Analog Audio Signal Research Articles

Multi-Channel Long-Distance Audio Transmission System Using Power-over-Fiber Technology

To establish stable communication networks in harsh environments where power supply is difficult, such as coal mines and underwater, we propose an effective scheme for co-transmission of analog audio signals and energy. By leveraging the advantages of optical fibers, such as corrosion resistance and strong resistance to electromagnetic interference, the scheme uses a 1550 nm laser beam as the carrier for analog audio signal propagation, which is then converted to electrical energy through a custom InGaAs/InP photovoltaic power converter (PPC) for energy supply and information transfer without an external power supply after a 25 km fiber transmission. In the experiment, with 160 mW of optical power injection, the scheme not only provides 4 mW of electrical power, but also transmits an analog signal with an acoustic overload point (AOP) of 105 dBSPL and a signal-to-noise ratio (SNR) of 50 dB. In addition, the system employs wavelength division multiplexing (WDM) technology to transform from single-channel to multi-channel communication on a single independent fiber, enabling the arraying of receiving terminals. The passive arrayed terminals make the multi-channel long-distance audio transmission system using power-over-fiber (PoF) technology a superior choice for establishing a stable communication network in harsh environments.

Ultra-Low-Power and Compact-Area Analog Audio Feature Extraction Based on Time-Mode Analog Filterbank Interpolation and Time-Mode Analog Rectification

To address the power and area bottleneck imposed by the frontend feature extractor relative to the backend neural network in on-device keyword spotting (KWS), we propose two time-mode analog signal processing (ASP) circuit techniques showcased in an analog audio feature extractor chip that advances the state of the art in power- and area-efficiency. Time-mode analog filterbank interpolation uses digital XOR gates to double the number of outputs of an analog bandpass filterbank. Time-mode analog rectification uses a single digital XOR gate as an analog full-wave rectifier. The 65 nm low power (LP) CMOS chip uses only 80 nW and 0.53 mm2 to extract from an input analog audio signal, an output digital auditory feature vector with 31 elements. This represents <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$18\times $ </tex-math></inline-formula> and <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> <tex-math notation="LaTeX">$3.3\times $ </tex-math></inline-formula> improvements in power/feature and area/feature, as compared, respectively, to the most area- and power-efficient published analog audio feature extractor chips. All the while, competitive classification accuracy is maintained at >90% across ten keywords, as evaluated by feeding the chip’s digital output directly into a small-footprint software backend classifier.

Audio Signal Deterioration Caused by Propagation Noise between Audio Equipment (2nd report)

Sound quality of music playback may be deteriorated due to various complex factors such as power supply state of the audio equipment. Therefore, in our previous study, we partly confirmed the mechanism that the power supply states affected alternative-current (AC)-potential difference and propagation noise between the equipment, which influenced audio-signal intermodulation distortion (IMD) on one of personal-computer (PC)-audio system. However, the research was insufficient in the number of measured audio equipment and measuring setup. In this study, we increased the model of digital-to-analog converter (DAC) and audio amplifier used for measurement and improved the measurement method to reduce the influence of measuring equipment. We measured ground AC potential, propagation noise, and the analog audio signal output of the amplifier. Their measurements were done in changing combination of audio equipment and each power supply states to analyze the relationship among these three domains. As a result, it is found that each domain depended on the model, combination, and power supply states of each audio equipment. There is a strong positive correlation between the ground AC potential and the propagation noise. There is also a positive correlation between the propagation noise and the audio-signal IMD in some combination of audio equipment.

Audio Signal Processing Based on FPGA

This paper presents a system of audio signal processing based on FPGA,the system uses audio codec chip LM4550 to A/D transform and D/A transform the input analog audio signal and output digital audio signal.Using FPGA as the high speed signal processor to realize volume adjustment and audio effect control,so it can output different style music.Meantime, the system designs a FFT computing module and control system of VGA display interface,to compute the digital audio signal which is A/D transformed,and real-time display the frequency spectrum of audio signal on VGA.

Digital removal of pulse‐width‐modulation‐induced distortion in class‐D audio amplifiers

In an all-digital, class-D audio amplifier, pulse-width-modulation (PWM) of a digital signal source is usually followed by a low-order analog low-pass filter to construct the analog audio waveform. This study shows how to remove the non-linear distortion usually associated with PWM, by prefiltering the digital signal prior to the PWM mapping in such a manner that the overall result is distortion-free. The prefiltering is done using computationally effective infinite-impulse-response filters combined with short-kernel anticausal finite-impulse-response filters, and relies on the interpretation of PWM as a Volterra filter. A case study is presented where a second-order Butterworth analog low-pass filter is used for reconstruction of the analog audio signal. A complete amplifier system is modelled, including upsampling, Volterra prefiltering and noise feedback coding. Computer simulations on CD music signals were performed. Using a third-order prefilter, a signal-to-noise ratio of 97–102 dB was obtained for the music signals tested. All necessary filter data needed for realisation of the prefilter are given in the Appendix.

