Adaptive Digital Audio Effects (A-Dafx): a New Class of Sound Transformations Vincent Verfaille, Udo Zölzer, Daniel Arfib
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Metaflanger Table of Contents
MetaFlanger Table of Contents Chapter 1 Introduction 2 Chapter 2 Quick Start 3 Flanger effects 5 Chorus effects 5 Producing a phaser effect 5 Chapter 3 More About Flanging 7 Chapter 4 Controls & Displa ys 11 Section 1: Mix, Feedback and Filter controls 11 Section 2: Delay, Rate and Depth controls 14 Section 3: Waveform, Modulation Display and Stereo controls 16 Section 4: Output level 18 Chapter 5 Frequently Asked Questions 19 Chapter 6 Block Diagram 20 Chapter 7.........................................................Tempo Sync in V5.0.............22 MetaFlanger Manual 1 Chapter 1 - Introduction Thanks for buying Waves processors. MetaFlanger is an audio plug-in that can be used to produce a variety of classic tape flanging, vintage phas- er emulation, chorusing, and some unexpected effects. It can emulate traditional analog flangers,fill out a simple sound, create intricate harmonic textures and even generate small rough reverbs and effects. The following pages explain how to use MetaFlanger. MetaFlanger’s Graphic Interface 2 MetaFlanger Manual Chapter 2 - Quick Start For mixing, you can use MetaFlanger as a direct insert and control the amount of flanging with the Mix control. Some applications also offer sends and returns; either way works quite well. 1 When you insert MetaFlanger, it will open with the default settings (click on the Reset button to reload these!). These settings produce a basic classic flanging effect that’s easily tweaked. 2 Preview your audio signal by clicking the Preview button. If you are using a real-time system (such as TDM, VST, or MAS), press ‘play’. You’ll hear the flanged signal. -
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. -
21065L Audio Tutorial
a Using The Low-Cost, High Performance ADSP-21065L Digital Signal Processor For Digital Audio Applications Revision 1.0 - 12/4/98 dB +12 0 -12 Left Right Left EQ Right EQ Pan L R L R L R L R L R L R L R L R 1 2 3 4 5 6 7 8 Mic High Line L R Mid Play Back Bass CNTR 0 0 3 4 Input Gain P F R Master Vol. 1 2 3 4 5 6 7 8 Authors: John Tomarakos Dan Ledger Analog Devices DSP Applications 1 Using The Low Cost, High Performance ADSP-21065L Digital Signal Processor For Digital Audio Applications Dan Ledger and John Tomarakos DSP Applications Group, Analog Devices, Norwood, MA 02062, USA This document examines desirable DSP features to consider for implementation of real time audio applications, and also offers programming techniques to create DSP algorithms found in today's professional and consumer audio equipment. Part One will begin with a discussion of important audio processor-specific characteristics such as speed, cost, data word length, floating-point vs. fixed-point arithmetic, double-precision vs. single-precision data, I/O capabilities, and dynamic range/SNR capabilities. Comparisions between DSP's and audio decoders that are targeted for consumer/professional audio applications will be shown. Part Two will cover example algorithmic building blocks that can be used to implement many DSP audio algorithms using the ADSP-21065L including: Basic audio signal manipulation, filtering/digital parametric equalization, digital audio effects and sound synthesis techniques. TABLE OF CONTENTS 0. INTRODUCTION ................................................................................................................................................................4 1. -
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]. -
Recording and Amplifying of the Accordion in Practice of Other Accordion Players, and Two Recordings: D
CA1004 Degree Project, Master, Classical Music, 30 credits 2019 Degree of Master in Music Department of Classical music Supervisor: Erik Lanninger Examiner: Jan-Olof Gullö Milan Řehák Recording and amplifying of the accordion What is the best way to capture the sound of the acoustic accordion? SOUNDING PART.zip - Sounding part of the thesis: D. Scarlatti - Sonata D minor K 141, V. Trojan - The Collapsed Cathedral SOUND SAMPLES.zip – Sound samples Declaration I declare that this thesis has been solely the result of my own work. Milan Řehák 2 Abstract In this thesis I discuss, analyse and intend to answer the question: What is the best way to capture the sound of the acoustic accordion? It was my desire to explore this theme