Log in

Relevant bibliographies by topics / Digital audio : Signal processing / Journal articles

To see the other types of publications on this topic, follow the link: Digital audio : Signal processing.

Journal articles on the topic 'Digital audio : Signal processing'

Author: Grafiati

Published: 4 June 2021

Last updated: 9 March 2023

Create a spot-on reference in APA, MLA, Chicago, Harvard, and other styles

Select a source type:

Consult the top 50 journal articles for your research on the topic 'Digital audio : Signal processing.'

Next to every source in the list of references, there is an 'Add to bibliography' button. Press on it, and we will generate automatically the bibliographic reference to the chosen work in the citation style you need: APA, MLA, Harvard, Chicago, Vancouver, etc.

You can also download the full text of the academic publication as pdf and read online its abstract whenever available in the metadata.

Browse journal articles on a wide variety of disciplines and organise your bibliography correctly.

1

Borawake, Prof Dr M. P. "Audio Signal Processing." International Journal for Research in Applied Science and Engineering Technology 10, no. 6 (June 30, 2022): 1495–96. http://dx.doi.org/10.22214/ijraset.2022.44063.

Full text

Abstract:

Abstract: Audio Signal Processing is also known as Digital Analog Conversion (DAC). Sound waves are the most common example of longitudinal waves. The speed of sound waves is a particular medium depends on the properties of that temperature and the medium. Sound waves travel through air when the air elements vibrate to produce changes in pressure and density along the direction of the wave’s motion. It transforms the Analog Signal into Digital Signals, and then converted Digital Signals is sent to the Devices. Which can be used in Various things., Such as audio signal, RADAR, speed processing, voice recognition, entertainment industry, and to find defected in machines using audio signals or frequencies. The signals pay important role in our day-to-day communication, perception of environment, and entertainment. A joint time-frequency (TF) approach would be better choice to effectively process this signal. The theory of signal processing and its application to audio was largely developed at Bell Labs in the mid-20th century. Claude Shannon and Harry Nyquist’s early work on communication theory and pulse-code modulation (PCM) laid the foundations for the field.

APA, Harvard, Vancouver, ISO, and other styles

2

Snyder, James H., and John Strawn. "Digital Audio Signal Processing: An Anthology." Computer Music Journal 10, no. 2 (1986): 77. http://dx.doi.org/10.2307/3679489.

Full text

APA, Harvard, Vancouver, ISO, and other styles

3

Zhang, Jing Bo, Xiao Feng Wang, and Shu Fang Zhang. "Audio Signal Processing Based on FPGA." Advanced Materials Research 1049-1050 (October 2014): 1759–64. http://dx.doi.org/10.4028/www.scientific.net/amr.1049-1050.1759.

Full text

Abstract:

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.

APA, Harvard, Vancouver, ISO, and other styles

4

Hongoh, Tsunehiko, and Hirotoshi Yamamoto. "Digital signal processing device and audio signal reproduction device." Journal of the Acoustical Society of America 120, no. 5 (2006): 2401. http://dx.doi.org/10.1121/1.2395106.

Full text

APA, Harvard, Vancouver, ISO, and other styles

5

Nasrulloh, Mohammad Dicky. "Designing a Digital Filter Based Crossover Audio System Using STM32L4." Jurnal Jartel: Jurnal Jaringan Telekomunikasi 9, no. 4 (December 25, 2019): 13–18. http://dx.doi.org/10.33795/jartel.v9i4.141.

Full text

Abstract:

Analog telecommunication system equipment is now starting to develop and be replaced with digital telecommunication systems, one of them is in the audio signal processing. The focus of audio processing is audio crossover. Audio crossover in development there are still many who use analog systems. This analog system has disadvantages when adjusting the sound balance because it still uses analog filters to balance it. It is necessary to develop a technology that aims to create a digital-based crossover audio system using the STM32L4, so that by using this digital-based signal processing it is able to adjust the sound more specifically than the signal processing used analog based. This digital filter uses the Finite Impulse Response (FIR) method. Testing audio crossover using STM32L4 produces a digital-based crossover audio system design using a STM32L4 microcontroller with a voltage of 3.3V as power supply, mp3 player as sound input device, FIR filter as digital filter processing, LM386 as sound amplifier and speaker as sound output for crossover audio on rangelow frequency (200Hz to 4000Hz), high (2200Hz to 6000Hz), medium (200Hz to 4000Hz).

APA, Harvard, Vancouver, ISO, and other styles

6

Han, Xiuqin. "Acquisition and its Basic Processing Technology of Multimedia Vocal Signal." International Journal of Pattern Recognition and Artificial Intelligence 34, no. 08 (November 12, 2019): 2058009. http://dx.doi.org/10.1142/s0218001420580094.

Full text

Abstract:

This paper briefly studies the method of collecting audio signals and the method of adding noise to audio signals. It comprehensively applies various basic knowledge of digital signal processing, and then performs spectrum analysis on noise-free frequency signals and spectral analysis of noise-added frequency signals, and filtering processing. Through theoretical derivation, the corresponding conclusions are drawn, and then MATLAB is used as a programming tool to carry out computer implementation to verify the conclusions derived. In the research process, the filter processing was completed by designing the IIR digital filter and the FIR digital filter, and MATLAB was used to draw the graphics and calculate and simulate some data in the whole design.

