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

The effect of dynamic range compression on the psychoacoustic quality and loudness of commercial music

It is common practice for music productions to be mastered with the aim of increasing the perceived loudness for the listener, allowing one record to stand out from another by delivering an immediate impact and intensity. Since the advent of the Compact Disc in 1980, music has increased in RMS level by up to 20dB. This results in many commercial releases being compressed to a dynamic range of 2–3 dB. Initial findings of this study have determined that amplitude compression adversely affects the audio signal with the introduction of audible artifacts such as sudden gain changes, modulation of the noise floor and signal distortion, all of which appear to be related to the onset of listener fatigue. In this paper, the history and changes in trends with respect to dynamic range are discussed and findings will be presented and evaluated. Initial experimentation, and both the roadmap and challenges for further and wider research are also described and discussed. The key aim of this research is to quantify the effects (both positive and negative) of dynamic range manipulation on the audio signal and subsequent listener experience. A future goal of this study is to ultimately define recommended standards for the dynamic range levels of mastered music in a similar manner to those associated with the film industry

A development and implementation of a tinnitus treatment method

Tinnitus is a physiological phenomenon where a person listens sounds which have not been generated by any external source. Today, many people suffer this condition. Although, in very few cases therapeutic methods completely eliminate tinnitus, it is possible to apply a variety of techniques to improve the quality of life of people with this condition. One of the most used methods to treat tinnitus consists of masking the tinnitus using an external sound. The main goal of this work is to present the development of a tinnitus treatment method, which optimizes the synthesized sounds in order to improve the life's quality of the user. Subjective tests and experimental results are used to analyze the performance of the method.Fil: Uriz, Alejandro José. Universidad Nacional de Mar del Plata. Facultad de Ingeniería. Departamento de Electrónica. Laboratorio de Comunicaciones; Argentina. Consejo Nacional de Investigaciones Científicas y Técnicas; ArgentinaFil: Agüero, Pablo Daniel. Universidad Nacional de Mar del Plata. Facultad de Ingeniería. Departamento de Electrónica. Laboratorio de Comunicaciones; ArgentinaFil: Tulli, Juan Carlos. Universidad Nacional de Mar del Plata. Facultad de Ingeniería. Departamento de Electrónica. Laboratorio de Comunicaciones; ArgentinaFil: Castiñeira Moreira, Jorge. Consejo Nacional de Investigaciones Científicas y Técnicas; Argentina. Universidad Nacional de Mar del Plata. Facultad de Ingeniería. Departamento de Electrónica. Laboratorio de Comunicaciones; ArgentinaFil: González, Esteban Lucio. Universidad Nacional de Mar del Plata. Facultad de Ingeniería. Departamento de Electrónica. Laboratorio de Comunicaciones; ArgentinaFil: Moscardi, Graciela. Universidad FASTA "Santo Tomas de Aquino"; ArgentinaFil: Sajama, Elber Emanuel. Universidad Nacional de Mar del Plata. Facultad de Ingeniería. Departamento de Electrónica. Laboratorio de Comunicaciones; Argentin

DSP algorithms for digital hearing instruments.

A new digital filter bank design and a new compression algorithm that can improve the performance of hearing instruments located completely in the ear canal (CIC) are developed in the thesis. In order to assess state-of-the-art hearing instruments employing advanced signal processing techniques the DynamEQ-II analog hearing instrument developed by the Gennum Corporation was studied extensively. A sophisticated SIMULINK model, involving the use of audio files, was developed to evaluate the performance characteristics of the strategies and algorithms used in the DynamEQ-II. The RangeEar algorithm employed in the DigiFocus hearing instrument from the Oticon Company was also studied using SIMULINK in a similar manner. Two recommended improvements for a new hearing instrument are presented. The first improvement involves the use of an eight-band digital filter bank based on an interpolated finite impulse response (IFIR) prototype filter that has been optimized using delay elements to give a maximally flat overall magnitude response. The resulting group delay is a constant and less than the value where self-hearing and lip reading problems occur. The second improvement uses a new compression algorithm based on a model of the human auditory system. The new algorithm replaces the existing constant homomorphic multiplication algorithms with an acoustic signal intensity weighted multiplication. The resulting nonlinear compression ratio expands low level signals and compresses high level signals in such a manner so as to improve noise immunity and increase the intelligibility of the sound. The MIT hearing loss simulator was employed to evaluate the effectiveness of the new proposed filter bank and compression algorithm by analysis of and listening to actual test audio files.Dept. of Electrical and Computer Engineering. Paper copy at Leddy Library: Theses & Major Papers - Basement, West Bldg. / Call Number: Thesis2001 .O53. Source: Masters Abstracts International, Volume: 41-04, page: 1157. Adviser: W. C. Miller. Thesis (M.A.Sc.)--University of Windsor (Canada), 2001

Noise reduction of dental drill noise

Dental drills produce a characteristic noise that is uncomfortable for patients and is also known to be harmful to dentists under prolonged exposure. It is therefore desirable to protect the patient and dentist whilst allowing two-way communication, which will require a headphone - type system. Re-establishing good communication between the dentist and patient will be achieved through a combination of three noise cancellation technologies, namely, Passive Noise Control (PNC), Adaptive Filtering (AF) and Active Noise Control (ANC). This paper describes how far a test-rig has been developed to achieve sufficient noise reduction that the uncomfortable noise can no longer be heard

Dental drill noise reduction using a combination of active noise control, passive noise control and adaptive filtering

Dental drills produce a characteristic high frequency, narrow band noise that is uncomfortable for patients and is also known to be harmful to dentists under prolonged exposure. It is therefore desirable to protect the patient and dentist whilst allowing two-way communication. A solution is to use a combination of the three main noise control methods, namely, Passive Noise Control (PNC), Adaptive Filtering (AF) and Active Noise Control (ANC). This paper discusses the application of the three methods to reduce dental drill noise while allowing two-way communication. Experimental setup for measuring the noise reduction by PNC is explained and results from different headphones and headphone types are presented. The implementation and results of an AF system using the Least Mean Square (LMS) algorithm are shown. ANC requires a modification of the LMS algorithm due to the introduction of the electro-acoustical cancellation path transfer function to compensate for the delays introduced by the control system. Therefore a cancellation path transfer function modeling method based on the filtered reference LMS (FXLMS) algorithm is presented along with preliminary results of the implementation

Active noise control for high frequencies

There are many applications that can benefit from Active Noise Control (ANC) such as in aircraft cabins and air conditioning ducts, i.e. in situations where technology interferes with human hearing in a harmful way or disrupts communication. Headsets with analogue ANC circuits have been used in the armed forces for attenuating frequencies below 1 kHz, which when combined with passive filtering offers protection across the whole frequency range of human hearing. A dental surgery is also a noisy environment; in which dental drill noise is commonly off-putting for many patients and is believed to harm the dentist’s hearing over a long period of time. However, dealing with dental drill noise is a different proposition from the applications mentioned above as the frequency range of the peak amplitudes goes from approximately 1.5 kHz to 12 kHz, whereas conventional ANC applications consider a maximum of 1.5 kHz. This paper will review the application of ANC at low frequencies and justify an approach for dealing with dental noise using digital technologies at higher frequencies. The limits of current ANC technologies will be highlighted and the means of improving performance for this dental application will be explored. In particular, technicalities of implementing filtering algorithms on a Digital Signal Processor will be addressed