Abstract: A communication system for transmitting voice and data information is disclosed. The communication system includes subscriber modems, system converters and at least one system data pump. The system also includes buffer memory interposed, in the upstream direction, between the converters and the data pump to allow subscriber modems to send a predetermined amount of information to the data pump on demand.

Abstract: A three-stage four-dimensional encoding system and method for DMT-based transceivers includes a first stage where a parallel shifter shifts three bits of input data bits from a shadow register into a holding register. Next, a convolutional encoder generates encoded bits v1, v0, w1 and w0, using the three bits in the holding register. The encoded bits w1 and w0 may be registered as w1′ and w0′ for later use. In the second stage, the encoded bits v1 and v0 are provided to the inputs of the parallel shifter for parallel parsing of bits for the first tone of a pair of tones of DMT and the number of bits needed for the first tone are parallel shifted to the holding register. The bits in the holding register are then mapped by a constellation mapper into a constellation point.

Abstract: Various systems and methods are provided in a client, a line card or other conversion device, and a server to implement either a linear or a nonlinear PCM transmission mode for data communication in a data communications network. The client and the server negotiate with each other to establish a specific data communications mode. The data communications mode may be, for example, linear pulse code modulation (PCM) or nonlinear PCM. To begin, the client informs the server whether it supports nonlinear PCM, linear PCM, or both. In light of the clients communications capability and the modes supported by the server itself, the server responds by informing the client which data communications mode is to be used. In the case that both the client and server support linear and nonlinear PCM, the server informs the client that linear PCM is to be used, such being more conducive to accurate high-speed data communication.

Abstract: According to one embodiment, a RAM array includes at least one RAM cell comprising a first access transistor driven by a first word line. When the first access transistor is turned on, it couples the RAM cell to a first bit line. The first bit line is connected to a single-ended sense amplifier such as an inverter. Similarly, the RAM cell comprises second, third, and fourth access transistors driven by respectively second, third, and fourth word lines. When the respective access transistors are turned on, they couple the RAM cell to respectively second, third, and fourth bit lines. The bit lines are connected to respective single-ended sense amplifiers such as inverters. In one embodiment, each of the first, second, third, and fourth access transistors is an NFET. The first, second, third, and fourth bit lines are coupled to respectively first, second, third, and fourth precharge transistors.

Abstract: Voiced speech preprocessing employs waveform interpolation or a harmonic model circuit to smooth a transition region and simplify speech coding. At low bit rates, the speech is coded by a system that maintains a high perceptual quality in the transition region from a voiced (quasi-periodic) portion of the speech signal to an unvoiced (non-periodic) portion of the speech signal. Similarly, the transition region from an unvoiced portion to a voiced portion is conditioned to maintain a high perceptual quality at a low bandwidth. The transition region from one type of voiced region to another type of voiced region is also smoothed. The transition region is smoothed to create a quasi-periodic speech signal.

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 codecs 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.

Abstract: A communication system for transmitting voice and data information is disclosed. The communication system includes a system converter and a system data pump configured to transform and transmit information between a subscriber and a central location. The system converters convert information between appropriate analog and digital formats. The data pump modulates and demodulates information and is configured to operatively couple to one or more system converters.

Abstract: A telephony services distribution system includes a voice/data server that receives and transmits telephony data in accordance with a suitable digital communication protocol. The voice/data server is associated with a customer premise that supports a number of enhanced telephony devices, which may include wired and/or wireless telephone devices, videophone devices, facsimile devices, or the like. The customer premise may also support any number of conventional POTS telephony instruments. The voice/data server and the enhanced telephony devices communicate signaling data therebetween via a signaling technique such as the V.21 modem protocol. The signaling information conveys data associated with the status of the enhanced telephony devices, calling party and called party identification, characteristics of the incoming and outgoing calls, and/or data related to any number of operating parameters and characteristics of the system.

Abstract: In an exemplary conversion scheme, a frame of a first speech signal comprising a plurality of frames encoded at a plurality of first rates, including a first non-speech rate, is received. The rate of the received frame is determined, and if the received frame is encoded at the first non-speech rate, then the received frame is re-encoded at either a second or third non-speech rate to generate a frame of a second speech signal. Moreover, a system for converting a speech signal comprises a receiver for receiving a frame of a first speech signal and a processor capable of determining the encoding rate of the received frame and re-encoding the received frame at either a second or third non-speech rate if the received frame was originally encoded at a first non-speech rate.

Abstract: A debugging environment for a multi-processor simulator or emulator is disclosed. The simulator or emulator is ideally suited for the development of embedded software. The simulator can contain multiple processor models, with each processor model representing a processor. The simulator or emulator also includes a scheduler which controls the execution of the processor models. Each processor also communicates with a debugger via a debug adapter. The debug adapter acts as a pass-through filter for non-control commands which are communicated between a processor and its attached debugger. However, the debug adapter routes control commands to the scheduler. The scheduler ensures that all of the processors and debuggers maintain synchronization. Other modules can also be included in the multi-processor simulation environment, for example, clock gate modules.