APA, Harvard, Vancouver, ISO, and other styles

7

Tsai, S. E., and S. M. Yang. "An Effective Watermarking Method Based on Energy Averaging in Audio Signals." Mathematical Problems in Engineering 2018 (June 25, 2018): 1–8. http://dx.doi.org/10.1155/2018/6420314.

Full text

Abstract:

Methods based on discrete cosine transform (DCT) have been proposed for digital watermarking of audio signals; however, the watermark is often vulnerable to data compression and signal processing. This paper presents an effective audio watermarking method by energy averaging of DCT coefficients such that an audio signal with watermark is robust to data processing. The method is to divide an audio signal into segments by three parameters defining the segment length, the segment sequence of watermark location, and the frequency range of DCT coefficients for watermark location. An error correcting code is also integrated to improve audio signal quality after watermarking. Experimental results show that the method is robust to data compression and many other kinds of signal processing. No original signal is required for decoding the watermark. Comparison of watermarking performance with a recent work validates that the watermarking method has better audio quality and higher robustness.

APA, Harvard, Vancouver, ISO, and other styles

8

Lim, Dukhwan. "Digital Signal Processing in Audiology." Audiology and Speech Research 4, no. 1 (June 30, 2008): 5–10. http://dx.doi.org/10.21848/audiol.2008.4.1.5.

Full text

APA, Harvard, Vancouver, ISO, and other styles

9

Mercs, Laura, and Paul M. Embree. "Audio noise reduction system implemented through digital signal processing." Journal of the Acoustical Society of America 108, no. 2 (2000): 474. http://dx.doi.org/10.1121/1.429557.

Full text

APA, Harvard, Vancouver, ISO, and other styles

10

Liu, Bin, and Yan Ren. "A design of laser array harp based on multi-dimensional wavelet transform and audio signal reconstruction." Journal of Physics: Conference Series 2113, no. 1 (November 1, 2021): 012059. http://dx.doi.org/10.1088/1742-6596/2113/1/012059.

Full text

Abstract:

Abstract This paper introduces a design scheme of laser array harp based on multi-dimensional wavelet transform and audio signal reconstruction. The green light beams from multiple high-power lasers simulate harp strings, use photoresistors as the signal receiving end, and use a signal conditioning system composed of analog circuits and LM393 comparators to collect and adjust the resistance signal of the laser sensor[1], and finally it is adjusted to a level signal that can be recognized by the CPU. After receiving the signal, the CPU core board analyzes the string signal, and sends control commands to the audio processing system through the industrial bus according to the analyzed digital signal. After receiving the control command, the audio processing system uses the audio signal reconstruction technology composed of multi-dimensional wavelet packets, deep learning and other algorithms to simulate the audio signals of various string music, so as to achieve the purposes of using the lasers as virtual strings and imitating musical instruments for musical performance.[2]

APA, Harvard, Vancouver, ISO, and other styles

11

Yin, Shu Hua. "Design of the Auxiliary Speech Recognition System of Super-Short-Range Reconnaissance Radar." Applied Mechanics and Materials 556-562 (May 2014): 4830–34. http://dx.doi.org/10.4028/www.scientific.net/amm.556-562.4830.

Full text

Abstract:

To improve the usability and operability of the hybrid-identification reconnaissance radar for individual use, a voice identification System was designed. By using SPCE061A audio signal microprocessor as the core, a digital signal processing technology was used to obtain Doppler radar signals of audio segments by audio cable. Afterwards, the A/D acquisition was conducted to acquire digital signals, and then the data obtained were preprocessed and adaptively filtered to eliminate background noises. Moreover, segmented FFT transforming was used to identify the types of the signals. The overall design of radar voice recognition for an individual soldier was thereby fulfilled. The actual measurements showed that the design of the circuit improved radar resolution and the accuracy of the radar identification.

APA, Harvard, Vancouver, ISO, and other styles

12

Sangwine, S. J. "A Digital Signal Processing Laboratory Based on the TMS320C25." International Journal of Electrical Engineering & Education 32, no. 1 (January 1995): 21–30. http://dx.doi.org/10.1177/002072099503200103.

Full text

Abstract:

A digital signal processing laboratory based on the TMS320C25 Students on a B. Eng. degree course at the University of Reading take a 20 hour lecture course on DSP and 15 hours of laboratory work using an audio-band DSP system designed around the Texas TMS320C25 DSP chip. The course and DSP system are described and experiences and conclusions are drawn.

APA, Harvard, Vancouver, ISO, and other styles

13

Nkurikiyeyezu, Kizito, Faustin Ahishakiye, Cyprien Nsengimana, and Etienne Ntagwirumugara. "Toolkits for Real Time Digital Audio Signal Processing Teaching Laboratory." Journal of Signal and Information Processing 06, no. 02 (2015): 92–98. http://dx.doi.org/10.4236/jsip.2015.62009.

Full text

APA, Harvard, Vancouver, ISO, and other styles

14

Agnello, Anthony, and Steven Hoge. "A Development System for Real-Time Digital Audio Signal Processing." Computer Music Journal 9, no. 3 (1985): 24. http://dx.doi.org/10.2307/3679574.

Full text

APA, Harvard, Vancouver, ISO, and other styles

15

Bruce, Lori Mann, Navaneethakrishnan Balraj, Yunlong Zhang, and Qingyong Yu. "Automated Accident Detection in Intersections via Digital Audio Signal Processing." Transportation Research Record: Journal of the Transportation Research Board 1840, no. 1 (January 2003): 186–92. http://dx.doi.org/10.3141/1840-21.

