal or the channel that may occur over time. The updated threshold value is based on a MIN value and a MAX value. The MIN values represent a value related to a recently received input that was determined to qualify for a particular slicer output and was less than the particular slicer output value. The MAX values represent a value related to a recently received input that was determined to qualify for a particular slicer output and was greater than the particular slicer output value. In one embodiment, the MAX and MIN value computation is based on scaling factors that dampen the rate of change of the threshold value.

Abstract: An exemplary multi-channel speech processor comprises a controller capable of interfacing with a plurality of channels, and at least one signal processing unit (SPU) coupled to the controller, where the multi-channel speech processor has a maximum execution time for processing all frames, one channel at a time, by processing a single frame from each of the plurality of channels. The signal processing unit encodes each of the single frames from each of the plurality of channels, one channel at a time, to generate encoded frames until the maximum execution time elapses or is about to elapse. The controller also transmits a pre-determined frame for each of the plurality of channels not processed during the encoding step, due to the maximum execution time elapsing or being about to elapse, such that the predetermined frame causes a decoder which receives the predetermined frame to generate a frame erase frame.

Abstract: A link layer controller comprises a network layer interface, a physical layer interface, and a memory controller. The network layer interface exchanges packets with the network layer system and transfers a status signal to the network layer system. The physical layer interface exchanges the packets with the physical layer system. The memory controller exchanges the packets with the network layer interface, a memory, and the physical layer interface. The memory controller also generates the status signal to indicate available space in the memory.

Abstract: An input signal enters a noise suppression system in a time domain and is converted to a frequency domain. The noise suppression system then estimates a signal to noise ratio of the frequency domain signal. Next, a signal gain is calculated based on the estimated signal to noise ratio and a voicing parameter. The voicing parameter may be determined based on the frequency domain signal or may be determined based on a signal ahead of the frequency domain signal with respect to time. In that event, the voicing parameter is fed back to the noise suppression system, for example, by a speech coder, to calculate the signal gain. After calculating the gain, the noise suppression system modifies the signal using the calculated gain to enhance the signal quality. The modified signal may further be converted from the frequency domain back to the time domain for speech coding.

Abstract: A flexible variable rate vocoder and related method of operation. The vocoder selects a target average data rate responsive to at least one network parameter and at least one external parameter.

Abstract: The invention provides a speech coding system with input signal transformation that may reduce or essentially eliminate “silence noise” from the input or speech signal. The speech coding system may comprise an encoder disposed to receive an input signal. The encoder ramps the input signal to a zero-level when a portion of the input signal comprises silence noise.

Abstract: In a coding procedure, coding parameters are selected for coding the speech signal to achieve enhanced perceptual quality of reproduced speech. At least one coding parameter value or preferential coding parameter value is selected to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

Abstract: Packet processing circuitry comprises a processor and a look-up engine. The look-up engine transfers a first selector to a CAM and receives a corresponding first result from the CAM. The look-up engine generates a second selector based on the first result. The look-up engine transfers the second selector to the CAM and receives a corresponding second result from the CAM. The look-up engine retrieves a first context structure based on the second result. The look-up engine builds a summation block using the first context structure and transfers the summation block to the processor. The processor receives and processes the summation block to control handling of the communication packet.

Abstract: A signal processing system is well suited for conditioning a speech signal prior to coding the speech signal to achieve enhanced perceptual quality of reproduced speech. The signal processing system may be incorporated into mobile or portable wireless communications devices, wireless infrastructure equipment, or both. The signal processing system includes a filtering arrangement for filtering an input speech signal to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

Abstract: A data communications system and method are provided, for example, in a facsimile machine to establish a data link with a gateway over a local loop. The data link is established employing pulse code modulation with prioritization in the upstream direction. Alternatively, a modulation type may be employed that is equivalent to pulse code modulation that provides similar or higher speed data communication. By using pulse code modulation in the upstream direction, the resulting facsimile transmission in accomplished with much greater speed.

Abstract: A speech encoder that analyzes and classifies each frame of speech as being periodic-like speech or non-periodic like speech where the speech encoder performs a different gain quantization process depending if the speech is periodic or not. If the speech is periodic, the improved speech encoder obtains the pitch gains from the unquantized weighted speech signal and performs a pre- vector quantization of the adaptive codebook gain Gp for each subframe of the frame before subframe processing begins and a closed-loop delayed decision vector quantization of the fixed codebook gain GC. If the frame of speech is non-periodic, the speech encoder may use any known method of gain quantization.

Abstract: The time domain equalizer filter of the present invention increases the data rate of a communications system while shortening a channel. Such a time domain equalizer filter is spectrally flat and the central tap is constrained to a non-zero real number. The error between the data filtered by the time domain equalizer filter and the data filtered by a target filter is reduced by adapting the time domain equalizer filter and the target filter. Adaptation of the time domain equalizer filter and the target filter may be accomplished by calculating new tap values for each filter while constraining the central tap of the time domain equalizer filter. In this way, the error between the data filtered by the time domain equalizer filter and the data filtered by the target filter may be reduced, so that a shortened channel may be provided. Accordingly, by shortening the channel and balancing the use of a spectrally flat time domain equalizer filter, the data rate of the communications system may be increased.