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: 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: 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: 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.

Abstract: A method of speech encoding comprises generating a first synthesized speech signal from a first excitation signal, weighting the first synthesized speech signal using a first error weighting filter to generate a first weighted speech signal, generating a second synthesized speech signal from a second excitation signal, weighting the second synthesized speech signal using a second error weighting filter to generate a second weighted speech signal, and generating an error signal using the first weighted speech signal and the second weighted speech signal, wherein the first error weighting filter is different from the second error weighting filter. The method may further generate the error signal by weighting the speech signal using a third error weighting filter to generate a third weighted speech signal, and subtracting the first weighted speech signal and the second weighted speech signal from the third weighted speech signal to generate the error signal.

Abstract: A fast startup procedure for a modem system utilizes known characteristics of a previously established communication channel to reduce the initialization period associated with subsequent connections over the same channel. In response to the establishment of a call, the modem devices determine whether the fast connect protocol is supported. If so, then the called modem transmits a modified answer tone to the calling modem. The calling modem analyzes the signal received in response to the modified answer tone to determine whether characteristics of the current channel are similar to stored characteristics associated with a previous connection over the same channel. If a channel “match” is found, then the modem devices carry out a fast initialization routine that eliminates, abbreviates, or modifies a number of procedures or protocols that are carried out in conventional modem startup processes.

Abstract: There is provided rate selection methods and systems for selecting coding rates for coding frames of a speech signal to realize an average bit rate indicated by a mode. For example, a mode 0, mode 1, and a mode 2 may be defined, with each mode requiring a different average bit rate. To achieve the average bit rate of a particular mode, a coding rate is selected for each frame of the speech signal, based on the characteristics of a frame. A frame can be categorized in a class, such as noise or silence, noise-like unvoiced speech, pulse-like unvoiced speech, transition into voiced speech, unstable voiced speech, stable voiced speech. Other parameters may also be used, such as the sharpness, noise-to-signal ratio, pitch correlation, energy, and reflection coefficient. A frame may then be coded at a full-rate, a half-rate, a quarter-rate, or an eighth-rate.

Abstract: Optical transducers disposed on an integrated circuit correspond to test points thereon. The state of optical transducers perceived by an optical sensor is correlated to test points. Alignment of the optical sensor relative to the optical transducers is accomplished electronically or mechanically.

Abstract: A system and method for performing power management in a communication system without losing data is disclosed. The G.Lite system is an exemplary system suitable for implementing the present invention because it includes at least one operational state and an idle state in which no data is transmitted. The G.Lite standard defines ATU states. The defined states are full on state L0, low power state L1 and idle state L3. In L0 state, the ADSL link is fully operational at full data rate. In L1 state, the ADSL link is fully operational at a lower data rate. In L3 state, no signal is transmitted. Because no signal is transmitted, no idle cells are transmitted. This will result in losing the cell delineation and ATM link connectivity. Therefore, when data is received for an ATU in L3 state, data cells may be lost.

Abstract: An integrated circuit processes communication packets and comprises a pointer cache and control logic. The pointer cache store pointers that correspond to external buffers that are external to the integrated circuit and configured to store the communication packets. The control logic allocates the external buffers as the corresponding pointers are read from the pointer cache and de-allocates the external buffers as the corresponding pointers are written back to the pointer cache.

Abstract: Boundary scan cells for driving internal logic and sensing internal logic of integrated circuit use external clocks synchronized with internal functional clocks. Synchronized clocks enable synchronous sampling of internal signals and synchronized of injection signals into a functional portion of the integrated circuit.

Abstract: A method and apparatus for early detection of in-band DTMF signals in voice transmissions over an IP network are disclosed. Using a digital signal processor, whenever any indication of signals possibly corresponding to the signals characteristic of DTMF tones are detected, packets of data that would otherwise be sent over the IP network are delayed and temporarily stored in a buffer. Buffering the data packets provides a period of time for digital signal processing techniques to reliably determine whether the characteristics of the detected signal are sufficient to characterize it as a DTMF signal. If a determination is made that the signal has the necessary characteristics of a valid DTMF signal, the buffered packets are discarded and a special control packet is sent formatted to convey to the recipient characteristic information for the detected DTMF signal sufficient to allow the DTMF signal to be generated by a DTMF generator.