Full text

Abstract:

A system for automated traffic accident detection in intersections was designed. The input to the system is a 3-s segment of audio signal. The system can be operated in two modes: the two-class and multiclass modes. The output of the two-class mode is a label of “crash” or “noncrash.” In the multiclass mode of operation, the system identifies crashes as well as several types of noncrash incidents, including normal traffic and construction sounds. The system is composed of three main signal processing stages: feature extraction, feature reduction, and classification. Five methods of feature extraction were investigated and compared; these are based on the discrete wavelet transform, fast Fourier transform, discrete cosine transform, real cepstral transform, and mel frequency cepstral transform. Statistical methods are used for feature optimization and classification. Three types of classifiers are investigated and compared; these are the nearest-mean, maximum-likelihood, and nearest-neighbor methods. The results of the study show that the optimum design uses wavelet-based features in combination with the maximum-likelihood classifier. The system is computationally inexpensive relative to the other methods investigated, and the system consistently results in accident detection accuracies of 95% to 100% when the audio signal has a signal-to-noise-ratio of at least 0 decibels.

APA, Harvard, Vancouver, ISO, and other styles

16

Chen, Chui Xin, and Yang Hong Mao. "Design and Implementation of Audio Process System Based on DSP." Advanced Materials Research 945-949 (June 2014): 1752–55. http://dx.doi.org/10.4028/www.scientific.net/amr.945-949.1752.

Full text

Abstract:

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.

APA, Harvard, Vancouver, ISO, and other styles

17

Perng, Jau-Woei, Tung-Li Hsieh, and Cheng-Yan Guo. "A Novel Dentary Bone Conduction Device Equipped with Laser Communication in DSP." Sensors 21, no. 12 (June 20, 2021): 4229. http://dx.doi.org/10.3390/s21124229.

Full text

Abstract:

In this study, we designed a dentary bone conduction system that transmits and receives audio by laser. The main objective of this research was to propose a complete hardware design method, including a laser audio transmitter and receiver and digital signal processor (DSP) based digital signal processing system. We also present a digital filter algorithm that can run on a DSP in real time. This experiment used the CMU ARCTIC databases’ human-voice reading audio as the standard audio. We used a piezoelectric sensor to measure the vibration signal of the bone conduction transducer (BCT) and separately calculated the signal-to-noise ratio (SNR) of the digitally filtered audio output and the unfiltered audio output using DSP. The SNR of the former was twice that of the latter, and the BCT output quality significantly improved. From the results, we can conclude that the dentary bone conduction system integrated with a DSP digital filter enhances sound quality.

APA, Harvard, Vancouver, ISO, and other styles

18

Xue, Yan, and Fei Yang. "Asynchronous Sampling Rate Conversion of Digital Audio Signal." Applied Mechanics and Materials 687-691 (November 2014): 4093–96. http://dx.doi.org/10.4028/www.scientific.net/amm.687-691.4093.

Full text

Abstract:

At present, in the digital audio processing sampling rate is respectively 32 kHz, 44.1 kHz, 48 kHz [1]. Because of the different criteria, there is much inconvenience in the process of research. Therefore, the sampling rate converter is a must, between any two kinds of sampling rate. In synchronous sampling rate conversion, you can use decimation and interpolation for sampling rate conversion, but in the asynchronous sampling rate system, due to the different input clock pulse with the output clock pulse, the above method cannot achieve. Therefore we introduce the fractional delay filter sampling rate conversion. This article introduces the principle of the sampling rate conversion and the fractional delay filter based on Farrow structure. At last, we simulate asynchronous sampling rate conversion of audio signal through the MATLAB.

APA, Harvard, Vancouver, ISO, and other styles

19

Dawson, Andrew, and Ken Steiglitz. "A Digital Signal Processing Primer, with Applications to Digital Audio and Computer Music." Computer Music Journal 22, no. 2 (1998): 68. http://dx.doi.org/10.2307/3680970.

Full text

APA, Harvard, Vancouver, ISO, and other styles

20

Morley, R., A. Engebretson, and J. Trotta. "A multiprocessor digital signal processing system for real-time audio applications." IEEE Transactions on Acoustics, Speech, and Signal Processing 34, no. 2 (April 1986): 225–31. http://dx.doi.org/10.1109/tassp.1986.1164828.

Full text

APA, Harvard, Vancouver, ISO, and other styles

21

Konishi, K., H. Hitomi, H. Naka, K. Oishi, and M. Yamazaki. "An EM audio LSI for VHS VCRs using digital signal processing." IEEE Transactions on Consumer Electronics 36, no. 3 (1990): 628–34. http://dx.doi.org/10.1109/30.103184.

Full text

APA, Harvard, Vancouver, ISO, and other styles

22

Iwaki, T., T. Okuda, K. Koyanagi, Y. Yokomachi, C. Yamawaki, and T. Sasada. "Signal processing of a 20-bit 8-channel digital audio recorder." IEEE Transactions on Consumer Electronics 36, no. 3 (1990): 647–54. http://dx.doi.org/10.1109/30.103187.

Full text

APA, Harvard, Vancouver, ISO, and other styles

23

Potchinkov, Alexander. "Digital signal processing methods of global nonparametric frequency domain audio testing." Signal Processing 85, no. 6 (June 2005): 1225–54. http://dx.doi.org/10.1016/j.sigpro.2004.12.007.

