Bridge Audio to Control, Filtered Afc This Module Converts an Audio Signal Into a Control Signal. Before Doing T
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UC Riverside UC Riverside Electronic Theses and Dissertations
UC Riverside UC Riverside Electronic Theses and Dissertations Title Sonic Retro-Futures: Musical Nostalgia as Revolution in Post-1960s American Literature, Film and Technoculture Permalink https://escholarship.org/uc/item/65f2825x Author Young, Mark Thomas Publication Date 2015 Peer reviewed|Thesis/dissertation eScholarship.org Powered by the California Digital Library University of California UNIVERSITY OF CALIFORNIA RIVERSIDE Sonic Retro-Futures: Musical Nostalgia as Revolution in Post-1960s American Literature, Film and Technoculture A Dissertation submitted in partial satisfaction of the requirements for the degree of Doctor of Philosophy in English by Mark Thomas Young June 2015 Dissertation Committee: Dr. Sherryl Vint, Chairperson Dr. Steven Gould Axelrod Dr. Tom Lutz Copyright by Mark Thomas Young 2015 The Dissertation of Mark Thomas Young is approved: Committee Chairperson University of California, Riverside ACKNOWLEDGEMENTS As there are many midwives to an “individual” success, I’d like to thank the various mentors, colleagues, organizations, friends, and family members who have supported me through the stages of conception, drafting, revision, and completion of this project. Perhaps the most important influences on my early thinking about this topic came from Paweł Frelik and Larry McCaffery, with whom I shared a rousing desert hike in the foothills of Borrego Springs. After an evening of food, drink, and lively exchange, I had the long-overdue epiphany to channel my training in musical performance more directly into my academic pursuits. The early support, friendship, and collegiality of these two had a tremendously positive effect on the arc of my scholarship; knowing they believed in the project helped me pencil its first sketchy contours—and ultimately see it through to the end. -
Pitch Shifting with the Commercially Available Eventide Eclipse: Intended and Unintended Changes to the Speech Signal
JSLHR Research Note Pitch Shifting With the Commercially Available Eventide Eclipse: Intended and Unintended Changes to the Speech Signal Elizabeth S. Heller Murray,a Ashling A. Lupiani,a Katharine R. Kolin,a Roxanne K. Segina,a and Cara E. Steppa,b,c Purpose: This study details the intended and unintended 5.9% and 21.7% less than expected, based on the portion consequences of pitch shifting with the commercially of shift selected for measurement. The delay between input available Eventide Eclipse. and output signals was an average of 11.1 ms. Trials Method: Ten vocally healthy participants (M = 22.0 years; shifted +100 cents had a longer delay than trials shifted 6 cisgender females, 4 cisgender males) produced a sustained −100 or 0 cents. The first 2 formants (F1, F2) shifted in the /ɑ/, creating an input signal. This input signal was processed direction of the pitch shift, with F1 shifting 6.5% and F2 in near real time by the Eventide Eclipse to create an output shifting 6.0%. signal that was either not shifted (0 cents), shifted +100 cents, Conclusions: The Eventide Eclipse is an accurate pitch- or shifted −100 cents. Shifts occurred either throughout the shifting hardware that can be used to explore voice and entire vocalization or for a 200-ms period after vocal onset. vocal motor control. The pitch-shifting algorithm shifts all Results: Input signals were compared to output signals to frequencies, resulting in a subsequent change in F1 and F2 examine potential changes. Average pitch-shift magnitudes during pitch-shifted trials. Researchers using this device were within 1 cent of the intended pitch shift. -
3-D Audio Using Loudspeakers
