摘要:
A multiple tone detector includes n tone detectors, each detecting one of n distinct tones, where n.gtoreq.2, and a background detector which generates a measure of accumulative background energy E.sub.avg in a frequency band or bands which do not include at least a subset of the n tones. The output of the background detector is applied to a smoothing filter, which generates the accumulative background energy measure E.sub.avg for a current frame as a weighted sum of the background detector output for the current frame and the background energy measure E.sub.avg from a previous frame. A parameter controlling response time of the smoothing filter is varied depending upon whether or not speech is determined to be present in the background portion of the input signal. A processor uses the energy measures from the n tone detectors and the background detector to compute n ratios, where a given ratio is the ratio of the energy measure of the ith tone to the accumulative background energy measure E.sub.avg. The processor determines if each of the n ratios are greater than a threshold, and if the maximum of the n ratios is less than a constant times the minimum of the n ratios, in order to generate a decision as to whether the n tones are present in the input signal.
摘要:
A processor formed from a signal processing unit operating according to instructions transmitted by a bus line, including a slave section provided with an address/data port for connection to a master signal processing circuit; a first buffer register in which data coming from the master processing circuit via the address/data port can be written and read in order to be processed by the processing unit, a second buffer register in which the data processed by the processing unit can be written, then read in order to be directed via the address/data port to the master processing circuit and a sequential control circuit so that access to these buffer registers is allocated in turn to the processing unit and to the master processing circuit. A master section is also provided intended to be connected to at least one slave circuit.
摘要:
Analog modems are enabled to better learn the slicing levels employed at the interface to a digital transmission network by reducing the effects of the various noise sources. Initially a training sequence is received to preliminarily adjust the analog modem's equalizer. Thereafter, a special training sequence, protected against intersymbol interference, is employed to collect samples of each slicing level, to ascertain the least mean squared value of each slicing level from the received samples and to obtain the channel's impulse response at each slicing level. To mitigate the effects of robbed bit signaling that may be employed in the digital transmission network, an array of slicers is provided to determine which bit position is being robbed and to base level learning on samples obtained from the non-robbed positions.
摘要:
A processor for carrying out a calculation mode from a selected plurality of different modes. The processor includes a clock pulse generator which generates clock pulses in an order for processing subsequent data. A mode circuit is included for detecting a mode declaration instruction. The mode declaration instruction is decoded to select a different clock pulse cycle for each different mode selected. Mode control signals and the selected clock pulse cycle are applied to a control code and borrow management circuit to enable the arithmetic and logic unit to carry out one or more operations of the mode control signals.
摘要:
A circuit compensating for the difference in transmission rate of digital samples generated in transmit and receive paths between a user and a transceiver processing in the frequency domain, such as a digital multi-tone (DMT) transceiver. Compensation of the DMT transmission rate in the receive path in accordance with exemplary embodiments of the present employs zero-padding of the frequency domain coefficients generated by the DMT transceiver prior to applying an inverse transform, such as the inverse fast Fourier transform (IFFT). Zero-padding the frequency domain coefficients allows for the compensation of the transmission rate in the receive path by generating digital samples from the frequency domain coefficients with an inverse transform having a rate matched to the frequency domain transform and rate employed in the transmit path.
摘要:
A data processor for executing a program of instructions stored in a program memory controlled by a program counter. To execute a loop control instruction, calling for repeated execution N times of a sequence of "i" instructions, the processor includes a loop circuit having an instruction counter which counts execution of the instructions in the loop sequence and produces an end-of-sequence signal upon each completion of the loop, a register which refreshes the program counter with the address of the first instruction in the loop in response to each end-of-sequence signal, and a loop counter which counts the number of completions of the loop and delivers a signal indicating the end of the loop portion of the entire program and enabling the program counter to continue on with the rest of the program. The delay in loop execution permits initializing of registers in the data processor so as to permit pipeline execution of the loop instruction.