Full text

APA, Harvard, Vancouver, ISO, and other styles

24

Gaydecki, Patrick. "The Foundations of Digital Signal Processing Using Signal Wizard Systems®." International Journal of Electrical Engineering & Education 49, no. 3 (July 2012): 310–20. http://dx.doi.org/10.7227/ijeee.49.3.10.

Full text

Abstract:

Signal Wizard Systems® is a digital signal processing (DSP) research venture within the School of EEE at the University of Manchester, UK. It specialises in the development and supply of real-time DSP products for audio signal analysis and processing. The unique and underpinning philosophy of these products is their ease of use. The systems require minimal knowledge of DSP theory on the part of the user and none of the mathematics associated with digital filter design. Filters and other algorithms can be designed in seconds, downloaded and executed in real time with just a few mouse clicks. Since 2004 Signal Wizard products have been sold all over the world for applications ranging from noise suppression, adaptive filtering and system modelling to musical instrument research. In particular, their ease of use ensures that they are ideally suited for teaching simple and more advanced concepts in DSP both at undergraduate and postgraduate level. For this purpose, a DSP laboratory teaching package has been developed using the Signal Wizard range of devices, and has proven an invaluable tool for training our student cohort in the practical aspects of DSP engineering design and programming.

APA, Harvard, Vancouver, ISO, and other styles

25

Chen, Aiyong. "Digital Filtering Technology in Industrial Measuring and Control System." Electronics Science Technology and Application 2 (December 3, 2015): 42. http://dx.doi.org/10.18686/esta.v2i1.8.

Full text

Abstract:

<p>This article aims at the technical problems in modernized industrial measuring and control system such as interruption signal, noise signal and other useless signal. First introduces the features and importance of digital filtering technology and then elaborates on the realization methods of digital filtering and frequently used digital filtering calculation methods in the industrial measuring system. The research reveals that integrated usage of numerous methods or even complex digital filtering technology is adopted to calculate and treat such digital signals like random interruption, heat noise, system noise, measuring error and zero-point offset and thus meet the system requirements. The digital filtering technology is widely applied in the processing of HD signal, such as digital audio, radar, image processing, data transmission and biological and medical fields and pledges to provide strong assurance for the real-time, stable and reliable properties of modernized industrial measuring and control system.</p><div> </div><div> </div>

APA, Harvard, Vancouver, ISO, and other styles

26

Liu, Yaqing, and Lunhui Deng. "Design of Audio Embedding and De-embedding for 3G-SDI Based on FPGA." MATEC Web of Conferences 173 (2018): 03021. http://dx.doi.org/10.1051/matecconf/201817303021.

Full text

Abstract:

This design introduces the theoretical basis of digital audio embedding and de-embedding, and proposes a solution that Verilog language can be used to achieve 3G-SDI audio embedding and de-embedding. SDI video and audio data are input to the FPGA, and the audio signals can be embedded in the SDI line blanking after processing. Moreover, some auxiliary information is embedded in the SDI data, when you need these auxiliary information, you need to use the audio de-embedding process. The process of audio de-embedding is inversed with the process of embedding. It has been proved through practice that this scheme can effectively embed digital audio in SDI data stream, synchronize audio and video data, and can de-embed audio signal. The design is very versatile and can improve the efficiency of the design, thus effectively reducing the cost of the product.

APA, Harvard, Vancouver, ISO, and other styles

27

Wang, Chun Mei, and Wei Cai Xiao. "Second-Order IIR Notch Filter Design and Implementation of Digital Signal Processing System." Applied Mechanics and Materials 347-350 (August 2013): 729–32. http://dx.doi.org/10.4028/www.scientific.net/amm.347-350.729.

Full text

Abstract:

In this paper the AC power 50Hz power interference, we use IIR digital notch filter method for industrial frequency interference filter. From the design of IIR digital filter method proceed with, on the IIR digital notch filter simulation, the algorithm deduced, on the fixed-point DSP programming method and overflow handling problems made elaborate incisively, and in digital audio signal processing system has been applied.

APA, Harvard, Vancouver, ISO, and other styles

28

Mizushima, T., K. Hashimoto, S. Fujii, E. Yamauchi, K. Kawakami, M. Okabe, T. Kashiro, M. Mitsuda, T. Nakagawa, and M. Satoh. "The development of audio and video signal processing LSI for digital VCR." IEEE Transactions on Consumer Electronics 43, no. 3 (1997): 344–51. http://dx.doi.org/10.1109/30.628623.

Full text

APA, Harvard, Vancouver, ISO, and other styles

29

Abdulsada, Ayad. "A Robust Wavelet Based Watermarking Scheme For Digital Audio." Iraqi Journal for Electrical and Electronic Engineering 8, no. 8 (June 1, 2012): 65–72. http://dx.doi.org/10.37917/ijeee.8.1.7.

Full text

Abstract:

In this paper, a robust wavelet based watermarking scheme has been proposed for digital audio. A single bit is embedded in the approximation part of each frame. The watermark bits are embedded in two subsets of indexes randomly generated by using two keys for security purpose. The embedding process is done in adaptively fashion according to the mean of each approximation part. The detection of watermark does not depend on the original audio. To measure the robustness of the algorithm, different signal processing operations have been applied on the watermarked audio. Several experimental results have been conducted to illustrate the robustness and efficiency of the proposed watermarked audio scheme.