Abstract: A system for executing computer programs on a target platform having a limited amount of memory. Directives are suitably placed in the computer program source code at natural boundaries. The system uses the directives to extract structural information from the computer program and to produce a description of all program objects; to estimate typical object usage; and, to trigger transparent object paging to and from the limited platform memory during execution of the computer program. The system makes paging decisions prior to runtime by using relevant factors such as the typical usage of program objects and the size of each program object.

Abstract: An integrated system comprises an authorization device that generates a second encrypted data stream in response to a first data stream, and a programmable logic device (PLD) that generates a third encrypted data stream in response to the first data stream, while simultaneously operating under at least partial control of program code during a first time interval This third encrypted data stream is preferably generated internal to the programmable logic device. Authorization detection circuitry is also preferably provided that compares the second and third encrypted data streams at least periodically during the first time interval. This circuitry may also disable operation of the programmable logic device if the second and third encrypted data streams indicate that the programmable gate array is not authorized to use the program code. The authorization detection circuitry is preferably provided within the programmable logic device and may utilize at least a portion of the proprietary program code (e.g.

Abstract: An approach to reduce the quality impact due to lost voiced frame data is presented. The decoder reconstructs the lost frame using the pitch track from a directly prior frame. When the decoder receives the next frame data, it makes a copy of the reconstructed frame data and continuously time warping it and the received frame data so that the peaks of their pitch cycles coincide. Subsequently, the decoder fades out the time-warped reconstructed frame data while fading in the time-warped received frame data. Meanwhile, the endpoint of the received frame data remains fixed to preclude discontinuity with the subsequent frame.

Abstract: A speech coding system is provided with time-domain noise attenuation. The speech coding system has an encoder operatively connected to a decoder via a communication medium. A preprocessor processes a digitized speech signal from an analog-to-digital converter. Speech coding systems are used to encode and decode a bitstream. Gains from the speech coding are adjusted by a gain factor Gf that provides time-domain background noise attenuation.

Abstract: Methods and systems for matching the communication protocol of a calling device with one of a plurality of communication protocols supported by an answering device are provided. Initially, a predetermined amount of data is received and buffered in a memory component. Following, an analysis is performed to match the received data with one of the communication protocols supported by the answering device. Communication protocols that can be detected and matched include, for example, the V.110, V.120 and ISDN protocols. During the buffering and analyzing steps, a predetermined data pattern such as mark idle can be transmitted to the calling device. Thereafter, the answering device is configured to answer the call according to the protocol matched in the analyzing step. A timer can be utilized to limit the period for matching to a predetermined length, after which time the answering device can be configured to answer the call in analog mode.

Abstract: There is provided a speech encoder comprising a first weighting means for performing an error weighting on a speech input. The first weighting means is configured to reduce an error signal resulting from a difference between a first synthesized speech signal and the speech input. In addition, the speech encoder includes a means for generating the first synthesized speech signal from a first excitation signal, and a second weighting means for performing an error weighting on the first synthesized speech signal. The second weighting means is also configured to reduce the error signal resulting from the difference between the speech input and the first synthesized speech signal. There is also included a first difference means for taking the difference between the first synthesized speech signal and the speech input, where the first difference means is configured to produce a first weighted error signal.

Abstract: A status queue in a host and a control queue in a segmentation and reassembly (SAR) subsystem are on opposite sides of a host bus in a control plane. Buffer descriptors in the host and the SAR and buffers in the host are in a data plane. To transfer cell payloads to a first line interfacing the SAR, the host writes the SAR that it has such cell payloads. The host writes the host buffer descriptors into the control queue to obtain the transfer of the buffer payload to the first line. The SAR writes the status queue when the transfer has been completed. To transfer cell payloads to the host memory, the host writes into the control queue the address of the buffers to receive the payload from the SAR. The SAR then writes the buffer descriptors to the status queue to obtain the transfer of the cell payloads to the buffers. Each of the control and status queues may be respectively considered to constitute two (2) control queues and two (2) status queues.