摘要:
A so-called post equalization echo canceler is utilized in conjunction with transmitter and receiver data timing synchronization to enhance tracking of the echo path impulse response and convergence of the transversal filter in the post equalization echo canceler. This is realized by employing the equalization error in the receiver to adapt coefficients of the post equalization echo canceler transversal filter, in conjunction, with the transmitter and receiver data timing synchronization. The timing synchronization is realized by using sample rate conversion of the transmit sample rate to the receive sample rate and, in one example, variable phase interpolation of the converted timing signal. The receiver timing is recovered, and a phase error signal generated by the timing recovery unit is advantageously employed to adjust a variable phase interpolator in the receiver and a variable phase interpolator in a path supplying the transmitter signal to an input of the post equalization echo canceler. This insures that both the adaptive transversal filter of the post equalization echo canceler and a transversal filter in an equalizer in the receiver are operating on data having the same timing. In this example, the timing is that of the received data signal. In an embodiment of the invention, the post equalization echo canceler is utilized in conjunction with a so-called conventional, e.g., a primary, echo canceler. The conventional echo canceler is employed before the equalizer to cancel a major portion of any echo signal, while the post equalization echo canceler is employed after the equalizer to cancel residual echo signals caused primarily by drift in the hybrid network. To this end, the conventional echo canceler is “trained” during the initial half-duplex operation of the modem and, then, updating of its impulse response is inhibited, while the post equalization echo canceler is allowed to continue adapting.
摘要:
A digital speech communication system having improved synchronization. The present digital speech communication system reduces the unit of degradation to a single speech sample, rather than a multi-sample frame, while maintaining the bit rate efficiency of the DSVD system and other systems where speech is encoded into large blocks and is subject to variable delay and mismatched clocks. The basic unit that is dropped or artificially inserted by the receiver, if the buffer overflows or empties, respectively, is reduced to a single speech sample. The speech frames produced by the demultiplexer are written into a frame buffer, in units of frames, at a rate determined by the clock signal, S2, that is extracted from the received signal by a timing recovery function in the modem. In accordance with the present invention, the frames are read out of the buffer into the decoder using the same extracted clock signal, S2. In this manner, once the buffer is partially full, the frame buffer will not overflow or empty. The speech decoder converts the coded speech into blocks of speech samples. The blocks of speech samples are then written to a variable frame buffer, in accordance with the extracted clock signal, S2. The variable frame buffer is allowed to partially fill, before the speech samples are read out to the digital-to-analog converter according to a clock signal, S3, at the 8 kHz sample rate, for presentation to the listener. When the variable frame buffer overflows, only a single speech sample needs to be discarded rather than an entire frame of multiple samples. Likewise, when the variable frame buffer empties, only a single extraneous speech sample need be inserted. The number of samples in the variable frame buffer will preferably be kept within predefined tolerances by a write process and a read process.
摘要:
A data processing apparatus in which a memory (10) is accessed at addresses stored in an address regiser (20). An incrementation circuit (38) successively increments or decrements the address stored in the principal register, under the control of an address cycling circuit (22). A pair of auxiliary registers (30, 35) respectively store the minimum and maximum address values to be reached in the principal register, and a comparison circuit (37) determines when the address therein matches the minimum or maximum value. The address cycling circuit, together with the comparison circuit, loads the principal register with the minimum address value when the address therein reaches the maximum value, the address therein thereafter being decremented, and loads it with the maximum address value when the address therein reaches the minimum value, the address therein thereafter being incremented. Such operation is particularly useful for preforming the functions of a fixed or adaptive transversal filter for data transmission, the data stored in the memory being the filter coefficients.
摘要:
A unique real time tuning (RTT) process is employed for obtaining the desired optimum device parameter adjustments. The RTT parameter adjustment process is utilized with IP phone or other device chipsets as desired. In one embodiment, RTT provides a graphical user interface to a digital signal processor (DSP), or the like, on the device chipset allowing for observation, evaluation and control of the device parameters in real time. The real time exchange of the device parameter information between the device and an external workstation, e.g., a personal computer or the like, is provided by a User Datagram Protocol (UDP) that runs on a controller on the device, e.g., an ARM processor or the like. In this example, the unique combination of the RTT, UDP and DSP cooperate advantageously to implement, in accordance with the principles of the invention, the desired observability, and control to designers to tune the device, e.g., IP Phone, in real time to specified hardware, plastics, audio requirements required by existing standards or the like.