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Electronic Theses, Treatises and Dissertations The Graduate School
Recording the Classical Guitar: A Documentation and Sound Analysis of Great Classical Guitar Recordings with a GPhiulipi Edugeen ef oLorga nSonic Emulation
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COLLEGE OF MUSIC
RECORDING THE CLASSICAL GUITAR:
A DOCUMENTATION AND SOUND ANALYSIS OF GREAT CLASSICAL GUITAR
RECORDINGS WITH A GUIDE FOR SONIC EMULATION
By
PHILIP EUGENE LOGAN
A Treatise submitted to the College of Music in partial fulfillment of the requirements for the degree of Doctor of Music
2019 Philip Eugene Logan defended this treatise on November 13th, 2019. The members of the supervisory committee were:
Bruce Holzman Professor Directing Treatise
Jane Piper Clendinning University Representative
Corinne Stillwell Committee Member
Brian Gaber Committee Member
The Graduate School has verified and approved the above-named committee members, and certifies that the treatise has been approved in accordance with university requirements.
ii
To Lennox
iii ACKNOWLEDGMENTS
I would like to acknowledge all of those who assisted me in the completion of this research, including those that willingly answered my research questions without hesitation. My thanks go out to Alan Bise, Asgerdur Sigurdardottir, David and Maria Russell, John Taylor, and
Norbert Kraft. I would also like to acknowledge my wonderful family for their incredible support, and for their unsubtle motivational methods.
iv TABLE OF CONTENTS
List of Figures ...... vi Abstract ...... ix
1. INTRODUCTION ...... 1
2. DOCUMENTATION OF RECORDINGS ...... 4
3. AUDIO ANALYSIS ...... 20
4. THE STUDIO ...... 56
5. RECORDING AND PROCESSING ...... 78
6. SONIC EMULATION ...... 105
APPENDIX A. GLOSSARY OF TERMS ...... 140
APPENDIX B. IRB APPROVAL ...... 148
APPENDIX C. RESEARCH INFORMATION SHEET ...... 150
APPENDIX D. DISCOGRAPHY ...... 151
References ...... 152
Biographical Sketch ...... 154
v LIST OF FIGURES
3.1 PAZ Stereo Position Display ...... 20
3.2 Waveform Statistics ...... 21
3.3 Spectral Graph ...... 22
3.4 Bream Stereo Image ...... 24
3.5 Bream Waveform Statistics ...... 25
3.6 Bream Spectrograph ...... 25
3.7 Kraft Stereo Image ...... 27
3.8 Kraft Waveform Statistics ...... 28
3.9 Kraft Spectrograph ...... 28
3.10 Isbin Stereo Image ...... 31
3.11 Isbin Waveform Statistics ...... 31
3.12 Isbin Spectrograph ...... 32
3.13 Russell Spectrograph ...... 34
3.14 Russell Waveform Statistics ...... 35
3.15 Russell Stereo Image ...... 36
3.16 Barrueco Waveform Statistics ...... 37
3.17 Baurreco Stereo Image ...... 38
3.18 Barrueco Spectrograph ...... 38
3.19 Iznaola Spectrograph ...... 40
3.20 Iznaola Stereo Image ...... 41
3.21 Iznaola Waveform Statistics ...... 41
3.22 Iznaola Waveform Compression ...... 42
vi 3.23 Vieaux Spectrograph ...... 43
3.24 Vieaux Waveform Statistics ...... 44
3.25 Vieaux Stereo Image ...... 45
3.26 VIDA Stereo Width ...... 46
3.27 VIDA Waveform Statistics ...... 47
3.28 VIDA Spectrograph ...... 47
3.29 Jara Stereo Image ...... 49
3.30 Jara Waveform Statistics ...... 49
3.31 Jara Spectrograph ...... 50
3.32 Kraft to Vieaux Emulation ...... 51
3.33 Russell to Iznaola Emulation ...... 53
4.1 Sonarworks Measurement ...... 64
6.1 Unaffected Signal ...... 111
6.2 BX_Digital V3 ...... 112
6.3 Izotope Ozone 7 Equalizer ...... 113
6.4 Brainworx Elysia Museq ...... 114
6.5 Brainworx Millenia TCL-2 ...... 115
6.6 Brainworx Millenia TCL-2 J-FET ...... 116
6.7 Kush Omega Tranformer: Model N ...... 117
6.8 Brainworx Black Box HG-2 ...... 118
6.9 Waves J37 Tape Machine ...... 119
6.10 Russell Emulation Pre-manipulated ...... 121
6.11 Russell Emulation Manual Manipulation ...... 122
vii 6.12 EQ Set for Russell Emulation ...... 123
6.13 Manual Manipulation Versus Izotope’s EQ Match ...... 125
6.14 Russell Emulation Result ...... 126
6.15 Ableton EQ with Oversampling Activated ...... 128
6.16 Soundtoys Radiator ...... 128
6.17 BX_Digital V3-1 ...... 129
6.18 BX_Digital V3-2 ...... 129
6.19 BX_Digital V3-3 ...... 130
6.20 BX_Cleansweep Pro ...... 130
6.21 Liquid Sonics’ Seventh Heaven Set to Produce Early Reflections (Return A) ...... 131
6.22 Ableton EQ to Shape Tone of Reverb (Return A) ...... 131
6.23 Seventh Heaven for Producing Late Reflections (Return B) ...... 131
6.24 Waves J37 ...... 132
6.25 Waves Kramer Tape ...... 132
6.26 Bream Emulation Pre-manipulation ...... 134
6.27 Bream Emulation Manual Manipulation ...... 135
6.28 Bream Emulation with EQ Match ...... 136
6.29 Bream Emulation Result ...... 137
viii ABSTRACT
Professional musicians have become increasingly independent of record companies and other investors that would seek to make a financial gain from the artist’s success. With the availability of relatively inexpensive professional recording equipment, musicians can now purchase and own the electronics they need to create their own high-quality recordings.
