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: 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: 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: 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: A DSL communication system including a DSL transmission unit at a central office (DTU-C) and a DSL transmission unit at a remote location (DTU-R) in communication over a communication link. DTU-C and DTU-R are capable of transmitting and receiving packets of data prior to synchronization or training. DTU-C transmits a discovery message to DTU-R and awaits receiving a discovery response message from DTU-R. After receiving the discovery response message, DTU-C transmits a probe message to DTU-R followed by a probe signal. DTU-R measures the line quality while receiving the probe signal. After receiving the probe signal, DTU-R transmits a probe signal to DTU-C, using which signal DTU-C measures the line quality. After transmitting the probe signal, DTU-R transmits a probe response message, including line quality measurements performed by DTU-R.

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.

Abstract: A data communication circuit includes a decoder and an alignment buffer. The decoder receives and decodes parallel (N) bit channels into parallel (M+X) bit channels with signaling bits that indicate headers in the parallel (M+X) bit channels. The decoder transfers the parallel (M+X) bit channels to the alignment buffer. The alignment buffer recovers and aligns parallel (M) bit channels using the signaling bits. The alignment buffer generates a clock selection signal using the signaling bits. The alignment buffer transfers the aligned parallel (M) bit channels and the clock selection signal. The alignment buffer can have a length that is a multiple of a frame length for the (M) bit parallel channels.

Abstract: Data (e.g. legacy LAN traffic) segmented into packets provide a header and a cell payload for each cell in each packet. The cell payloads are transferred to a region address in a host memory in accordance with determinations by a control memory. When the cell payload is to be transmitted from the host memory, the cell payload for a particular region address is combined with the header stored in the control memory for such address. Streaming data (e.g. voice or video) occurs at a regular rate and is not necessarily broken into packets. The streaming data is segmented to provide cell headers and cell payloads. The cell payloads are then transferred to a host receive FIFO in accordance with a determination by the control memory and are stored in a data sink. Cell payloads from a data source are transferred into a host transmit FIFO at a particular rate and are transferred from the host transmit FIFO preferably at a substantially constant rate higher than the particular rate.

Abstract: A system, method, and signal are disclosed for line probing. The invention provides an improved signal for line probing having desirable characteristics. In various embodiments the signal comprises a periodic sequence having good autocorrelation characteristics. The periodic sequence of the invention provides a faster and more accurate evaluation of the channel. Another embodiment of the invention comprises processing of the received sequence to determine a desired or maximum data rate for the line or channel. The invention also provides an evaluation and calculation of the noise on the line or channel.