3-D Audio Using Loudspeakers William G. Gardner B. S., Computer Science and Engineering, Massachusetts Institute of Technology, 1982 M. S., Media Arts and Sciences, Massachusetts Institute of Technology, 1992 Submitted to the Program in Media Arts and Sciences, School of Architecture and Planning in Partial Fulfillment of the Requirements for the Degree of Doctor of Philosophy at the Massachusetts Institute of Technology September, 1997 © Massachusetts Institute of Technology, 1997. All Rights Reserved. Author Program in Media Arts and Sciences August 8, 1997 Certified by Barry L. Vercoe Professor of Media Arts and Sciences Massachusetts Institute of Technology Accepted by Stephen A. Benton Chair, Departmental Committee on Graduate Students Program in Media Arts and Sciences Massachusetts Institute of Technology 3-D Audio Using Loudspeakers William G. Gardner Submitted to the Program in Media Arts and Sciences, School of Architecture and Planning on August 8, 1997, in Partial Fulfillment of the Requirements for the Degree of Doctor of Philosophy. Abstract 3-D audio systems, which can surround a listener with sounds at arbitrary locations, are an important part of immersive interfaces. A new approach is presented for implementing 3-D audio using a pair of conventional loudspeakers. The new idea is to use the tracked position of the listener’s head to optimize the acoustical presentation, and thus produce a much more realistic illusion over a larger listening area than existing loudspeaker 3-D audio systems. By using a remote head tracker, for instance based on computer vision, an immersive audio environment can be created without donning headphones or other equipment. -
Reference Manual
1 6 - V O I C E R E A L A N A L O G S Y N T H E S I Z E R REFERENCE MANUAL February 2001 Your shipping carton should contain the following items: 1. Andromeda A6 synthesizer 2. AC power cable 3. Warranty Registration card 4. Reference Manual 5. A list of the Preset and User Bank Mixes and Programs If anything is missing, please contact your dealer or Alesis immediately. NOTE: Completing and returning the Warranty Registration card is important. Alesis contact information: Alesis Studio Electronics, Inc. 1633 26th Street Santa Monica, CA 90404 USA Telephone: 800-5-ALESIS (800-525-3747) E-Mail: [email protected] Website: http://www.alesis.com Alesis Andromeda A6TM Reference Manual Revision 1.0 by Dave Bertovic © Copyright 2001, Alesis Studio Electronics, Inc. All rights reserved. Reproduction in whole or in part is prohibited. “A6”, “QCard” and “FreeLoader” are trademarks of Alesis Studio Electronics, Inc. A6 REFERENCE MANUAL 1 2 A6 REFERENCE MANUAL Contents CONTENTS Important Safety Instructions ....................................................................................7 Instructions to the User (FCC Notice) ............................................................................................11 CE Declaration of Conformity .........................................................................................................13 Introduction ................................................................................................................ 15 How to Use this Manual....................................................................................................................16 -
Designing Empowering Vocal and Tangible Interaction
Designing Empowering Vocal and Tangible Interaction Anders-Petter Andersson Birgitta Cappelen Institute of Design Institute of Design AHO, Oslo AHO, Oslo Norway Norway [email protected] [email protected] ABSTRACT on observations in the research project RHYME for the last 2 Our voice and body are important parts of our self-experience, years and on work with families with children and adults with and our communication and relational possibilities. They severe disabilities prior to that. gradually become more important for Interaction Design due to Our approach is multidisciplinary and based on earlier studies increased development of tangible interaction and mobile of voice in resource-oriented Music and Health research and communication. In this paper we present and discuss our work the work on voice by music therapists. Further, more studies with voice and tangible interaction in our ongoing research and design methods in the fields of Tangible Interaction in project RHYME. The goal is to improve health for families, Interaction Design [10], voice recognition and generative sound adults and children with disabilities through use of synthesis in Computer Music [22, 31], and Interactive Music collaborative, musical, tangible media. We build on the use of [1] for interacting persons with layman expertise in everyday voice in Music Therapy and on a humanistic health approach. situations. Our challenge is to design vocal and tangible interactive media Our results point toward empowered participants, who that through use reduce isolation and passivity and increase interact with the vocal and tangible interactive designs [5]. empowerment for the users. We use sound recognition, Observations and interviews show increased communication generative sound synthesis, vibrations and cross-media abilities, social interaction and improved health [29]. -