However, the creation of a great recording requires more than a financial investment; it requires an understanding of acoustics, recording techniques that have produced great recordings of the past, and modern audio processing methods.
The purpose of this treatise is to create a guide that will assist classical guitarists who wish to make their own professional recordings. This guide will include the documentation of nine significant guitar recordings from the last 60 years, and it will attempt to quantify sound relationships among these recordings via modern sound analysis tools. The documentation and sound analysis of these recordings will then be used in conjunction with current digital audio techniques to generate a guide for emulating a favored sonic architecture. This treatise should also provide guitarists with adequate knowledge concerning the production of professional recordings using modern recording tools.
ix CHAPTER 1
INTRODUCTION
The purpose of this research is to assist classical guitarists who might consider themselves to have little or no knowledge about recording techniques. For that reason, this treatise is intended to be accessible. However, it does contain some concepts that may require supplemental research to fully comprehend, such as the interpretation of visual graphs used for audio analysis in Chapter Three and Six, room mode calculation, microphone design and techniques, and dithering. Having over a decade of recording experience, I feel that I am able to introduce necessary concepts in a concise and effective manner, and, ultimately, my wish is to help classical guitarists disseminate their art.
As far as I am concerned, there are many classical guitarists that have little to no knowledge about basic audio recording techniques and concepts. However, within the last few years, I have become familiar with successful players of the classical guitar that have established themselves as skilled audio engineers, such as Norbert Kraft, John Taylor, Drew Henderson, and
Ricardo Marui. In my experience, it is evident that this is a rare occasion, but that is not without reason. Accomplished players put much of their efforts into performing and teaching, and this leaves little time to study a new skill, especially one that is largely based in science and technology. With that, I believe there to be a lack of desire among classical guitarists that stems from the absence of resources that relate directly to them, especially ones that are comprehensive. For that reason, this treatise was built to be a classical guitarist’s one-stop-shop for learning how to make their own professional recording.
Classical guitarists who are informed about the methods used for important recordings of the past, as well as acoustics, sound emulation techniques, and more, are equipped with the
1 ability to monetize and spread their artistry. These players can create a marketable product, and one that music consumers will want to buy. They are no longer reliant on investments from recording labels, if the artist can even get a contract, who will then expect a large return sum, and they will no longer have to pay a recording engineer thousands of dollars per session to do something that I think any motivated person can accomplish.
Content Overview
This treatise begins with a documentation of nine highly respectable recordings that I have picked based on the artist, time period, sound quality, acclaim, and/or other factors, then there is an analysis of those recordings using digital sound analysis tools, as well as a discussion about the audible and measureable characteristics of those recordings.1 The discussion compares the sonic architecture of those recordings, and it is based on the sound analysis as well as my own subjective thoughts. From there, I present the basics of home studio design, room acoustics, and recording and audio processing techniques, which is followed by a method for emulating a favored sonic architecture. At the end of the chapter on sound emulation, I document my own attempts to digitally manipulate two tracks from an anonymous artist to sound like two reference recordings. Also, a glossary to assist with audio-related nomenclature and colloquialisms has been included in the appendices.
Alternate Reading Map
For those primarily interested in information that is directly related to acquiring audio equipment and making a recording, here is a reading map specific to you:
1 All analyzed recordings were CD quality (44.1kHz and 16-bit). 2 • In Chapter Two and Three, you can skip ahead to recordings that interest you. However, you will benefit from reading the introductions for both chapters and checking the glossary when you encounter unfamiliar audio-based vocabulary.