VLC 기반 멀티 홉 멀티미디어 데이터 전송 네트워크 시스템

In this paper, we propose VLC(visible light communication) based multi-hop multimedia data transmission system. The main contributions and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit multimedia data(audio data, video data) with long distance. Second, the developed system has the following features: In transmitter, audio data and video data are transmitted via multi-hops using two channels. The relay in audio channel receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The relay in video channel receives video signal by using photo diode and then amplify the video signal using OP-AMP and then transmits the signal to receiver. The receiver amplifies the received signal from photo diode and then sends it to the monitor. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality multimedia data transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted multimedia(audio and video) quality according to the used LED colors.

Design and Implementation of Audio Process System Based on DSP

The real-time processing for the input analog audio signal, audio processing program is proposed based on DSP. The system use FFT algorithm as the core, first, the input analog audio signal is sampled and A/D conversion using TLV320AIC23, and then use high speed digital signal processor to make real-time processing for the signal. Theoretical and experimental results show that the system can meet the design requirements, it has the advantage of high real-time and simple structure. The system has a good application and reference value for the development and design of data collecting and remote monitoring.

Simultaneous transmission of audio and video signals using visible light communications

Abstract A visible light communication system for parallel transmission of high-quality audio and video signals using illumination white and red light-emitting diodes (LEDs) is presented in this paper. For video transmission, we adopts the pulse width modulation (PWM) scheme at the carrier frequency of 15 MHz, whereas for the analog audio signal, the Sony/Philips Digital standard is used. We show that successful simultaneous high-quality audio and video signal transmissions over a link span of 50 cm are possible without the need for optimized optics. For audio transmission, only the link span is doubled to 100 cm.

LED 통신 기반 멀티 홉 오디오 데이터 전송네트워크시스템

In this paper, we propose a LED communication based multi-hop audio data transmission network system. The main contribution and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit audio data signal with long distance via multi-hops. Second, the developed system has the following features: In transmitter, audio data is transmitted after encoding with S/PDIF format via a general LED. The relay receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality audio transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted audio quality according to the used LED colors.

오디오 D/A 컨버터를 위한 인터폴레이티드 디지털 델타-시그마 변조기

디지털 입력 D급 증폭기는 보청기에서 사용되고 있으며 D급 증폭기는 디지털 회로와 아날로그 회로로 구성되어진다. 아날로그 회로는 가청 주파수 대역에서 잡음을 억제하고 디지털 입력을 아날로그 신호로 변환한다. 본 논문에서 제안한 인터폴레이티드 디지털 델타-시그마 변조기는 디지털 신호 처리기의 출력 신호를 D/A 변조기 입력에 적합하도록 데이터를 변조시킨다. 디지털 필터는 16-bit, 25-kbps 펄스 코드 변조 신호를 16-bit, 50-kbps 신호로 보간 작업을 한다. 이 보간 필터 출력은 3차 디지털 델타-시그마 변조기를 통하여 노이즈 쉐이핑(noise shaping) 처리된다. 최종적으로, 1.5-bit, 3.2-Mbps 신호가 D/A변조기 입력으로 인가된다.

CONFERENCE AUDIO SYSTEM

Even in a conference audio system having an automatic mute release device, it is possible to reduce a delay time from an utterance to output from a speaker. The system includes: an A/D converter (33) for converting an audio signal from a plurality of microphones into a digital signal; audio level detection means to detect whether the converted digital signal level indicates an utterance or no sound; audio data storage means (32) for temporarily storing the digital signal in which the audio level detection means has detected an utterance; control means (31) for controlling storage of the audio data into the audio data storage means (32) and read out of the audio data; and a D/A converter (34) for converting the read-out audio data into an analog audio signal. When the audio level detection means has detected no sound in the series of audio data, the control means (31) hastens the audio data read-out timing in correspondence to the time of the no sound portion.