APA, Harvard, Vancouver, ISO, and other styles

30

Aggarwal, Shruti, Vasukidevi G, S. Selvakanmani, Bhaskar Pant, Kiranjeet Kaur, Amit Verma, and Geleta Negasa Binegde. "Audio Segmentation Techniques and Applications Based on Deep Learning." Scientific Programming 2022 (August 19, 2022): 1–9. http://dx.doi.org/10.1155/2022/7994191.

Full text

Abstract:

Audio processing has become an inseparable part of modern applications in domains ranging from health care to speech-controlled devices. In automated audio segmentation, deep learning plays a vital role. In this article, we are discussing audio segmentation based on deep learning. Audio segmentation divides the digital audio signal into a sequence of segments or frames and then classifies these into various classes such as speech recognition, music, or noise. Segmentation plays an important role in audio signal processing. The most important aspect is to secure a large amount of high-quality data when training a deep learning network. In this study, various application areas, citation records, documents published year-wise, and source-wise analysis are computed using Scopus and Web of Science (WoS) databases. The analysis presented in this paper supports and establishes the significance of the deep learning techniques in audio segmentation.

APA, Harvard, Vancouver, ISO, and other styles

31

Chen, Bo, Heung Kou, Bowen Hou, and Yanbing Zhou. "Music Feature Extraction Method Based on Internet of Things Technology and Its Application." Computational Intelligence and Neuroscience 2022 (April 18, 2022): 1–10. http://dx.doi.org/10.1155/2022/8615152.

Full text

Abstract:

Due to the influence of factors such as strong music specialization, complex music theory knowledge, and various variations, it is difficult to identify music features. We have developed a music characteristic identification system using the Internet-based method. The physical sensing layer of our designed system deploys audio sensors on various coordinates to capture the raw audio signal and performs audio signal processing and analysis using the TMS320VC5402 digital signal processor; the Internet transport layer places audio sensors at various locations to capture the raw audio signal. The TMS320VC5402 digital signal processor is used for audio signal diagnosis and treatment. The network transport layer transmits the finished audio signal to the data base of song signal in the application layer of the system; the song characteristic analysis block in the application layer adopts dynamics. The music characteristic analysis block in the applied layer adopts dynamic time warping algorithm to acquire the maximal resemblance between the test template and the reference template to achieve music signal characteristic identification and identify music tunes and music modes based on the identification results. The application layer music feature analysis block adopts dynamic time regularization algorithm and mel-frequency cepstrum coefficient to achieve music signal feature recognition and identify music tunes and music patterns based on the recognition results. We have verified through experiments, and the results show that the system operates consistently, can obtain high-quality music samples, and can extract good music characteristics.

APA, Harvard, Vancouver, ISO, and other styles

32

Sokolov, Sergei A., and Yuri A. Kovalgin. "INFLUENCE OF THE COMPRESSION ALGORITHMS ON THE QUALITY OF AUDIO PROGRAMS IN DRM DIGITAL RADIO BROADCASTING SYSTEM." T-Comm 15, no. 7 (2021): 4–13. http://dx.doi.org/10.36724/2072-8735-2021-15-7-4-13.

Full text

Abstract:

This work studied the influence of MPEG-4 HE-AAC v.2 and MPEG-4 xHE-AAC compression algorithms on the reproduction performance of high-quality audio signals. The first part of the article is based on the analysis of earlier publications, studied quality of the compression algorithms for MPEG-4 ISO/IEC 14493-3 (AAC, AAC+SBR, PS, AAC+SBR+PS) and MPEG D Surround standards applied in MPEG-4 HE-AAC v.2 and MPEG-4 xHE-AAC codecs of DAB and DRM systems for digital audio broadcasting, respectively. It has been demonstrated that their use for processing of high-quality audio signals in the frequency range of 40…15000 Hz, allowed to reduce the digital-data rate at the output to 24…30 êbit/s, without observing noticeable phenomena. This conclusion was confirmed in the second part of the article, which described the results of relative comparison of the quality of reproduction of audio signals broadcasted via analogues FM or digital DRM channels. The equipment employed for their realization was described, as well as the methodology to obtain test recordings in order to perform the statistical data analysis. Further, the specific data scales were shown to evaluate the signal pairs being compared. Results of the experiments and their analysis confirmed that, for a digital data rate of 30 êbit/s obtained at the output of the MPEG-4 xHE-AAC coder, the difference in quality was rather weak for audio signals ranging from 40 to 15000 Hz, broadcasted via both analogues and digital channels. In addition, only a marginal difference was noticed for the signal pairs received from the digital output channels of the transmitter processor and DRM receiver. It has been demonstrated that the MPEG-4 xHE-AAC codec, compared to the previous MPEG-4 HE-AAC v.2 version, exhibited clear advantages for practical applications in terms of sound quality. Based on the gained experience, common criteria of deterioration of the data quality were formulated and generalized for all audio compression MPEG standards.

APA, Harvard, Vancouver, ISO, and other styles

33

Putra A, I. Nengah, Nuri Nur Cahyono, Gesit Pratiknyo, and M. Sigit Purwanto. "DESIGN OF DETECTING ACOUSTIC WAVES AT THE EXERCISE SMART MINE USING ACOUSTIC SENSOR." JOURNAL ASRO 11, no. 2 (April 20, 2020): 64. http://dx.doi.org/10.37875/asro.v11i2.270.

