摘要:
An echo canceller in a system, having a long impulse response such as in an acoustic system, which employs a fast Fourier transform to transform input data represented in the time domain into signals represented in the frequency domain to reduce calculations. To solve the problem of a long delay, the impulse response length is divided into a plurality of blocks. Each block then has a decreased number of samples within each block. Thus, a fast Fourier transform and finite impulse response type digital filtering are effected, so that the processing delay is decreased while the amount of calculations is kept small.
摘要:
Disclosed is a voice coding/decoding system having a transmitting part for transmitting a coded signal of an input voice signal at a bit rate lower than a predetermined transmission bit rate and a receiving part for receiving and decoding the coded signal transmitted from the transmission part. To enable the coding and transmitting of an input voice signal in an optimum state without passing through a buffer memory and without having a negative influence on the coder, the transmitting part provides coders for coding the input voice signal and groups of entropy coders. The inputs of the entropy coders in each group are connected to the output of one of the plurality of coders. The transmitting part further provides an evaluation part for evaluating the characteristics of the outputs of the coders and the entropy coders. The evaluation part extracts those entropy coders having output bit rates lower than the transmission bit rate and extracts, from the coders connected to the extracted entropy coders, a coder having the best output characteristic. Then, the evaluation part outputs a selecting signal indicating the combination of the selected coder and an entropy coder from the extracted entropy coders. The transmitting part further provides a selecting part for selecting, in response to the selecting signal, the codeword passed through the combination of the coder and the entropy coder to be transmitted.
摘要:
A timing-phase recovery circuit, which is suitably mounted on a digital LSI, includes a timing phase recovery circuit. The timing-phase recovery circuit includes a first circuit for extracting a digital timing signal from a received input analogue signal, a second circuit for detecting a virtual zero crossing included in the digital timing signal in synchronism with a sampling signal and a third circuit for carrying out a phase shift with respect to the sampling signal in order to tune the timing signal to the frequency of the virtual zero crossings in a very short period of time.
摘要:
An echo canceller provided with an expected echo generating unit which generates an expected echo of a transmission signal and a frequency offset correction unit which detects a phase error between an echo in a received signal and an expected echo from the expected echo generating unit and corrects the frequency offset for the expected echo by an offset frequency estimated based on the phase error, wherein use is made of the expected echo after offset correction by the frequency offset correction unit so as to suppress the echo in the received signal, the frequency offset correction unit using the phase error obtained after normalizing the phase error by the magnitude of the echo in the received signal. By this, it is possible to perform frequency offset correction adaptive to the far-end echo characteristics. Further, it is desirable to perform the frequency offset correction after it is confirmed that the rate of change of the output from a subtractor for cancellation of the near-end echo has become almost zero.
摘要:
An automatic equalization device used in a data communication system includes a unit for extracting a single pulse from a training signal sent from a transmitter and initializing a tap coefficient using the extracted single pulse. The automatic equalization device also includes a first equalization circuit and a second equalization circuit. In the first equalization circuit, an auto-correlation series of the signal corresponding to the single pulse is calculated to provide a symmetric single pulse. In the second equalization circuit, an inverse matrix is calculated from the symmetric single pulse using the auto-correlational series of single pulses. By these calculations, the speed of the initial setting of the tap coefficient is increased.
摘要:
The present invention relates to a digital phase locked loop circuit, particularly to a circuit which realizes accurately digital phase locked loop pull-in operation at a high speed with a simplified circuit structure.In the present invention, in order to obtain a phase difference between a single frequency signal and the digital phase locked loop clock signal which is obtained by dividing a specified frequency signal with a dividing counter, the phase difference is obtained in accordance with the signs, absolute values and amplitude ratio of two adjacent sample values. The sample values of said single frequency signal are taken at two points based on said digital phase locked loop clock signal corresponding to a phase difference of .pi./2 radians of said single frequency signal. A fast pull-in of the digital phase locked loop is realized by setting a value corresponding to the obtained phase difference into a dividing counter.
摘要:
A digital automatic gain control circuit provided with a first AGC loop and a second AGC loop; activated alternately. The first AGC loop is activated when an input signal is initially supplied. The loop produces a certain digital value calculated as an inverse number of the digital level of the input through predetermined digital arithmetic operations utilizing an approximation polynominal. The calculated digital value is then preset in certain portions of the second AGC loop. The second AGC loop starts operating by using said present value, then uses the digital level of the input itself, instead of the preset value, and produces an automatic gain controlled digital output. Utilizing said preset value as an initial input to the second AGC loop allows for rapid stabilization of the automatic gain controlled digital output.
摘要:
A variable rate coder for transforming an input signal into a low bit rate digital signal, comprising: a plurality of coding units each comprising an output quantizer having a different number of output bits, and each independently transforming the input signal into a respective compressed digital signal. A plurality of bit rate determined circuits are provided to determine an output of one of the plurality of coding units as a candidate output of the coder by a respective individual determining method, in which method, the quality of the signal which is regenerated from the output of each coding unit is compared with an individual predetermined quality evaluating standard, and then the candidate output of the coding unit having the smallest number of output bits, among the candidate outputs of coding units satisfying each standard, is determined as the actual output of the coder. One candidate output, among the plurality of the determined candidate outputs and, which is further determined by a most suitable determining method relative to the level of the input signal, thus is selected as the actual output of the coder. Further, when the number of the output bits in each of the coding units can be variably set, the most suitable number of output bits for the actual output of the coder is calculated based on prediction gains of the coding units, a required signal-to-noise ratio of the coder, and an ADC signal-to-noise ratio at the analog to digital converter, and the number of the output bits in each of the coding units is set based on the above calculated value, before the above operation by the bit rate determining circuits.
摘要:
Speech presence versus silence is decided by a discriminator which can use a certain combination of parameter values: signal power, prediction error power, prediction error power deviation, and zero crossings.