Fixed FIR Filter Research Articles

New tuning model for rectangular windowed FIR filter using fractional Fourier transform

In this paper, new mathematical model is developed for computing the rectangular windowed low-pass FIR filter transfer function where the rotation angle of fractional Fourier transform is used as a free parameter. By changing the rotation angle in the range from \(0\) to \(\pi /2\), the transition bandwidth and the stop-band attenuation can be tuned. The closed-form expression derived establishes a direct relationship between magnitude response of the filter and fractional angle of the transform. It is shown that sharp transition bandwidth followed with an increase in the stop-band attenuation can be achieved with the rectangular window for small values of the rotation angle. Digital filters with variable frequency responses can be obtained directly by raising the filter length. This involves recomputation of impulse response coefficients and redesigning of a new filter from the existing one. More number of data registers, multipliers and filter coefficients will be needed, which adds to computational burden. Using proposed fractional Fourier transformation technique, the response characteristics of FIR filter can be tuned during operation by changing only one parameter, while keeping coefficients of the fixed FIR filter unchanged.

All-Digital Background Calibration Technique for Time-Interleaved ADC Using Pseudo Aliasing Signal

A new digital background calibration technique for gain mismatches and sample-time mismatches in a Time-Interleaved Analog-to-Digital Converter (TI-ADC) is presented to reduce the circuit area. In the proposed technique, the gain mismatches and the sample-time mismatches are calibrated by using pseudo aliasing signals instead of using a bank of adaptive FIR filters which is conventionally utilized. The pseudo aliasing signals are generated and subtracted from an ADC output. A pseudo aliasing generator consists of the Hadamard transform and a fixed FIR filter. In case of a two-channel 10-bit TI-ADC, the proposed technique reduces the requirement for a word length of the FIR filter by about 50% without a look-up table (LUT) compared with the conventional technique. In addition, the proposed technique requires only one FIR filter compared with the bank of adaptive filters which requires (M-1) FIR filters in an M-channel TI-ADC.

Efficient Implementation of Sample Rate Converter

Within wireless base station system design, manufacturers continue to seek ways to add value and performance while increasing differentiation. Transmit/receive functionality has become an area of focus as designers attempt to address the need to move data from very high frequency sample rates to chip processing rates. Digital Up Converter (DUC) and Digital Down Converter (DDC) are used as sample rate converters. These are the important block in every digital communication system; hence there is a need for effective implementation of sample rate converter so that cost can be reduced. With the recent advances in FPGA technology, the more complex devices providing high-speed as required in DSP applications are available. The filter implementation in FPGA, utilizing the dedicated hardware resources can effectively achieve application-specific integrated circuit (ASIC)-like performance while reducing development time cost and risks. So in this paper the technique for an efficient design of DDC for reducing sample rate is being suggested which meets the specifications of WiMAX system. Its effective implementation also ensures the pathway for the efficient applications in VLSI designs. Different design configurations for the sample rate converter are explored. The sample rate converter can be designed using half band filters, fixed FIR filters, poly-phase filters, CIC filters or even farrow filters.

Adaptive blind calibration of timing offset and gain mismatch for two-channel time-interleaved ADCs

In this paper, we describe a blind calibration method for gain and timing mismatches in a two-channel time-interleaved low-pass analog-to-digital converters (ADC). The method requires that the input signal should be slightly oversampled. This ensures that there exists a frequency band around the zero frequency where the Fourier transforms of the ADC subchannels are alias free. Low-pass filtering the ADC subchannels to this alias-free band reduces the blind calibration problem to a conventional gain and time delay estimation problem for an unknown signal in noise. An adaptive filtering structure with three fixed FIR filters and two adaptive gain and delay parameters is employed to achieve the calibration. A convergence analysis is presented for the blind calibration technique. Numerical simulations for a bandlimited white noise input and for inputs containing several sinusoidal components demonstrate the effectiveness of the proposed method

Active field control (AFC) -electro-acoustic enhancement system using acoustical feedback control

AFC is an electro-acoustic enhancement system using FIR filters to optimize auditory impressions, such as liveness, loudness, and spaciousness. This system has been under development at Yamaha Corporation for more than 15 years and has been installed in approximately 50 venues in Japan to date. AFC utilizes feedback control techniques for recreation of reverberation from the physical reverberation of the room. In order to prevent coloration problems caused by a closed loop condition, two types of time-varying control techniques are implemented in the AFC system to ensure smooth loop gain and a sufficient margin in frequency characteristics to prevent instability. Those are: (a) EMR (electric microphone rotator) -smoothing frequency responses between microphones and speakers by changing the combinations of inputs and outputs periodically; (b) fluctuating-FIR -smoothing frequency responses of FIR filters and preventing coloration problems caused by fixed FIR filters, by moving each FIR tap periodically on time axis with a different phase and time period. In this paper, these techniques are summarized. A block diagram of AFC using new equipment named AFC1, which has been developed at Yamaha Corporation and released recently in the US, is also presented.

Noise cancellation system based on active noise control. 1st Report, Modeling of FIR filter system for short-length ductlike, structures.

Active noise control techniques have been studied in recent years as an available method of machinery noise reduction. Especially, there are many applications of them to ductlike structures where sound waved radiated from noise sources can be assumed to be traveling as plane waves. In most of these cases, the lengths by ducts were assumed to be infinitely long, and acoustic properties such as reflections and resonances inside the ducts have seldom been considered. In actual machines, however, the duct lengths ate often short because of the limit of machine dimensions; hence a basic study on active noise control for such short-length ducts is needed. This paper describes the modeling method of an active noise control system using a fixed FIR filter for short-length ducts. The results of simulation bclculations were compared with experimental data and good agreement was obtained.

Some complexity issues in digital signal processing

Over the past decade a large class of problems, called NP-complete [5], have been shown to be equivalent in the sense that if a fast algorithm can be found for one, fast algorithms can be found for all. At the same time, despite much effort, no fast algorithms have been found for any, and these problems are widely regarded as intractable. This class includes such notoriously difficult problems as the traveling salesman problem, graph coloring, and satisfiability of Boolean expressions. Using FIR filter implementation as an illustration, we describe some problems in digital signal processing that are NP-complete. These include: 1) minimize the number of additions needed to implement a fixed FIR filter; 2) minimize the number of registers needed to implement a fixed FIR filter; and 3) minimize the time to perform the additions of such an FIR filter using P adders. Large-seale instances of such problems may become important with the use of programmable chips to implement signal processing. Our main purpose in this paper is to illustrate the usefulness of asymptotic complexity theory in the field of digital signal processing. The theory discriminates between tractable and intractable problems, Sometimes identifies fast algorithms for the former, and justifies heuristics for the latter.