Full text

Abstract:

The Indonesian Navy is a unit of defense of the Republic of Indonesia (NKRI). Sea mines are explosive devices placed in waters to destroy ships. Acoustics is a branch of physics that deals with all mechanical waves in liquids, gases and solids, such as vibration, sound and ultrasonic. Arduino has made microcontroller technology more accessible and easier for even novice users. Because of this, there are special functions in audio or acoustic processing techniques aimed at using Arduino. The application of acoustic signals in the military field of the Navy, in the identification of vessels caught by the microphone receiver. The movement of objects that are caught by the microphone will be analyzed to conclude the object. From the results of the sound characteristics raised by the object, it will give a different picture for different variations of the object. Processing acoustic signals using DSP will make it easier to infer the captured acoustic signal. The circuit board used is a microphone signal amplifier and signal conditioner, for processing the audio signal obtained from the microphone and to produce output data which is the result of the process using Arduino. Analog-to-digital converter (ADC) of the microcontroller works in the voltage range from 0.0 V to +3.3 V or +5.0 V. Toachieve good sampling results, that the signal peak is close to the maximum value of the ADC. Additionally, voltage amplitudes above this threshold can damage the controller input port or produce unwanted harmonics. Therefore, a signal pre-amp circuit is needed to guarantee the appropriate audio sample signal as well as to protect the system. The test results obtained from the system analyzer frequency and signal meters can be used to identify acoustic signals. Keyword: Singnal Acoustic/michophone, controller, Arduino Uno, pre-amp frequency analyzer

APA, Harvard, Vancouver, ISO, and other styles

34

Yultrisna, Yultrisna, and Andi Syofyan. "SINUSOIDAL NOISE CANCELATION DENGAN MENGGUNAKAN DIGITAL SIGNAL PROCESSING STARTER KIT TMS320C6713." Elektron : Jurnal Ilmiah 4, no. 2 (December 10, 2012): 67–74. http://dx.doi.org/10.30630/eji.4.2.33.

Full text

Abstract:

Original speech signal is needed both in telecommunications and in some instruments in a variety of fields. Not infrequently, the original audio signal is damaged due to noise. This noise can cause the original signal changes in the actual form. In the final project will be designed FIR filter to remove noise by using TMS320C6713 DSK. Sound signal to be input to the mixed noise removal filter system noise. The mixed voice signal will be searched by subtracting the signal difference to noise signal output FIR filter to get the signal e (n), and then do an adaptation resulting filter coefficients. Results of the adaptive filter coefficients would be put back to calculate the noise signal output next FIR filter. Original voice signal used is the word "sinus" uttered by teenage boys, teenage girls, boys and girls. Girl's voice had the highest frequency with an average 522.50 Hz, the frequency of the sound of the boys 462.63 Hz, the sound frequency of 222.58 Hz girls and voice frequencies teenage boys at 201.49 Hz. Noise signal used is 100 Hz sinusoidal noise. From the test results obtained for the system output signal SNR sound input teenage boy was 19.94 dB. SNR output signal to the input of 21.39 dB girls. SNR signal input output system for boys was 34.70 dB. SNR output signal to the input system daughters of 35.52 dB

APA, Harvard, Vancouver, ISO, and other styles

35

Gilski, Przemysław, Sławomir Gajewski, and Jacek Stefański. "Quality Aspects in Digital Broadcasting and Webcasting Systems: Bitrate versus Loudness." Journal of Telecommunications and Information Technology, no. 2 (June 30, 2017): 26–31. http://dx.doi.org/10.26636/jtit.2017.114217.

Full text

Abstract:

In this paper the quality aspects of bitrate and loudness in digital broadcasting and webcasting systems are examined. The authors discuss a survey concerning user preferences related with processing and managing audio content. The coding efficiency of a popular audio format is analyzed in the context of storing media. An objective study on a representative group of signal samples, as well as a subjective study of the perceived quality of real-time broadcasted and webcasted radio programs are performed.

APA, Harvard, Vancouver, ISO, and other styles

36

Spirintsev, Vyacheslav, Dmitry Popov, and Olga Spirintseva. "VIRTUAL DIGITAL ASSISTANT WITH VOICE INTERFACE SUPPORT." System technologies 2, no. 133 (March 1, 2021): 42–51. http://dx.doi.org/10.34185/1562-9945-2-133-2021-06.

Full text

Abstract:

A virtual digital assistant which can work with arbitrary systems and provide an effective solution of narrowly focused user tasks for interaction with Ukrainian services voice inter-face supported has been proposed. The developed web service was implemented by using the PHP programming language, Wit.ai service for audio signal processing, FANN library for neural network construction, Telegram service for creating an interface.

APA, Harvard, Vancouver, ISO, and other styles

37

Greenspun, Philip. "Audio Analysis V: Time-and Frequency-Domain Distortions in Digital Signal Processing Systems." Computer Music Journal 10, no. 4 (1986): 79. http://dx.doi.org/10.2307/3680099.

Full text

APA, Harvard, Vancouver, ISO, and other styles

38

Escola, João Paulo Lemos, Uender Barbosa de Souza, Rodrigo Capobianco Guido, Ivan Nunes da Silva, Jovander da Silva Freitas, and Lucas de Araújo Oliveira. "A mesh network case study for digital audio signal processing in Smart Farm." Internet of Things 17 (March 2022): 100488. http://dx.doi.org/10.1016/j.iot.2021.100488.

Full text

APA, Harvard, Vancouver, ISO, and other styles

39

Caputi, M. J. "Developing real-time digital audio effects for electric guitar in an introductory digital signal processing class." IEEE Transactions on Education 41, no. 4 (November 1998): 341. http://dx.doi.org/10.1109/te.1998.787367.

