Abstract: A method and apparatus is disclosed for reducing the error rate in a received signal by determining and establishing an optimal slice point for a decision device, or optimal operational parameters. In one embodiment, a processor monitors a received signal to determine signal characteristics, such as a peak signal level or phase value. A table look-up operation may occur in a data table or other processing may occur and, based on the signal characteristics, an optimal slice point may be determined. In one embodiment the look-up operation may also reveal one or more optimal operational parameters that, if adopted, will further reduce the error rate. A receiving station may communicate these optimal operational parameters to a transmitting station to modify operation of the transmitting station. Also disclosed is a method and apparatus for self-testing a communication system and channel to determine optimal slice points and operational parameters.

Abstract: In one exemplary embodiment, the disclosed VLIW processor comprises a number of threads where each thread includes a processing unit. For example, there can be two threads, where each of the two threads has its own processing unit. According to this exemplary embodiment, a number of VLIW packets are divided into a number of issue groups. As an example, two VLIW packets are divided into two issue groups each. The first issue group in the first VLIW packet is provided to a first thread for execution in the first thread processing unit during a first clock cycle. Concurrently, the first issue group in the second VLIW packet is provided to a second thread for execution in the second thread processing unit during the same clock cycle, i.e. during the first clock cycle. Moreover, the second issue group in the first VLIW packet is provided to the first thread for execution in the first thread processing unit during a second clock cycle.

Abstract: Debugging a multiprocessor system utilizing at least two debuggers is disclosed. A user command is issued through a first debugger to a first processor in the multiprocessor system. The user command can be converted to a debugger RPC command by a first proxy module associated with the first debugger. The user command can be intercepted by a scheduler. A second processor in the multiprocessor system is then informed, for example by the scheduler, of the issuance of the user command. The user command is executed by both the first and second processors and both the first and second debuggers can then be informed by means of processor RPC commands of the execution of the user command. The processor RPC commands can be converted to callback commands by respective first and second proxy modules associated with the first and second processors when being sent to the first and second debuggers.

Abstract: A method for detecting music in a speech signal having a plurality of frames. The method comprises defining a music threshold value for a first parameter extracted from a frame of the speech signal, defining a background noise threshold value for the first parameter, and defining an unsure threshold value for the first parameter. The unsure threshold value falls between the music threshold value and the background noise threshold value. If the first parameter falls between the music threshold value and the background noise threshold value, the speech signal is classified as music or background noise based on analyzing a plurality of first parameters extracted from the plurality of frames.

Abstract: Various media stream processing systems and methods are provided. In one exemplary embodiment, a media stream processing method is provided that includes the brokering of a number of communications channels between a communications network and a packet switched network with a number of media stream processors. Each of the communications channels is specified by a profile having a number of channel parameters. The method also includes storing each of the profiles in a standby media stream processor when each of the media stream processors is operational. Finally, the method includes brokering respective ones of the communications channels originally brokered by one of the media stream processors with the standby media stream processor upon a failure of the one of the media stream processors.

Abstract: Speech coding systems include multi-rate speech codecs having an encoder and a decoder. Silence description coding for multi-rate speech coding systems that employ discontinued transmission is performed in either the encoder or the decoder of the multi-rate speech codec. It may also be performed in a distributed manner wherein it is performed partially in the encoder and partially in the decoder. The silence description coding is performed on a speech signal having a substantially non-speech-like characteristic. Voice activity detection classifies the speech signal as being either substantially speech-like or substantially non-speech-like. The silence description coding is selected from a plurality of coding modes. In certain embodiments of the invention, the silence description coding is a source coding mode that operates at a bit rate that fits within a bit rate budget as determined by all of the available source coding modes within the plurality of coding modes.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codec are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech. The overall quality of the system is strongly related to the excitation. In order to enhance the excitation, the system contains a fixed codebook comprising several subcodebooks.

Abstract: A method of automatically translating an opcode summary table into an Architecture Description Language (“ADL”) can be employed to efficiently design and test a microprocessor instruction set. An opcode summary table is analyzed and code is generated in an ADL. The generated code can be optimized by first analyzing the opcode summary table to find groupings and sub-groupings of instructions based on similarities. The optimized code can be generated by first generating code for common elements within the sub-group, then generating code for each instruction within the sub-group. This process would repeated for each group in the opcode summary table. The result is an ADL description of the opcode summary table.

Abstract: A system including an adaptive code book and a fixed code book for code excited linear prediction coding of speech signals is provided. The invention includes an embedded adaptive code book in the fixed code book and the selection procedure for selecting excitation vector parameters. A code book update system updates the fixed code book with embedded adaptive code book based on the long term processing excitation vector parameters with previous synthesized excitation.