Improved Audio Filtering Using Extended High Pass Filters
International Journal of Engineering Research & Technology (IJERT) ISSN: 2278-0181 Vol. 2 Issue 6, June - 2013 Improved Audio Filtering Using Extended High Pass Filters 1 2 Radhika Bhagat Ramandeep Kaur Department of Computer Science & Engineering, Guru Nanak Dev University Amritsar, Punjab, India Abstract—Noise Filtering in audio signals has always been a challenge since the noise is spread across a large bandwidth and overlaps the spectral range of the audio signal being recovered. In audio systems all these can be kept below the audible level but ambient noise may not be avoided even if the audio system is designed according to the best practices. So, there have been several audio filtering techniques such as spectral subtraction, Dolby noise reduction, use of low-pass and high pass filters, FIR and IIR filtering, etc. Noise filtering improvements were assessed for both noise reduction and signal degradation effects by different signal to noise ratio computations. Keywords: Audio Filtering, Low Pass Filters, High Pass Filters, FIR Filters, IIR Filters. I. INTRODUCTION With the development of communication technology, voice communication has become a major communication medium for people to transmit information more convenient. However, the widespread nature of noise makes the voice communication quality has declined. Therefore, to reduceIJERTIJERT the noise on the performance of voice communications, improve the quality of voice communications [1], voice denoising for technology has become a hot research topic. In this paper we will discuss various audio filtering techniques that will help in overcoming noise related issues. Audio Filter: - Is a frequency dependent amplifier circuit, working in the audio frequency range,0Hz- beyond 20 KHz. -
SSB and the Phasing Exciter Hints and Kinks for Best Performance and Being Nice to One’S Neighbor’S
SSB and the Phasing Exciter Hints and Kinks for Best Performance and Being Nice to One’s Neighbor’s Nick Tusa – K5EF How is SSB Generated? Brute Force v. Elegance A Tug or War in the 1950s One way is to design a steep-sided Steep-sided filters was expensive in the bandpass filter that passes one 1940-early 50s. Required carefully sideband and hugely attenuates the selected and ground individual crystals other. or expert machining of temperature- stable materials for a mechanical filter. Or Or We can use a mathematical analog of phase relationships to cancel the Audio filters having linear phase shift unwanted sideband and reinforce made with inexpensive Rs and Cs. the desired sideband. In Amateur Circles, Phasing Led the Way Don Norgaard really got the Amateur Ball rolling with his SSB, Jr. published in GE HAM News. (Refer to George W1LSB’s SSB discussion at last year’s W9DYV Event and CE website). Companies such as CE, Lakeshore Industries, Eldico, Johnson, Hallicrafters, Heathkit, B&W and others produced exciters using the Phasing Principal. By the late 1950s, filter technology improved and costs dropped, pushing Phasing aside…at least until the Software Defined Radio came along… Basic Phasing SSB Exciter Audio Phase Shift Linearity is Key to SB Suppression The AF Phase Shifter must maintain 90° differential across 300-3500Hz audio band. Small deviations result in degraded SB suppression i.e., a 2˚ difference = 35db suppression. Typical CE 10A/20A based on the Norgaard design = 40db if perfect! Central Electronics PS-1 Network Based on Norgaard’s SSB, Jr design. -
Pitch-Shifting Algorithm Design and Applications in Music
