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About 419 papers for “Audio analyzer”

Audio Analyzer

France, Jeffrey M · Opus: Research & Creativity (Indiana University – Purdue University Fort Wayne) · 1990

Cited by 0

Audio spectrum analyzer

Ovsec, Janez · Repository of the University of Ljubljana (University of Ljubljana) · 2025

V diplomski nalogi je bil zasnovan, izdelan in funkcionalno preizkušen avdio spektralni analizator, ki temelji na mikrokrmilniku Teensy 4.0, z namenom prikaza frekvenčne vsebine analognega zvočnega signala v realnem času. Naprava je bila razvita kot samostojen sistem, ki zajema zvočni signal preko elektretnega mikrofona, ga ustrezno analogno obdela, izvede digitalno analizo signala s pomočjo hitre Fourierjeve transformacije (FFT) in prikaže rezultat v obliki grafičnega prikaza na standardnem 16x2 znakovnem LCD zaslonu. Sistem je sestavljen iz treh ključnih sklopov: vhodnega analognega vezja, digitalnega mikrokrmilniškega dela in prikazovalnika. Analogni del vključuje mikrofon ter aktivni pasovno-propustni filter s teoretičnim ojačanjem 100 in prepustnim območjem med 16 Hz in 16 kHz. V praksi se je pri meritvah izkazalo, da ojačanje znaša približno 70, vendar to ne vpliva bistveno na delovanje, saj je signal še vedno dovolj močan za digitalizacijo. Digitalni del temelji na knjižnicah Audio.h za zajem in FFT obdelavo ter LiquidCrystal.h za prikaz rezultatov. Zaslon prikazuje 32-stolpčni bar-diagram z uporabo definiranih znakov, pri čemer vsak stolpec predstavlja povprečje dveh frekvenčnih binov iz 256-točkovne FFT. Prikaz zajema frekvence do približno 11 kHz, kar je bilo izbrano kot praktičen kompromis med ločljivostjo, preglednostjo in učinkovitostjo prikaza. Delovanje sistema je bilo preizkušeno z generiranjem sinusnih tonov iz mobilnega telefona. Frekvence so bile ustrezno zaznane in prikazane, kar potrjuje funkcionalnost celotnega sistema. Nadalje je bilo potrjeno, da je sistem sposoben mehko realnočasnega delovanja, saj se prikaz osvežuje približno 28-krat na sekundo (glede na LED indikator), kar je dovolj hitro za tekoče sledenje spremembam v zvoku. Projekt uspešno združuje analogno elektroniko in digitalno obdelavo signalov ter predstavlja praktičen in dostopen primer DSP-sistema za zvočno analizo. Sistem je bil razvit z uporabo enostavnih in komercialno dostopnih komponent, zato je primeren tudi za nadaljnji razvoj, nadgradnjo ali uporabo v izobraževalne namene. Med možnimi izboljšavami so povečanje prikaznega območja, uporaba večjega zaslona, izboljšano napajanje, brezžični prenos podatkov ali vizualizacija s pomočjo LED trakov.

Cited by 0Open access

AUDIO PROCESSING ANALYZER

Sana Rizwan rizwan · 2012

The project emphasizes simulation of various DSP effects using elementary phenomenon of audio processing, and by manipulating audio using various filters in order to enhance the quality. There are many commercially available systems, which provide facilities such as channel equalizers, karaoke systems, and a few audio processors based on Digital Signal Processing. Software systems are also available which provide a fairly good and cost effective solution to audio enhancement. Yet they are limited due to resources issues and thus make a trade-off between performance and quality. The project at first studies and analyses proceeds as study and analysis of audio processing phenomena and various effects involved in it. In the second phase algorithms have been developed for these phenomena and their simulation in MATLAB.

Cited by 0

Audio frequency analyzer

Lee, Liang De · DR-NTU (Nanyang Technological University) · 2015

The most primitive way of looking at waveforms is in its time domain – looking at how the signal changes in amplitude as time passes. This is normally observed through an oscilloscope, and it is quite natural for us to look at waveforms through an oscilloscope display. However, signals can be displayed in other ways such as in the frequency domain. Jean Baptiste Fourier, a French mathematician and physicist, started to observe how signals are seen in the frequency domain, where signals are viewed as a function of their frequency rather than time. He discovered that any waveform seen in the time domain, there is an equivalent representation in the frequency domain. Meaning that, any signal is made up of many different frequencies, i.e. square waveform are made up of multiple sine waveforms. By observing signals in the frequency domain with a spectrum analyzer enables us to analyze the harmonic and spurious content of a signal. Also the width of signals when modulated has been applied is important for developing Radio Frequency (RF) signal sources, and especially any form of transmitter including those in cellular, Wi-Fi, and other radio or wireless application. In this project, I will be focusing on creating a device capable of sampling audio frequencies in the human audible range.