Full text

APA, Harvard, Vancouver, ISO, and other styles

40

Caputa, M. J. "Developing real-time digital audio effects for electric guitar in an introductory digital signal processing class." IEEE Transactions on Education 41, no. 4 (1998): 10 pp. http://dx.doi.org/10.1109/13.728274.

Full text

APA, Harvard, Vancouver, ISO, and other styles

41

Elshazly, A. R., Mohamed E. Nasr, M. M. Fouad, and Fathi E. Abdel-Samie. "Intelligent High Payload Audio Watermarking Algorithm Using Colour Image in DWT-SVD Domain." Journal of Physics: Conference Series 2128, no. 1 (December 1, 2021): 012019. http://dx.doi.org/10.1088/1742-6596/2128/1/012019.

Full text

Abstract:

Abstract Copyright protection and ownership verification of digital audio tracks have become increasingly important to be enabled by digital watermarking techniques. A novel high payload intelligent audio watermarking scheme with RGB color watermark image is proposed in this paper. The color watermark image is encrypted using Arnold chaotic map and passed through an adaptive scaling filter to scale the image to match the required payload. The encoding process is performed on the scaled encrypted version of the watermark image. A portion of the audio signal is used to embed a synchronization code and the other one is decomposed into short frames. These frames are processed with a two-level discrete wavelet transform (DWT), followed by a singular value decomposition (SVD) process on the approximation coefficients. The encoded watermark is inserted into the diagonal matrix using quantization index modulation (QIM). The inverse process of SVD and DWT is applied to obtain the marked audio signal. Blind extraction of the hidden information from the marked audio signal is performed in the reverse order of the embedding process. Experiments show that security, high payload, transparency and imperceptibility of the algorithm are satisfied. The robustness against several kinds of audio signal processing attacks is shown. Performance evaluation tests with SNR, BER, and FSIM are conducted.

APA, Harvard, Vancouver, ISO, and other styles

42

Agrawal, S. K., and O. P. Sahu. "Two-Channel Quadrature Mirror Filter Bank: An Overview." ISRN Signal Processing 2013 (September 3, 2013): 1–10. http://dx.doi.org/10.1155/2013/815619.

Full text

Abstract:

During the last two decades, there has been substantial progress in multirate digital filters and filter banks. This includes the design of quadrature mirror filters (QMF). A two-channel QMF bank is extensively used in many signal processing fields such as subband coding of speech signal, image processing, antenna systems, design of wavelet bases, and biomedical engineering and in digital audio industry. Therefore, new efficient design techniques are being proposed by several authors in this area. This paper presents an overview of analysis and design techniques of the two-channel QMF bank. Application in the area of subband coding and future research trends are also discussed.

APA, Harvard, Vancouver, ISO, and other styles

43

Sinha, Ria. "Digital Assistant for Sound Classification Using Spectral Fingerprinting." International Journal for Research in Applied Science and Engineering Technology 9, no. 8 (August 31, 2021): 2045–52. http://dx.doi.org/10.22214/ijraset.2021.37714.

Full text

Abstract:

Abstract: This paper describes a digital assistant designed to help hearing-impaired people sense ambient sounds. The assistant relies on obtaining audio signals from the ambient environment of a hearing-impaired person. The audio signals are analysed by a machine learning model that uses spectral signatures as features to classify audio signals into audio categories (e.g., emergency, animal sounds, etc.) and specific audio types within the categories (e.g., ambulance siren, dog barking, etc.) and notify the user leveraging a mobile or wearable device. The user can configure active notification preferences and view historical logs. The machine learning classifier is periodically trained externally based on labeled audio sound samples. Additional system features include an audio amplification option and a speech to text option for transcribing human speech to text output. Keywords: assistive technology, sound classification, machine learning, audio processing, spectral fingerprinting

APA, Harvard, Vancouver, ISO, and other styles

44

Sikora, T. "MPEG Digital Audio-and Video-Coding Standards." IEEE Signal Processing Magazine 14, no. 5 (September 1997): 58. http://dx.doi.org/10.1109/msp.1997.618008.

Full text

APA, Harvard, Vancouver, ISO, and other styles

45

Lattner, Stefan, and Javier Nistal. "Stochastic Restoration of Heavily Compressed Musical Audio Using Generative Adversarial Networks." Electronics 10, no. 11 (June 5, 2021): 1349. http://dx.doi.org/10.3390/electronics10111349.

Full text

Abstract:

Lossy audio codecs compress (and decompress) digital audio streams by removing information that tends to be inaudible in human perception. Under high compression rates, such codecs may introduce a variety of impairments in the audio signal. Many works have tackled the problem of audio enhancement and compression artifact removal using deep-learning techniques. However, only a few works tackle the restoration of heavily compressed audio signals in the musical domain. In such a scenario, there is no unique solution for the restoration of the original signal. Therefore, in this study, we test a stochastic generator of a Generative Adversarial Network (GAN) architecture for this task. Such a stochastic generator, conditioned on highly compressed musical audio signals, could one day generate outputs indistinguishable from high-quality releases. Therefore, the present study may yield insights into more efficient musical data storage and transmission. We train stochastic and deterministic generators on MP3-compressed audio signals with 16, 32, and 64 kbit/s. We perform an extensive evaluation of the different experiments utilizing objective metrics and listening tests. We find that the models can improve the quality of the audio signals over the MP3 versions for 16 and 32 kbit/s and that the stochastic generators are capable of generating outputs that are closer to the original signals than those of the deterministic generators.