Abstract: In one embodiment, move buses utilized in presently known VLIW processors are eliminated and replaced with a busing scheme which results in transfer of operands from each register file bank to any data path block while also reducing the total bus width and total power consumption associated with transport of operands from register file banks to data path blocks. According to this busing scheme, the speed of VLIW processor is also improved since the need for one clock cycle to move operands from one register file bank to another is overcome. In another embodiment, a scheduling restriction is used to eliminate the need for the presently required write back buses used by various data path blocks. In yet another embodiment, a scheduling restriction is imposed which results in a reduction of the number of ports, a reduction in the width of buses, and a reduction of power consumption.

Abstract: An integrated circuit for processing communication packets having separate data buffers and separate state information buffers. Each data buffer and each state information buffer (hereinafter termed resources) has an associated in-use counter. Multiple events can share the same resource. The counter associated with a resource is incremented when a resource becomes associated with a particular event. The counter associated with a resource is decremented when an event completes the use of that particular resource. When the in-use counter for a resource becomes zero, the in-use counter indicates that the resource is unassigned and that the resource can be assigned to a new event.

Abstract: Ring access control circuitry comprises a receive interface, a host interface coupled to the receive interface, a mate interface coupled to the host interface, a flow control agent coupled to the mate interface, a transmit interface coupled to the flow control agent, and a wrap path coupled from between the flow control agent and the transmit interface to between the receive interface and the host interface. A wrap is initiated over the wrap path in response to a link failure. The ring being controlled could be a resilient packet ring.

Abstract: A intelligent network interface network port (INIP) is provided that enables a visiting computer to be connected to a local area network (LAN), even though the LAN is not configured to support the visiting computer. Upon detecting one of several specific types of messages, the INIP modifies the message, retransmits the modified message and notifies the visiting computer that the INIP is the appropriate target for the message, thus acting as a relay point between the visiting computer and other devices both on the LAN and accessible through it. This enables the visiting computer to transparently connect to a LAN that is not configured to support it.

Abstract: A fax over Internet protocol client driver. The present invention enables a client to support a fax over Internet protocol. Whereas conventional systems commonly rely upon an Internet service provider (ISP) to support the fax over Internet protocol functionality, the invention ensures that a client can perform multiple Internet sessions, one of which is a fax over Internet protocol session. There is no need to disconnect from a current Internet session, and then re-establish another Internet session. The client is any number of devices including a personal computer, a laptop computer, a pen based computer, an Internet capable fax machine, or any other fax capable device. By providing a fax over Internet protocol driver within the client, the client can send a fax transmission over the Internet, even in the situation where the ISP that the client solicits does not support that particular functionality.

Abstract: A communication system is provided for transferring data through a digital switching network. The communication system includes a client modem, a linecard in communication with the client modem over a local loop, and a linecard modem interfacing with the linecard and the digital switching network. The client modem modulates client data to generate modulated client data for transmission to the linecard over the local loop, and the linecard modem receives the modulated client data from the linecard and demodulates the modulated client data to generate the client data for transmission through the digital switching network. Further, the linecard modem modulates network data from the digital switching network to generate modulated network data for transmission to the client modem over the local loop, and the client modem receives and demodulates the modulated network data to generate the network data.

Abstract: A multi-channel speech processor for encoding speech in a packet network environment is disclosed. In one illustrative aspect, a complexity resource manager (CRM) is executed by a controller or processor. The CRM manages the level of complexity of encoding which is used by a signal processing unit (SPU) to convert the speech signal into packet data. In general, the CRM determines the level of complexity of encoding based on a calculated complexity budget, where the complexity budget is determined based on the time required to process prior speech signal channels and the time available to process the remaining channels. In this way, the CRM is able to control the overall complexity of the speech processor through its ability to signal the SPU to encode speech signal in a complexity reduced mode based on the calculated complexity budget under certain conditions.

Abstract: Various methods and systems for establishing modem relay connections are provided. For example, a modem relay method for use by a first gateway device in communication with a calling modem comprises the exemplary steps of: receiving a call from the calling modem for establishing a communication with a remote modem; requesting a second gateway device to place a call to the remote modem for establishing the communication; transmitting ANSam to the calling modem for a predetermined period of time according to an ANSam message received from the second gateway device in an RFC2833 message via a packet network; monitoring for detection of a CM signal from the calling modem; monitoring for detection of an AA signal from the calling modem; monitoring for receipt of a modulation type message from the second gateway device; and monitoring for receipt of a pass through mode message from the second gateway device.

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 predetermined 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: In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis.

Abstract: Various time-domain noise suppression methods and devices for suppressing a noise signal in a speech signal are provided. For example, a time-domain noise suppression method comprises estimating a plurality of linear prediction coefficients for the speech signal, generating a prediction error estimate based on the plurality of prediction coeficients, generating an estimate of the speech signal based on the plurality of linear prediction coefficients, using a voice activity detector to determine voice activity in the speech signal, updating a plurality of noise parameters based on the prediction error and if the voice activity detector determines no voice activity in the speech signal, generating an estimate of the noise signal based on the plurality of noise parameters, and passing the speech signal through a filter derived from the estimate of the noise signal and the estimate of the speech signal to generate a clean speech signal estimate.