Cited by 0

A monolithic audio spectrum analyzer

L.T. Lin, Hsin-Fu Tseng, Douglas Cox et al. · IEEE Transactions on Acoustics Speech and Signal Processing · 1983

A monolithic audio spectrum analyzer with 16 channels of bandpass filters, half-wave rectifiers, and postfiltering has been fabricated with double-poly NMOS technology. The chip was designed using switched-capacitor filter techniques. It performs a total of 84 poles of filtering and contains 100 operational amplifiers in an area of 225 × 280 mils. A system dynamic range of better than 43 dB, linearity of better than 1 percent, and center frequency accuracy of better than 1 percent have been achieved.

Cited by 7

Software Analyzer for Audio Effects

Frenštátský, Petr · Brno University of Technology Digital Library (Brno University of Technology) · 2014

Ke zjišťování kvalitivních parametrů zvukových zařízení je stále více využíván stolní počítač, ve kterém díky rozvoji číslicového zpracování signálů je umožněna implementace analýz. Mezi ně patří zjišťování kmitočtové a převodní charakteristiky, parametrů zkreslení (THD, THD+N, WHD, SINAD), míry přeslechů a odstupu signálu od šumu. V práci je vytvořena aplikace, která umužňuje zjišťovat kvalitativní parametry hardwarových zvukových zařízení, které jsou propojeny se zvukovou kartou. Ta využívá pro efektivní komunikaci se stolním počítačem technologii ASIO. Aplikace také umožňuje měřit parametry zvukových zařízení, které jsou implementovány pomocí technologie VST. Aplikace je vyvíjena v jazyce C++. Implementace vychází doporučení AES17.

Cited by 0

DV Analyzer Sample: DV Audio Errors

Bulletin of Miscellaneous Information (Royal Gardens Kew)

File Background This file is digitized video from an NTSC miniDV tape from The Archives of Traditional Music at Indiana University. This particular file is a severe example of audio errors in a DV ingest, rendering the digitized file mostly unusable. The source tape was recorded at LP (long-play) mode and in 4 track audio mode. By recording in LP the amount of data stored within a fixed surface area of the tape is greatly reduced, thus increasing the likelihood of errors. The tape also records in 4 track, 32kHz, 12 bit audio (as opposed to 2 track, 48kHz, 16 bit audio), thus each audio track is stored with half as much data, again increasing the likelihood for error. Given that the 3rd and 4th audio tracks of this scene were not utilized, there was no advantage to record in 4 track, 32kHz, 12 bit. Reading Audio Datafrom a DV Tape and Associated Error Detection When a DV tape plays in a deck, each pass of the head across the tape reads 9 blocks of audio data. The data transmitted from each head pass is referred to as a DIF sequence. In NTSC 10 tape passes (or DIF sequences) are needed to complete a full frame of DV (or 12 in PAL). Thus each NTSC frame contains a total of 90 audio blocks (or 108 in PAL). Each audio block contains parity data for error correction that can be used if the primary audio data is damaged or could not be read. In the case where the primary audio data and the parity data are both excessively damaged or not read properly, the deck may not have sufficient resources to perform error correction on that block of audio in the very short amount of time allowed to perform this work, thereby causing a drop out. When the deck fails to correct the audio it uses a reserved audio sample value 0x8000 (or in 12 bit mode 0x808000) to state that the audio sample is erroneous. These audio error codes may be parsed from a dv file in order to automate the detection of audio errors (without non-stop listening). Conclusions Since this recording is configured to record 4 track audio but only contains 2 tracks, only DIF sequences 0, 1, 2, 3, and 4 contain audio data (DIF sequences 5-9 are empty although still often error-filled). This list reports on the total number of reported audio blocks are error-filled for the entire duration of the file (see this document for a more detail technical synopsis of the error detection performed): 648 Dseq=0 54 Dseq=1 702 Dseq=2 45 Dseq=3 684 Dseq=4 Interestingly, in this sample the great majority of the errors are on even-numbered tape passes, indicating that the problems are more likely generated by a bad recording or playback head than by tape damage. This XML document reports on each frame that contains errors, providing the absolute time, timecode of the source tape, date and time of the recording, and information on the errors. For example frame number 6 has an error statement: “CH1: 27 audio errors ( 9 Dseq=0, 9 Dseq=2, 9 Dseq=4)”. The “Dseq” refers to the DIF sequence, or the number of the head pass along the tape to read a frame (in NTSC 10 head passes are used for a single frame). Since there are 9 audio blocks per pass (or DIF sequence), the statement “9 Dseq=0” means that all 9 audio blocks failed to be read during the first head pass of that frame, thus 10% of that frames audio is missing. This happens again during head pass number 2 and 4 for a total of 30% audio loss. However since this is a 4 track recording, DIF sequences (or head passes) 0-4 contain the audio for tracks 1 and 2 while sequences 5-9 contain the audio for tracks 3 and 4 (unused). Thus audio tracks 1 and 2 of frame 6 are missing 60% of the audio. For more information: http://www.avpreserve.com