that led me to this research, and I believe that this question is important to many other accordionists as well. From the very beginning, I wanted the thesis to be not only an academic material but also that it can be used as an instruction manual, which could serve accordionists and others who are interested in this subject, to delve deeper into it, understand it and hopefully get answers to their questions about this subject. The thesis contains five main chapters: Amplifying of the accordion at live events, Processing of the accordion sound, Recording of the accordion in a studio - the specifics of recording of the accordion, Specific recording solutions and Examples of recording and amplifying of the accordion in practice of other accordion players, and two recordings: D. Scarlatti - Sonata D minor K 141, V. Trojan - The Collasped Cathedral. -
User's Manual
USER’S MANUAL G-Force GUITAR EFFECTS PROCESSOR IMPORTANT SAFETY INSTRUCTIONS The lightning flash with an arrowhead symbol The exclamation point within an equilateral triangle within an equilateral triangle, is intended to alert is intended to alert the user to the presence of the user to the presence of uninsulated "dan- important operating and maintenance (servicing) gerous voltage" within the product's enclosure that may instructions in the literature accompanying the product. be of sufficient magnitude to constitute a risk of electric shock to persons. 1 Read these instructions. Warning! 2 Keep these instructions. • To reduce the risk of fire or electrical shock, do not 3 Heed all warnings. expose this equipment to dripping or splashing and 4 Follow all instructions. ensure that no objects filled with liquids, such as vases, 5 Do not use this apparatus near water. are placed on the equipment. 6 Clean only with dry cloth. • This apparatus must be earthed. 7 Do not block any ventilation openings. Install in • Use a three wire grounding type line cord like the one accordance with the manufacturer's instructions. supplied with the product. 8 Do not install near any heat sources such • Be advised that different operating voltages require the as radiators, heat registers, stoves, or other use of different types of line cord and attachment plugs. apparatus (including amplifiers) that produce heat. • Check the voltage in your area and use the 9 Do not defeat the safety purpose of the polarized correct type. See table below: or grounding-type plug. A polarized plug has two blades with one wider than the other. -
In the Studio: the Role of Recording Techniques in Rock Music (2006)
21 In the Studio: The Role of Recording Techniques in Rock Music (2006) John Covach I want this record to be perfect. Meticulously perfect. Steely Dan-perfect. -Dave Grohl, commencing work on the Foo Fighters 2002 record One by One When we speak of popular music, we should speak not of songs but rather; of recordings, which are created in the studio by musicians, engineers and producers who aim not only to capture good performances, but more, to create aesthetic objects. (Zak 200 I, xvi-xvii) In this "interlude" Jon Covach, Professor of Music at the Eastman School of Music, provides a clear introduction to the basic elements of recorded sound: ambience, which includes reverb and echo; equalization; and stereo placement He also describes a particularly useful means of visualizing and analyzing recordings. The student might begin by becoming sensitive to the three dimensions of height (frequency range), width (stereo placement) and depth (ambience), and from there go on to con sider other special effects. One way to analyze the music, then, is to work backward from the final product, to listen carefully and imagine how it was created by the engineer and producer. To illustrate this process, Covach provides analyses .of two songs created by famous producers in different eras: Steely Dan's "Josie" and Phil Spector's "Da Doo Ron Ron:' Records, tapes, and CDs are central to the history of rock music, and since the mid 1990s, digital downloading and file sharing have also become significant factors in how music gets from the artists to listeners. Live performance is also important, and some groups-such as the Grateful Dead, the Allman Brothers Band, and more recently Phish and Widespread Panic-have been more oriented toward performances that change from night to night than with authoritative versions of tunes that are produced in a recording studio. -
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.