Multi-channel active noise cancellation using the DSP56001 (digital signal processor)

The authors report on the performance of a portable active noise cancellation (ANC) system based around a PC hosted 20-MHz Motorola DSP56001 processor with a four-channel analog input/output (I/O) board connected to the real world via standard consumer audio components. The system will perform active noise cancellation over the frequency range of 65-500 Hz. Quantitative results are presented for the cancellation of single tone noise and of narrowband noise, and a measure of the ANC power spectrum is calculated for various parameters of the filtered-X LMS algorithm in different acoustic environments. Qualitative results based on human hearing perception of the attenuation of various narrowband and real world noise sources are also discussed

Estimation-based synthesis of H∞-optimal adaptive FIR filtersfor filtered-LMS problems

This paper presents a systematic synthesis procedure for H∞-optimal adaptive FIR filters in the context of an active noise cancellation (ANC) problem. An estimation interpretation of the adaptive control problem is introduced first. Based on this interpretation, an H∞ estimation problem is formulated, and its finite horizon prediction (filtering) solution is discussed. The solution minimizes the maximum energy gain from the disturbances to the predicted (filtered) estimation error and serves as the adaptation criterion for the weight vector in the adaptive FIR filter. We refer to this adaptation scheme as estimation-based adaptive filtering (EBAF). We show that the steady-state gain vector in the EBAF algorithm approaches that of the classical (normalized) filtered-X LMS algorithm. The error terms, however, are shown to be different. Thus, these classical algorithms can be considered to be approximations of our algorithm. We examine the performance of the proposed EBAF algorithm (both experimentally and in simulation) in an active noise cancellation problem of a one-dimensional (1-D) acoustic duct for both narrowband and broadband cases. Comparisons to the results from a conventional filtered-LMS (FxLMS) algorithm show faster convergence without compromising steady-state performance and/or robustness of the algorithm to feedback contamination of the reference signal

Interpolated-DFT-Based Fast and Accurate Amplitude and Phase Estimation for the Control of Power

The quality of energy produced in renewable energy systems has to be at the high level specified by respective standards and directives. The estimation accuracy of grid signal parameters is one of the most important factors affecting this quality. This paper presents a method for a very fast and accurate amplitude and phase grid signal estimation using the Fast Fourier Transform procedure and maximum decay sidelobes windows. The most important features of the method are the elimination of the impact associated with the conjugate's component on the results and the straightforward implementation. Moreover, the measurement time is very short - even far less than one period of the grid signal. The influence of harmonics on the results is reduced by using a bandpass prefilter. Even using a 40 dB FIR prefilter for the grid signal with THD = 38%, SNR = 53 dB and a 20-30% slow decay exponential drift the maximum error of the amplitude estimation is approximately 1% and approximately 0.085 rad of the phase estimation in a real-time DSP system for 512 samples. The errors are smaller by several orders of magnitude for more accurate prefilters.Comment: in Metrology and Measurement Systems, 201

Real-time adaptive filtering of dental drill noise using a digital signal processor

The application of noise reduction methods requires the integration of acoustics engineering and digital signal processing, which is well served by a mechatronic approach as described in this paper. The Normalised Least Mean Square (NLMS) algorithm is implemented on the Texas Instruments TMS320C6713 DSK Digital Signal Processor (DSP) as an adaptive digital filter for dental drill noise. Blocks within the Matlab/Simulink Signal Processing Blockset and the Embedded Target for TI C6000 DSP family are used. A working model of the algorithm is then transferred to the Code Composer Studio (CCS), where the desired code can be linked and transferred to the target DSP. The experimental rig comprises a noise reference microphone, a microphone for the desired signal, the DSK and loudspeakers. Different load situations of the dental drill are considered as the noise characteristics change when the drill load changes. The result is that annoying drill noise peaks, which occur in a frequency range from 1.5 kHz to 10 kHz, are filtered out adaptively by the DSP. Additionally a schematic design for its implementation in a dentist’s surgery will also be presented

Development of Real-Time Adaptive Noise Canceller and Echo Canceller

In this paper, the adaptive cancellation structure is firstdeveloped based on the LMS algorithm and FIR adaptivefiltering. Then the novel practical noise and echo cancellationsystems are built using the proposed adaptive technique andimplemented using TX320TMS67C13 DSKs, which are TexasInstruments’ Digital Signal Processing (TI DSP) boards.Although adaptive filtering is an exciting topic in which manyreal-life applications can be explored [1]-[6], [9], building such areal-time system is often challenging due to the use of theoreticalmath, advanced DSP knowledge and practical industrial hands-onexperience [1],[4]-[6],[9]. Therefore, this paper indicates that it ispossible to apply traditional mathematics in adaptive filteringtheory to real-time practical DSP systems. With the MATLABsoftware tool, we can simulate and verify various adaptivefiltering designs first. Then, development and implementation ofdifferent noise or echo cancellation systems with adaptive filteringtechniques can be successfully performed using the floating-pointdigital signal processor, TX320TMS67C13 DSK. Furthermore, itcan be shown that TX320TMS67C13 DSKs with their stereochannels offer more effective and flexible tools for various noisecancellation applications

Sound Localization using VHDL

Sound localization based on listener's potential to address the location or source of a located sound in orientation and distance. Rather than detecting what has been spoken, the work concerns about the detection of where the sound comes from. Speech enhancement aims to improve speech quality by using various algorithms. The purpose of enhancement is to increase user-friendliness and overall excellence of degraded speech signal using audio signal filtering techniques. The better speech quality improves by sign-error LMS algorithm. FPGAs have become a competitive alternative for high performance DSP applications, previously dominated by general purpose DSP and ASIC devices. The FPGA is useful for many multimedia applications and functional systems. The FPGA can be programmed to perform any number of parallel paths

Digital Signal Processing Education: Technology and Tradition

In this paper we discuss a DSP course presented to both University students and to participants on industrial short courses. The "traditional" DSP course will typically run over one to two semesters and usually cover the fundamental mathematics of z-, Laplace and Fourier transforms, followed by the algorithm and application detail. In the course we will discuss, the use of advanced DSP software and integrated support software allow the presentation time to be greatly shortened and more focussed algorithm and application learning to be introduced. By combining the traditional lecture with the use of advanced DSP software, all harnessed by the web, we report on the objectives, syllabus, and mode of teaching

Active Noise Control with Sampled-Data Filtered-x Adaptive Algorithm

Analysis and design of filtered-x adaptive algorithms are conventionally done by assuming that the transfer function in the secondary path is a discrete-time system. However, in real systems such as active noise control, the secondary path is a continuous-time system. Therefore, such a system should be analyzed and designed as a hybrid system including discrete- and continuous- time systems and AD/DA devices. In this article, we propose a hybrid design taking account of continuous-time behavior of the secondary path via lifting (continuous-time polyphase decomposition) technique in sampled-data control theory