APA, Harvard, Vancouver, ISO, and other styles

46

Hemis, Mustapha, Bachir Boudraa, and Thouraya Merazi-Meksen. "New secure and robust audio watermarking algorithm based on QR factorization in wavelet domain." International Journal of Wavelets, Multiresolution and Information Processing 13, no. 03 (May 2015): 1550020. http://dx.doi.org/10.1142/s0219691315500204.

Full text

Abstract:

Digital watermarking consists in embedding imperceptible information into a host signal. It has been proposed to solve problems as varied as the protection of the copyright, content authentication, fingerprinting and broadcast monitoring. This paper presents a new approach for audio watermarking using the QR factorization in wavelet domain. This approach is based on embedding a watermark binary image in the R matrices of low frequency blocks DWT coefficients of audio signal. In this algorithm, the watermark is embedded by applying a Quantization Index Modulation (QIM) process on the determined optimal sample for each matrix R. The watermark can be blindly extracted without the knowledge of the original audio signal. Experimental results show that the proposed audio watermarking scheme maintains high quality of the audio signal. Signal to Noise Ratio (SNR), Log Spectral Distortion (LSD) and Mean Opinion Score (MOS) are about 40 dB, 0.37 dB and 4.84, respectively. Moreover, the scheme is quite robust against common signal processing attacks such as noise addition, filtering and MP3 compression. In addition, this method ensures a secure extraction process by using a private key, making it suitable for secure applications such as copyright protection.

APA, Harvard, Vancouver, ISO, and other styles

47

Petrovsky, Alexander, Wanggen Wan, Manuel Rosa-Zurera, and Alexey Karpov. "Signal Processing Platforms and Algorithms for Real-Life Communications and Listening to Digital Audio." Journal of Electrical and Computer Engineering 2017 (2017): 1–2. http://dx.doi.org/10.1155/2017/2913236.

Full text

APA, Harvard, Vancouver, ISO, and other styles

48

Zahringer, E. "TV Multichannel Audio - A Chipset for US Stereo TV Receivers Using Digital Signal Processing." IEEE Transactions on Consumer Electronics CE-31, no. 3 (August 1985): 469–73. http://dx.doi.org/10.1109/tce.1985.289961.

Full text

APA, Harvard, Vancouver, ISO, and other styles

49

Ramos dos Santos, Marcos, and Andres Eduardo Coca Salazar. "Optical Digital Theremin with Audio Synthesis and Graphic Interface." International Journal of Circuits, Systems and Signal Processing 15 (November 3, 2021): 1613–23. http://dx.doi.org/10.46300/9106.2021.15.174.

Full text

Abstract:

he theremin is one of the first elec- tronic musical instruments and one of the few played without physical contact since it only re- quires hand and finger movements to control the amplitude and frequency of the musical note. However, the capacitive functioning of the anten- nas increases the sensitivity to electrical interfer- ence, its timbre is fixed, and the frequency an- tenna's vertical arrangement could limit the use of people with amputated fingers. Furthermore, it does not contain any help to guide the execu- tion, which makes it a very difficult instrument to play. In this paper, we present the development of a digital optical theremin with an audio syn- thesis process, intuitive graphical interface, frequency antenna in the horizontal position, and linearization of the frequency-distance relationship. These features are intended to aid learn- ing and interpretation of the instrument and ex- tend access to people with finger limitations. In order to validate the instrument's behavior and characteristics, we conducted three experiments: 1) accuracy analysis of the linearization through the mean absolute error in units of cents and the Kruskal-Wallis statistical inference test, 2) val- idation of the steps of the audio synthesis mod- ule, and 3) checking of the timbral diversity, both through the Fourier spectrum. This prototype could be used as an auxiliary tool in musical initi- ation and the development of musical perception.

APA, Harvard, Vancouver, ISO, and other styles

50

Battisti, Luca, Angelo Farina, Antonella Bevilacqua, and Antonella Bevilacqua. "Implementation of non-equal-partition multi-channel convolver." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 265, no. 6 (February 1, 2023): 1570–81. http://dx.doi.org/10.3397/in_2022_0220.

Full text

Abstract:

Convolution has become a largely exploited signal operation thanks to his several applications in digital signal processing. In the realm of audio elaboration, convolution has the particular meaning of imposing a spectral and/or temporal structure onto a sound. These structures are completely given by the signal with which the signal is being convolved, called Impulse Response (IR). These signals contain a sort of acoustical footprint that can be completely transferred to another sound, earning the same acoustic characteristics as a consequence. With a multichannel approach, convolution assumes even a further meaning and a wider application field. Indeed, it's exploited to deal with modern spatial sound techniques such as Ambisonics which necessitate matrix elaborations of the involved signals. Ambisonics recordings, for example, are made by special coincident multi-capsule microphone arrays, whose signals can be converted to standard Ambisonics format by a multi-channel convolver. A similar concept can apply to the mixing stage of audio production, where direction-based audio objects must be converted to the Ambisonics format to be reproduced in the relative speaker setups. The aim of the work is to analyse an existing algorithm of a multichannel convolver software evaluating his efficiency. Moreover, the managing of the matrix of filters has showed weaknesses when assembling new matrices. Solution proposes a handy way to deal with matrices and to improve the efficiency of the algorithm.

APA, Harvard, Vancouver, ISO, and other styles

We offer discounts on all premium plans for authors whose works are included in thematic literature selections. Contact us to get a unique promo code!