Cited by 0

Aud-Sur: An Audio Analyzer Assistant for Audio Surveillance Applications

Phat Lam, Lam Pham, Dat Tran et al. · ArXiv.org · 2025

In this paper, we present an audio analyzer assistant tool designed for a wide range of audio-based surveillance applications (This work is a part of our DEFAME FAKES and EUCINF projects). The proposed tool, refered to as Aud-Sur, comprises two main phases Audio Analysis and Audio Retrieval, respectively. In the first phase, multiple open-source audio models are leveraged to extract information from input audio recording uploaded by a user. In the second phase, users interact with the Aud-Sur tool via a natural question-and-answer manner, powered by a large language model (LLM), to retrieve the information extracted from the processed audio file. The Aud-Sur tool was deployed using Docker on a microservices-based architecture design. By leveraging open-source audio models for information extraction, LLM for audio information retrieval, and a microservices-based deployment approach, the proposed Aud-Sur tool offers a highly extensible and adaptable framework that can integrate more audio tasks, and be widely shared within the audio community for further development.

Cited by 0Open access

Input analog part of spectral audio analyzer

Taťák, Martin · Brno University of Technology Digital Library (Brno University of Technology) · 2014

Tato práce pojednává o možnostech realizace vstupní části spektrálního audio analyzátoru. Rozebírá jednotlivé teoretické možnosti a nejvhodnější je pak zpracována ve formě prototypové desky, jejíž návrh tato práce obsahuje.

Cited by 0

Electronic analyzer of audio-magnetotelluric signals, An

Andre B. Lebel · Digital Collections of Colorado (Colorado State University) · 1970

An electronic instrument which measures magnetotelluric signals in the frequency range between 1.78 Hz and 715 Hz is described.Based on a design proposed by Berdichevskii (1 9 6 9 ) t the instrument computes the autocorrelation of a horizontal component of the magnetic field and the cross correlation of this magnetic component with an orthogonal, horizontal component of the electric field.Field tests of the analyzer, near Livermore, Colorado, over Precambrian granite, and at the Lowry Bombing Range, Colorado, over a thick sedimentary section, are described.The apparent resistivities determined by the AMT measurements at frequencies above 152 Hz agree within an order of magnitude with DC resistivity measurements made previously at these sites.However* the apparent resistivities for frequencies below 152 Hz are rejected because of the observed incoherence of the signals.The design objectives of the analyzer were met.However, if the full potential of the instrument is to be realized, artificially generated source fields should be used.

Cited by 0Open access

Performance evaluation of an audio-frequency power analyzer

Gerard N. Stenbakken, Thomas L. Nelson, B.C. Waltrip · IEEE Power Engineering Society General Meeting, 2005 · 2005

In response to industry requests to have calibration services for distorted power instrumentation, NIST is developing a new sampling system to provide this service. This development effort is aimed at providing calibration of instruments that makes measurements in accordance with the IEEE trial use standard 1459-2000 on power measurements in distorted and unbalanced conditions. The system makes use of several NIST developed instruments and sensors.

Cited by 0