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

Abstract: Circuitry to free the core processor from performing the explicit read operation required to read data into the internal register set. The processor's register set is expanded and a “shadow register” set is provided. While the core processor is processing one event the “context” and “data” and other associated information for the next event is loaded into the shadow register set. When the core processor finishes processing an event, the core processor switches to the shadow register set and it can begin processing the next event immediately. With short service routines, there might not be time to fully pre-fetch the “context” and “data” associated with the next event before the current event ends. In this case, the core processor still starts processing the next event and the pre-fetch continues during the event processing.

Abstract: In one disclosed embodiment an instruction loop having at least one instruction is identified. For example, each instruction can be a VLIW packet comprised of several individual instructions. The instructions of the instruction loop are fetched from a program memory. The instructions are then stored in a register queue. For example, the register queue can be implemented with a head pointer which is adjusted to select a register in which to write each instruction that is fetched. It is then determined whether the processor requires execution of the instruction loop, for example, by checking a program counter (PC) value corresponding to each instruction. When the processor requires execution of the instruction loop, the instructions are output from the register queue. For example, the register queue can be implemented with an access pointer which is adjusted to select a register from which to output each instruction that is required.

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: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

Abstract: An integrated circuit comprises co-processor circuitry and a core processor. The co-processor circuitry comprises context buffers and data buffers. The co-processor circuitry receives and stores one of the communication packets in one of the data buffers. The co-processor circuitry correlates the one communication packet with one of a plurality of channel descriptors. The co-processor circuitry associates the one data buffer with one of the context buffers holding the one channel descriptor to maintain the correlation between the one communication packet and the one channel descriptor. The co-processor circuitry prevents multiple valid copies of the one channel descriptor from existing in the context buffers. In some examples of the invention, this is accomplished by tracking a number of the data buffers associated with the one context buffer.

Abstract: A speech communication system is provided that uses pitch information, pitch lags, pitch gains, energy and/or other speech characteristics about the outgoing speech and the unknown signal on a frame basis to determine if the unknown signal is an echo signal of the outgoing speech or if the unknown signal also contains speech from a second talker (double talk). Additionally, a plurality of frames of these characteristics of the outgoing speech signal and the unknown incoming signal may be buffered so that the analysis and comparison can be made more efficiently and quickly in the frame domain as opposed to a time domain. Multiple characteristics may be optionally weighted and then analyzed. The system and method may further determine a level of confidence, based on any criterion, in the determination that may then be used to adjust the gain of a filter.

Abstract: There is provided system, device and method for use by a first gateway device to enable communication between a first modem and a second modem, the first gateway device being capable of communicating with the first modem over a first communication line, the first gateway device being capable of communicating with a second gateway device over a packet network, the second gateway device being capable of communicating with the second modem over a second communication line. An example method comprises receiving a call request for the first modem from the second gateway device, placing a call to the first modem, detecting an answer tone transmitted from the first modem over the first communication line in response to the call, detecting a phase reversal in the answer tone, and transmitting a message indicative of the phase reversal to the second gateway device over the packet network.

Abstract: Packet processing circuitry comprises a look-up engine and a processor. The look-up engine transfers a selector to a CAM and receives a corresponding result from the CAM. The look-up engine retrieves a context structure from a context memory based on the result and transfers the context structure to the processor. The processor receives and processes the context structure to control handling of the communication packet. The processor modifies the context structure and transfers the modified context structure to the look-up engine. The processor generates an update instruction and transfers the update instruction to the look-up engine. The look-up engine receives the update instruction and the modified context structure. The look-up engine automatically writes the modified context structure to the context memory in response to the update instruction.

Abstract: An approach for improving quality of speech synthesized using analysis-by-synthesis (ABS) coders is presented. An unstable perceptual quality in analysis-by-synthesis type speech coding (e.g. CELP) may occur because the periodicity degree in a voiced speech signal may vary significantly for different segments of the voiced speech. Thus, the present invention uses a voicing index, which may indicate the periodicity degree of the speech signal, to control and improve ABS type speech coding. The voicing index may be used to improve the quality stability by controlling encoder and/or decoder in: fixed-codebook short-term enhancement including the spectrum tilt; perceptual weighting filter; sub-fixed codebook determination; LPC interpolation; fixed-codebook pitch enhancement; post-pitch enhancement; noise injection into the high-frequency band at decoder; LTP Sinc window; signal decomposition, etc.

Abstract: An approach for adaptively adjusting the correlation window for open-loop pitch determination is presented. Correlation between a windowed reference signal (or target signal) and a candidate signal is maximized under most conditions by sliding the reference window by a delta increment in either direction to capture peak energy. The traditional fixed size of the correlation window is maintained. However, the window slides forward and/or backwards to capture peak energy within the window. The position of the adjusting or sliding window is allowed to shift in a small range or increment in either direction to maximize the energy of the windowed signal thus making sure that at least one peak energy is captured within the window.

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: 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: A content-addressable memory is comprised of processing logic and selector logic. The processing logic receives a first selector including packet header information from the packet processing circuitry. The processing logic transfers the first selector to the selector logic. The processing logic generates additional selectors and transfers the additional selectors to the selector logic. The selector logic receives and processes selectors for matches and provides results corresponding to the matches. The processing logic receives the results from the selector logic and transfers at least some of the results that point to packet processing context structures to the packet processing circuitry.

Abstract: In accordance with one aspect of the present invention, the detection of voice signals versus modem signals in a data communication system may be conducted, a determination can be made whether existing VoIP algorithm functions or characteristics may be appropriate for modem communications, and then the VoIP algorithms may be modified to improve operating conditions for a modem connection. Additionally, the existing VoIP algorithm functions and network characteristics within the data communication system that may degrade modem performance can be identified prior to the determination of whether the VoIP algorithm functions are appropriate for modem communications. In accordance with another aspect, the detection between voice signals and modem signals can also comprise the determination as to a particular type of modems signal. Further, control of the adaptive jitter buffer to remove or reduce adaptation during detection of modem signals can be provided.

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: A convolutional interleaver includes an interleaver memory partitioned into a plurality of circular buffers, wherein each of the circular buffers has associated write pointers and read pointers, and wherein the interleaver is configured to selectively read symbols from an input vector and store the input symbols in the interleaver memory in accordance with the write pointers, and to selectively read symbols from the interleaver memory to form an output vector in accordance with the read pointers. In one aspect, symbols are written to the interleaver prior to reading; in another, the position of the write pointer corresponds to the position of the read pointer within the circular buffer, and symbols are read from said interleaver memory prior to writing. In another aspect, a de-interleaver applies the concepts and algorithms described above in an inverse manner.

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 speech coding system includes an adaptive codebook containing excitation vector data associated with corresponding adaptive codebook indices (e.g., pitch lags). Different excitation vectors in the adaptive codebook have distinct corresponding resolution levels. The resolution levels include a first resolution range of continuously variable or finely variable resolution levels. A gain adjuster scales a selected excitation vector data or preferential excitation vector data from the adaptive codebook. A synthesis filter synthesizes a synthesized speech signal in response to an input of the scaled excitation vector data. The speech coding system may be applied to an encoder, a decoder, or both.

Abstract: An improved data communication technique may be employed with modems through a packet network. The disclosed technique facilitates a virtual end-to-end connection between two modems such that the two modems can effectively behave as if directly connected to each other, unaware of any modifications to the data being transferred or to the protocols configured within the communication connection. Preferably, a data communication system for carrying out the communication technique demodulates data coming from a first modem, transports the demodulated data in packets between two gateways, and then remodulates the data before delivering to a second modem at the other end.