Noise Elimination Circuit For Audio Equipment

A noise elimination circuit of a audio output device includes a sound card for receiving a digital audio signal and transforming it to an analog audio signal; an audio reproduction apparatus such as a audio output device connected to the sound card for receiving the analog audio signal; a power supply connected to the sound card for providing a working voltage thereto; a capacitor connected between the power supply and ground; and a discharging circuit also connected between the power supply and ground, for discharging deposited current in the capacitor when the audio output device is cut off. The discharging circuit is controlled according to the state of the power supply.

Dynamically Programmable Analog Arrays in Acoustic Frequency Range Signal Processing

Dynamically Programmable Analog Arrays in Acoustic Frequency Range Signal ProcessingField programmable analog arrays (FPAA), thanks to their flexibility and reconfigurability, give the designers quite new possibilities in analog circuit design. The number of both academic projects on FPAA and applications of commercially available programmable devices is still growing. This paper explores the properties and parameters of two most popular FPAA circuits: the AnadigmVortex AN221E04 and AnadigmApex AN231E04 from the Anadigm company. The research conducted by the authors led to the discovery of some undocumented features of these devices. Several applications for audio processing were built and tested. The results show that these circuits can be used in medium-demanding audio applications. Thanks to dynamic reconfigurability, they also allow to build an universal analog audio signal processor. These circuits can also act as a versatile platform for rapid prototyping and educational purposes.

Self-descriptive microphone array

A self-descriptive microphone array includes a microphone array memory, such as, for example a ROM, EEPROM, or other conventional memory, which contains a microphone array device description. This device description includes parametric information which defines operational characteristics and configuration of the microphone array. In further embodiments, the microphone array uses any of a variety of conventional wired or wireless computer interfaces, including serial, IEEE 1394, USB, Bluetooth™, etc., to connect to a computing device. Once connected, the microphone array provides its device description to the computing device. Sound processing software residing within the computing device is then automatically configured for optimally interacting with one or more analog or digital audio signals provided by the microphone array. In another embodiment, the microphone array performs integrated self calibration for automatically updating the device description. The self calibration is performed either upon connection to the computing device, or upon regular or user-specified intervals.

Audio optic cable

An invention is disclosed whereby an analog audio signal are transmitted over distances through a flexible medium using electro-optical radiation as the carrier of the audio signal. The analog audio signal is never converted into a digital mode nor is the signal ever filtered. The resultant analog sound that is delivered to the opposite end of the flexible medium is 100% modulated with a de minimus amount of signal loss due to noise. Though the best mode of this invention is it's application to guitar cords, any form of audio analog input will function within this invention, whereby a bundle of fiber optic cables that are capable of a high amount reflectivity, with or without the presence of an element that increases or facilitates luminescence, will become saturated with light which varies in intensity with the amplitude of the sound. This invention concedes that safety is a predominant factor when associating electrical energy with those products that people can be in contact therewith. This invention uses both direct and alternating currents but does so at very low voltages. Accordingly, safety precautions for the user and for the equipment used therewith are of the utmost concern, and the circuitry designed facilitates those desires with multiple grounding means, blocking of spurious voltages, stray capacitance and electromagnetic fields and preventing induction caused by opposing voltages.

Precoded Modulo-Precanceling Systems for Simulcasting Analog FM and Digital Data

We present techniques for simulcasting low-power digital data and analog FM over fading channels. Our methods have strong connections to what are referred to as dirty paper coding techniques in that a low-power digital data signal is modulo-added to the host analog FM signal. Due to the low power levels of the digital data and the suppression capability of analog FM, a standard analog FM receiver can be used to reliably recover the analog audio signal. We develop digital receivers that work reliably over fading channels given imperfect channel state information. Spread- response preceding is exploited to not only provide rate-1 temporal diversity benefits, but also to simplify the design of the modulo-precanceler and the receiver. Our theoretical and simulation-based performance analysis of the digital receivers and our MSE-based analog FM distortion analysis suggest that precoded modulo-precanceling schemes provide substantially higher data rates than existing precanceling strategies at the cost of transmission delay and transmitter and receiver complexity. The systems we present are also readily compatible with conventional channel coding techniques as well as with analog FM postcancelers, which can improve the digital receiver bit-error-rate performance at the expense of receiver complexity.