Abstract: A speech compression system with a special fixed codebook structure and a new search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A better way is used to calculate a criterion value, minimizing an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Abstract: A bi-directional pitch enhancement system for speech coding systems. As speech data applications continue to operate in areas having intrinsic bandwidth limitations, the perceptual quality of reproduced speech data in typical speech coding systems suffers significantly. The present invention employs forward pitch enhancement and backward pitch enhancement to maintain a high perceptual quality in reproduced speech. In certain embodiments of the invention, the forward pitch enhancement and the backward pitch enhancement are performed in a single portion of the entire speech coding system. For example, in speech codecs, the forward and the backward pitch enhancement are performed only in the speech codec's encoder, or alternatively, only in the speech codec's decoder. If desired, the forward and the backward pitch enhancement are performed in a distributed manner, each being performed, at least in part, in each one of the encoder and the decoder of the speech codec.

Abstract: Cycle and phase accurate hardware-software coverification is disclosed. According to the present disclosure, “cycle accurate” or “phase-accurate” hardware-software coverification is performed while substantially maintaining a high speed for performing the coverification. System activities are divided into a “pre-compute” and a “post-compute” category. Those activities which fall in the pre-compute category are evaluated to ensure that updates to the state of the system are performed on a clock edge immediately following a change in system inputs affecting the state of the system. The state of the system is then updated on the same clock edge at which the pre-compute operation was performed. Those activities which fall in the post-compute category are then evaluated at the same clock edge on which the state of the system was updated.

Abstract: A digitized signal detection system where the bit rate encoding is changed dynamically to provide encoding for different type signals and formats at bit rates optimized to properly reconstruct the input signal whether speech or non-speech and therefore can transfer signals of different character on a frame by frame basis. A change of encoding format can make the system a speech or music recognizer dependent what is to be listened for. Three basic components a recognizer which categorizes the type of input signal, an evaluator which evaluates the category of quality of the reconstructed signal and a recommender which make as recommendation based on the quality to change standards to encode the signals received pursuant to a standard which provides for improved quality. The dynamic signal detector receives the input signal directly and extracts the parameters for evaluation. These parameters are tested and a determination made if a switch of standards are required. To improve the reconstructed signal.

Abstract: A system which compensates for jitter in the transfer of voice data over packet data networks. A predetermined quality of service factor is used to determine the jitter delay for received voice packets in order to optimize the number of voice packets received. The invention uses packet sequence information in the voice packet protocol to determine which voice packets are missing and which voice packets are late within a predetermined statistically significant interval. The jitter delay is decreased when the number of missing packets is significantly less than that specified by the quality of service factor. In most cases, the jitter delay is increased when the number of missing packets is greater than that specified by the quality of service factor. The jitter delay is not increased past a certain value when no late packets are observed during the predetermined interval.

Abstract: The invention provides a speech coding system with a music classifier. An encoder is disposed to receive an input signal and provides a bitstream based upon a speech coding of a portion of the input signal. The encoder provides a classification of the input signal as one of noise, speech, and music. The music classifier analyzes or determines signal properties of the input signal. The music classifier compares the signal properties to thresholds to determine the classification of the input signal.

Abstract: A text-prompted speaker verification system that can be configured by users based on a desired level of security. A user is prompted for a multiple-digit (or multiple-word) password. The number of digits or words used for each password is defined by the system in accordance with a user set preferred level of security. The level of training required by the system is defined by the user in accordance with a preferred level of security. The set of words used to generate passwords can also be user configurable based upon the desired level of security. The level of security associated with the frequency of false accept errors verses false reject errors is user configurable for each particular application.

Abstract: An apparatus and method for issue grouping of instructions in a VLIW processor is disclosed. There can be one, two, or three issue groups (but no greater than three issue groups) in each VLIW packet. In one embodiment, a template in the VLIW packet comprises two issue group end markers where each issue group end marker comprises three bits. The three bits in the first issue group end marker identifies the instruction which is the last instruction in the first issue group. Likewise, the three bits in the second issue group end marker identifies the instruction which is the last instruction in the second issue group. Any instructions in the VLIW packet falling outside the two expressly defined first and second issue groups are placed in a third issue group. As such, three issue groups can be identified by use of the two issue group end markers. In one embodiment, the template of the VLIW packet includes a chaining bit.

Abstract: A packet-based or cell-based data communication system is capable of determining whether a new call is a modem call or a voice call before a modem connection is established between the two end devices. The network switches are configured to monitor for distinctive facsimile modem and data modem calling and answer signals. In response to the detection of such calling or answer signals, the data communication system selects a speech coding scheme and/or a data transmission technique for use in connection with the current call, while a conventional 64 kbps PCM transmission scheme is maintained for communications between the network and the end devices. A speech coding protocol, such as ADPCM, may be enabled for voice calls, thus conserving network bandwidth. The 64 kbps PCM scheme may be employed to facilitate high speed modem transmissions.

Abstract: A speech-coding device includes a fixed codebook, an adaptive codebook, a short-term enhancement circuit, and a summing circuit. The short-term enhancement circuit connects an output of the fixed codebook to a summing circuit. The summing circuit adds an adaptive codebook contribution to a fixed codebook contribution. The short-term enhancement circuit can also be connected to a synthesis filter to emphasize the spectral formants in an encoder and a decoder.