DEGREE PROJECT IN ELECTRICAL ENGINEERING, SECOND CYCLE, 30 CREDITS STOCKHOLM, SWEDEN 2019 Pitch-shifting algorithm design and applications in music THÉO ROYER KTH ROYAL INSTITUTE OF TECHNOLOGY SCHOOL OF ELECTRICAL ENGINEERING AND COMPUTER SCIENCE ii Abstract Pitch-shifting lowers or increases the pitch of an audio recording. This technique has been used in recording studios since the 1960s, many Beatles tracks being produced using analog pitch-shifting effects. With the advent of the first digital pitch-shifting hardware in the 1970s, this technique became essential in music production. Nowa- days, it is massively used in popular music for pitch correction or other creative pur- poses. With the improvement of mixing and mastering processes, the recent focus in the audio industry has been placed on the high quality of pitch-shifting tools. As a consequence, current state-of-the-art literature algorithms are often outperformed by the best commercial algorithms. Unfortunately, these commercial algorithms are ”black boxes” which are very complicated to reverse engineer. In this master thesis, state-of-the-art pitch-shifting techniques found in the liter- ature are evaluated, attaching great importance to audio quality on musical signals. Time domain and frequency domain methods are studied and tested on a wide range of audio signals. Two offline implementations of the most promising algorithms are proposed with novel features. Pitch Synchronous Overlap and Add (PSOLA), a sim- ple time domain algorithm, is used to create pitch-shifting, formant-shifting, pitch correction and chorus effects on voice and monophonic signals. Phase vocoder, a more complex frequency domain algorithm, is combined with high quality spec- tral envelope estimation and harmonic-percussive separation to design a polyvalent pitch-shifting and formant-shifting algorithm. -
Effects List
Effects List Multi-Effects Parameters (MFX1–3, MFX) The multi-effects feature 78 different kinds of effects. Some of the effects consist of two or more different effects connected in series. 67 p. 8 FILTER (10 types) OD->Flg (OVERDRIVE"FLANGER) 68 OD->Dly (OVERDRIVE DELAY) p. 8 01 Equalizer (EQUALIZER) p. 1 " 69 Dist->Cho (DISTORTION CHORUS) p. 8 02 Spectrum (SPECTRUM) p. 2 " 70 Dist->Flg (DISTORTION FLANGER) p. 8 03 Isolator (ISOLATOR) p. 2 " 71 p. 8 04 Low Boost (LOW BOOST) p. 2 Dist->Dly (DISTORTION"DELAY) 05 Super Flt (SUPER FILTER) p. 2 72 Eh->Cho (ENHANCER"CHORUS) p. 8 06 Step Flt (STEP FILTER) p. 2 73 Eh->Flg (ENHANCER"FLANGER) p. 8 07 Enhancer (ENHANCER) p. 2 74 Eh->Dly (ENHANCER"DELAY) p. 8 08 AutoWah (AUTO WAH) p. 2 75 Cho->Dly (CHORUS"DELAY) p. 9 09 Humanizer (HUMANIZER) p. 2 76 Flg->Dly (FLANGER"DELAY) p. 9 10 Sp Sim (SPEAKER SIMULATOR) p. 2 77 Cho->Flg (CHORUS"FLANGER) p. 9 MODULATION (12 types) PIANO (1 type) 11 Phaser (PHASER) p. 3 78 SymReso (SYMPATHETIC RESONANCE) p. 9 12 Step Ph (STEP PHASER) p. 3 13 MltPhaser (MULTI STAGE PHASER) p. 3 14 InfPhaser (INFINITE PHASER) p. 3 15 Ring Mod (RING MODULATOR) p. 3 About Note 16 Step Ring (STEP RING MODULATOR) p. 3 17 Tremolo (TREMOLO) p. 3 Some effect settings (such as Rate or Delay Time) can be specified in 18 Auto Pan (AUTO PAN) p. 3 terms of a note value. The note value for the current setting is 19 Step Pan (STEP PAN) p. -
MS-100BT Effects List
Effect Types and Parameters © 2012 ZOOM CORPORATION Copying or reproduction of this document in whole or in part without permission is prohibited. Effect Types and Parameters Parameter Parameter range Effect type Effect explanation Flanger This is a jet sound like an ADA Flanger. Knob1 Knob2 Knob3 Depth 0–100 Rate 0–50 Reso -10–10 Page01 Sets the depth of the modulation. Sets the speed of the modulation. Adjusts the intensity of the modulation resonance. PreD 0–50 Mix 0–100 Level 0–150 Page02 Adjusts the amount of effected sound Sets pre-delay time of effect sound. Adjusts the output level. that is mixed with the original sound. Effect screen Parameter explanation Tempo synchronization possible icon Effect Types and Parameters [DYN/FLTR] Comp This compressor in the style of the MXR Dyna Comp. Knob1 Knob2 Knob3 Sense 0–10 Tone 0–10 Level 0–150 Page01 Adjusts the compressor sensitivity. Adjusts the tone. Adjusts the output level. ATTCK Slow, Fast Page02 Sets compressor attack speed to Fast or Slow. RackComp This compressor allows more detailed adjustment than Comp. Knob1 Knob2 Knob3 THRSH 0–50 Ratio 1–10 Level 0–150 Page01 Sets the level that activates the Adjusts the compression ratio. Adjusts the output level. compressor. ATTCK 1–10 Page02 Adjusts the compressor attack rate. M Comp This compressor provides a more natural sound. Knob1 Knob2 Knob3 THRSH 0–50 Ratio 1–10 Level 0–150 Page01 Sets the level that activates the Adjusts the compression ratio. Adjusts the output level. compressor. ATTCK 1–10 Page02 Adjusts the compressor attack rate. -