Voice over Internet protocol for the orthodontic practice: A sensible switch from plain old telephone service

Voice over Internet protocol (VoIP) is a revolutionary new technology that is causing a stir in the telecommunications industry and threatening the existence of traditional telephone service providers. Based on a simple method of converting analog audio signals into digital data before being transmitted over the Internet, VoIP has gained immense popularity among consumers. The technology is now regarded as an alternative to traditional telephone service for the orthodontic office. When the economics are considered, it is cost-effective, especially for a busy orthodontic practice where the call volumes both in and out are always high. VoIP has the potential to reduce costs, break the barriers between local vs long-distance calling, and make life easier for the office staff. However, deploying VoIP requires a cautious and thought-out process. Users should fully understand the risks and benefits before switching from the public switched telephone network. VoIP customers and service providers are vulnerable to many of the same impersonation-based attacks by those who attempt toll fraud, and identity and information theft. In this article, VoIP is introduced to orthodontic practitioners, who might be unfamiliar with this technology. Internet protocol based private branch exchange systems that are currently marketed as open-source technologies are also reviewed. Additionally, VoIP is compared with the traditional public switched telephone network technology and evaluated for its potential applications in an orthodontic office for both increased efficiency and cost savings.

Minimizing resource consumption for speech recognition processing with dual access buffering

In a processor system for audio processing, such as voice recognition and text-to-speech, a dedicated front-end processor, a core processor and a dedicated back-end processor are provided which are coupled by dual access stack. When an analog audio signal is inputted core processor is invoked only when a certain amount of data is present in the dual access stack. Likewise the back-end processor is invoked only when a certain amount of data is present in the dual access stack. This way the overall processing power required by the processing task is minimized as well as the power consumption of the processor system.

45. Education at distance: Broadcasting ECG rounds to Southeastern Ontario (BESO Project). An innovative approach for teaching electrocardiography

Electrocardiography (ECG) interpretation is an essential skill for physicians as well as for many other health care professionals. Continuing education is necessary to maintain these skills. The process of teaching and learning ECG interpretation is complex and involves both deductive mechanisms and recognition of patterns for different clinical situations (“pattern recognition”). The successful methodologies of interactive sessions and real time problem based learning have never been evaluated with a long distance education model.&#x0D; To evaluate the efficacy of broadcasting ECG rounds to different hospitals in the Southeastern Ontario region; to perform qualitative research to determine the impact of this methodology in developing and maintaining skills in ECG interpretation.&#x0D; ECG rounds are held weekly at Kingston General Hospital and will be transmitted live to Napanee, Belleville, Oshawa, Peterborough and Brockville. The teaching methodology is based on real ECG cases. The audience is invited to analyze the ECG case and the coordinator will introduce comments to guide the case through the proper algorithm. Final interpretation will be achieved emphasizing the deductive process and the relevance of each case. An evaluation will be filled out by each participant at the end of each session.&#x0D; Videoconferencing works through a vast array of internet LANs, WANs, ISDN phone lines, routers, switches, firewalls and Codecs (Coder/Decoder) and bridges. A videoconference Codec takes the analog audio and video signal codes and compresses it into a digital signal and transmits that digital signal to another Codec where the signal is decompressed and retranslated back into analog video and audio. This compression and decompression allows large amounts of data to be transferred across a network at close to real time (384 kbps with 30 frames of video per second). Videoconferencing communication works on voice activation so whichever site is speaking has the floor and is seen by all the participating sites. A continuous presence mode allows each site to have the same visual and audio involvement as the host site. A bridged multipoint can connect between 8 and 12 sites simultaneously.&#x0D; This innovative methodology for teaching ECG will facilitate access to developing and maintaining skills in ECG interpretation for a large number of health care providers. &#x0D; Bertsch TF, Callas PW, Rubin A. Effectiveness of lectures attended via interactive video conferencing versus in-person in preparing third-year internal medicine clerkship students for clinical practice examinations. Teach Learn Med 2007; 19(1):4-8.&#x0D; Yellowlees PM, Hogarth M, Hilty DM. The importance of distributed broadband networks to academic biomedical research and education programs. Acad Psychaitry 2006;30:451-455

Inter-channel communication in a multi-channel digital hearing instrument

A multi-channel digital hearing instrument is provided that includes a microphone, an analog-to-digital (A/D) converter, a sound processor, a digital-to-analog (D/A) converter and a speaker. The microphone receives an acoustical signal and generates an analog audio signal. The A/D converter converts the analog audio signal into a digital audio signal. The sound processor includes channel processing circuitry that filters the digital audio signal into a plurality of frequency band-limited audio signals and that provides an automatic gain control function that permits quieter sounds to be amplified at a higher gain than louder sounds and may be configured to the dynamic hearing range of a particular hearing instrument user. The D/A converter converts the output from the sound processor into an analog audio output signal. The speaker converts the analog audio output signal into an acoustical output signal that is directed into the ear canal of the hearing instrument user.