摘要:
A phase control signal generation device generating a phase control signal for each of frequency bands for an audio signal converted into a frequency domain, the phase control signal generation device comprising: a setting change means that is able to change setting of a propagation delay time for each of predetermined frequency bands; a difference obtaining means that obtains a difference between propagation delay times before and after setting change; an updating means that updates a phase control amount of the frequency band for which the propagation delay time is changed, based on the obtained difference; and a phase control signal generating means that generates a phase control signal of each frequency band by performing a smoothing process for the phase control amount in a frequency domain using the updated phase control amount.
摘要:
Embodiments of the present invention provide a signal processing method and apparatus. The method includes: performing M-way filtering on an input signal to obtain M filtered signals, performing extraction on M filtered signals separately to obtain M extracted signals, performing fast Fourier transform FFT on the M extracted signals separately to obtain M frequency-domain signals, and finally determining output signals according to the M frequency-domain signals. According to the embodiments of the present invention, signal filtering and extraction are performed and then FFT is performed. In this way, computation complexity can be reduced.
摘要:
Method, user device and computer program product for suppressing echo. An audio signal is output from a speaker. A microphone receives an audio signal, wherein the received audio signal includes an echo resulting from said outputted audio signal. A Finite Impulse Response filter estimate ĥ(t) is dynamically adapted in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(t) of the echo in the received audio signal. At least one power response is determined from the filter estimate ĥ(t) and used to estimate the echo power of the echo in the received audio signal. The estimated echo power is used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal.
摘要:
A digital signal processing apparatus includes a frame generator configured to generate a plurality of frames from a row of sample data of a time-domain, a part of each frame overlapping with adjoining frames, a Fourier transform unit configured to transform at least one of the generated frames into a frequency domain by Fourier transformation, an addition unit configured to add predetermined frequency characteristic to the transformed frame, and an inverse Fourier transform unit configured to transform the added frame into the time-domain by inverse Fourier transformation and to delete the overlap of the frame of the time-domain transformed.
摘要:
A method for long impulse response digital filtering of an input data stream, by use of a digital filtering system. Where the input data stream is divided into zero-input signals and zero-state signals. One of the zero-input signals and a corresponding impulse response of the digital filtering system is converted to the frequency domain to determine a respective zero-input response of the digital filtering system. One of the zero-state signals is convolved with a corresponding impulse response of the digital filtering system to determine a respective zero-state response of the digital filtering system, wherein at least part of the zero-input signal precedes the zero-state signal. The zero-state response of the digital filtering system is added to the zero-input response of the digital filtering system to determine the response of the digital filtering system. Apparatus for effecting this method is also disclosed.
摘要:
The present invention relates generally to the problem of filtering, decimation or interpolation and frequency conversion in the digital domain, and more particularly to the use of the stand-alone or improved modified fast convolution algorithm in wideband multichannel receiver, channelization, and transmitter, de-channelization structures of a radio communication system. The invention consists of essentially 3 steps: making sure that we use different overlaps on consecutive blocks that, on average, give the same overlap on both the input and output ends; aligning the signal in consecutive blocks of time; and compensating for phase shifts due to frequency shifting. The essence of the invention is that it decouples the input and output transform lengths in the fast convolution algorithm from each other and from the overlap, making it possible to use any transform length on the input together with any transform length on the output and at the same time use any overlap. This provides an enormous amount of freedom compared with the limitations of state of the art.
摘要:
In a wireless telecommunications system, data processing delays associated with digital channelization and de-channelization may be reduced through the use of a technique that involves processing data blocks in conjunction with the transformation of the data blocks by a large, Fast Fourier Transform (FFT) algorithm, and makes use of multiple transmission and reception branches. In accordance with this technique, the processing delays associated with the FFT algorithm are minimized, but not to the detriment of other important channelizer/de-channelizer design characteristics, such as power consumption, die area and computational complexity.
摘要:
Apparatus for and methods of operation of digital filters and certain other electronic digital signal processing devices are provided to improve the accuracy and efficiency of filtering. Particularly, the apparatus and method includes a digital filter with a long impulse response and lower latency, built by operating a numbr of small component filters (F1, F2, F3) in parallel and combining their outputs by addition, with each component filter operating with a different delay such that the net operation of the ensemble of said component filters is the same as a single filter with a longer impulse response, and the latency of the ensemble is equal to the shortest latency of the component filters. At least one group of the component filters is implemented using a Discrete Fourier Transform technique. A Fourier transform processor adapted to efficiently transform strings of real data is also described.
摘要:
A parallel transfer rate converter (4) inputs first parallel data with number of samples being S1 pieces in synchronism with a first clock, and outputs second parallel data with number of samples being S2=S1×(m/p) pieces (p is an integer equal to or larger than 1) in synchronism with a second clock having a frequency which is p/m times of a frequency of the first clock. A convolution operation device (5) inputs the second parallel data in synchronism with the second clock, generates third parallel data with number of samples being S3=S2×(n/m) pieces (S3 is an integer equal to or larger than 1) by executing a convolution operation with a coefficient indicating a transmission characteristic to the second parallel data, and outputs the third parallel data in synchronism with the second clock.
摘要:
The present invention relates to a method for processing a digital input signal by a Finite Impulse Response, FIR, filtering means, comprising partitioning the digital input signal at least partly in the time domain to obtain at least two partitions of the digital input signal; partitioning the FIR filtering means in the time domain to obtain at least two partitions of the FIR filtering means; Fourier transforming each of the at least two partitions of the digital input signal to obtain Fourier transformed signal partitions; Fourier transforming each of the at least two partitions of the FIR filtering means to obtain Fourier transformed filter partitions; performing a convolution of the Fourier transformed signal partitions and the corresponding Fourier transformed filter partitions to obtain spectral partitions; combining the spectral partitions to obtain a total spectrum; and inverse Fourier transforming the total spectrum to obtain a digital output signal.