Timbretron:Awavenet(Cyclegan(Cqt(Audio))) Pipeline for Musical Timbre Transfer
Published as a conference paper at ICLR 2019 TIMBRETRON:AWAVENET(CYCLEGAN(CQT(AUDIO))) PIPELINE FOR MUSICAL TIMBRE TRANSFER Sicong Huang1;2, Qiyang Li1;2, Cem Anil1;2, Xuchan Bao1;2, Sageev Oore2;3, Roger B. Grosse1;2 University of Toronto1, Vector Institute2, Dalhousie University3 ABSTRACT In this work, we address the problem of musical timbre transfer, where the goal is to manipulate the timbre of a sound sample from one instrument to match another instrument while preserving other musical content, such as pitch, rhythm, and loudness. In principle, one could apply image-based style transfer techniques to a time-frequency representation of an audio signal, but this depends on having a representation that allows independent manipulation of timbre as well as high- quality waveform generation. We introduce TimbreTron, a method for musical timbre transfer which applies “image” domain style transfer to a time-frequency representation of the audio signal, and then produces a high-quality waveform using a conditional WaveNet synthesizer. We show that the Constant Q Transform (CQT) representation is particularly well-suited to convolutional architectures due to its approximate pitch equivariance. Based on human perceptual evaluations, we confirmed that TimbreTron recognizably transferred the timbre while otherwise preserving the musical content, for both monophonic and polyphonic samples. We made an accompanying demo video 1 which we strongly encourage you to watch before reading the paper. 1 INTRODUCTION Timbre is a perceptual characteristic that distinguishes one musical instrument from another playing the same note with the same intensity and duration. Modeling timbre is very hard, and it has been referred to as “the psychoacoustician’s multidimensional waste-basket category for everything that cannot be labeled pitch or loudness”2. -
Audio Plug-Ins Guide Version 9.0 Legal Notices This Guide Is Copyrighted ©2010 by Avid Technology, Inc., (Hereafter “Avid”), with All Rights Reserved
Audio Plug-Ins Guide Version 9.0 Legal Notices This guide is copyrighted ©2010 by Avid Technology, Inc., (hereafter “Avid”), with all rights reserved. Under copyright laws, this guide may not be duplicated in whole or in part without the written consent of Avid. 003, 96 I/O, 96i I/O, 192 Digital I/O, 192 I/O, 888|24 I/O, 882|20 I/O, 1622 I/O, 24-Bit ADAT Bridge I/O, AudioSuite, Avid, Avid DNA, Avid Mojo, Avid Unity, Avid Unity ISIS, Avid Xpress, AVoption, Axiom, Beat Detective, Bomb Factory, Bruno, C|24, Command|8, Control|24, D-Command, D-Control, D-Fi, D-fx, D-Show, D-Verb, DAE, Digi 002, DigiBase, DigiDelivery, Digidesign, Digidesign Audio Engine, Digidesign Intelligent Noise Reduction, Digidesign TDM Bus, DigiDrive, DigiRack, DigiTest, DigiTranslator, DINR, DV Toolkit, EditPack, Eleven, EUCON, HD Core, HD Process, Hybrid, Impact, Interplay, LoFi, M-Audio, MachineControl, Maxim, Mbox, MediaComposer, MIDI I/O, MIX, MultiShell, Nitris, OMF, OMF Interchange, PRE, ProControl, Pro Tools M-Powered, Pro Tools, Pro Tools|HD, Pro Tools LE, QuickPunch, Recti-Fi, Reel Tape, Reso, Reverb One, ReVibe, RTAS, Sibelius, Smack!, SoundReplacer, Sound Designer II, Strike, Structure, SYNC HD, SYNC I/O, Synchronic, TL Aggro, TL AutoPan, TL Drum Rehab, TL Everyphase, TL Fauxlder, TL In Tune, TL MasterMeter, TL Metro, TL Space, TL Utilities, Transfuser, Trillium Lane Labs, Vari-Fi, Velvet, X-Form, and XMON are trademarks or registered trademarks of Avid Technology, Inc. Xpand! is Registered in the U.S. Patent and Trademark Office. All other trademarks are the property of their respective owners.