• In Chapter 4, read these subsections: The Anatomy of a Modern Digital Recording
System, The Space, The Control Room, Acquisition of Recording Gear, and The Mobile
Studio.
• Read all of Chapter 5.
3 CHAPTER 2
DOCUMENTATION OF RECORDINGS
Much of the information documented in this chapter was collected from a questionnaire, which was sent to the recording engineer or producer for each album. All other data derived from either liner notes, online sources, or presumptions based on the recording company, recording studio, and audio technology present at the time. The producers and engineers that participated in this research are Alan Bise (Play), Asgerdur Sigurdardottir (Solo Piazolla), Norbert Kraft
(Guitar Favourites and Xavier Jara’s Guitar Recital), and John Taylor (The Leaves Be Green and Heritage: The Guitar in Venezuela). I also contacted David and Maria Russell regarding
Aire Latino, David Russell’s Grammy-winning album, to acquire any information not already included in the CD’s very thorough liner notes.
The documentation of Julian Bream’s and Sharon Isbin’s recordings are partially speculation, as I was unable to find out the exact specifics for those sessions. For Bream’s album, The Art of Julian Bream, the microphone, preamp, and recording device are all assumptions based on published books and online searches about RCA’s audio technology during the late 1950s and early 1960s. For Isbin’s album, Dreams of a World, I was unable to connect with Tobias Lehmann, the producer, or Jens Schünemann, the recording engineer. I used the Teldex website, as well as other sources not related to Isbin’s recording session, to produce the listed components and techniques that I presume to have been employed for Dreams of a
World.
The albums selected for this chapter were based on multiple factors. The most important condition was that all recordings must be of classical guitar. Next, I sought to include albums that were either Grammy-winning or Grammy-nominated, recorded by prolific recording
4 engineers, or were of guitarists that have been highly influential. The data’s intention is to illuminate the processes behind these recordings, so I chose recordings that I felt where of the highest caliber. My hope is that this knowledge will assist classical guitarists who seek to create their own high-quality recordings.
Before I move on, I am including the list of questions that were sent to each engineer, producer, and artist, and any information not gathered for a particular CD will be listed as unknown. Here is the questionnaire:
1. Where was the recording made?
2. Was the recording captured with a computer? If not, how?
3. If recorded with a computer, what software was chosen, and why?
4. If recorded with a device other than a computer, what was the device and why did you
choose to record with it?
5. What guitar was used for the recording?
6. What is the architecture and overall sound characteristic of that guitar?
7. What were the microphones and preamps used for the recording?
8. What was the approximate distance of the microphone from the subject?
9. What microphone technique was used to record the performer?
10. For digital capture, which recording interface or A/D converters were used?
11. Who facilitated the recording (i.e., recording engineer(s), mixing and mastering
engineer(s), etc.)?
12. What is the prominent surface material, sound characteristic, and approximate size of the
acoustic space that was used for the recording?
13. Was there any acoustic treatment present in the room during the recording session?
5 14. Were there any processing effects used pre- or post-recording? If yes, what effects were
used?
The following documentation has been organized chronologically, with newer recordings closer to the end of this chapter.
The Art of Julian Bream, 1960
Artist: Julian Bream
Label: RCA Victor Red Seal
Significance: This was Bream’s first recording with RCA Victor Records. Julian Bream is one of
the most revered guitarists of his generation, and he has an incredibly large collection of
recordings, which are all documented on the webpage JulianBreamGuitar.com. This site
details Bream’s life and work, and it is maintained and updated regularly by William
Chávez, a Bream enthusiast.2
Recording Date: October and November 19593
Recording Location: RCA Studio B, New York, NY, 4 although the room sound of some tracks
suggests another location may have also been used
Room Design and Acoustics: According to photos of RCA’s New York studio, Studio A had
polycylindrical diffusers and absorbers along the walls and ceiling. I am uncertain of
Studio B’s exact design, but it was meant for smaller ensembles and pianists. The
building has since been destroyed.5
Acoustic Treatment: Unknown
2 Chávez, William. 2015. JulianBreamGuitar.com. Accessed Octobober 13, 2019. www.julianbreamguitar.com. 3 Ibid. 4 Ibid. 5 Roy, James V. 2014. scottymoore.net. Accessed October 17, 2019. www.scottymoore.net/studio_rca.html. 6 Guitar: 1957 Herman Hauser Jr.6
Recording Device (presumed): Ampex tape machine7
DAW: Not available during this time period
Microphone (presumed): (2) RCA Type 77-DX (ribbon, bidirectional)8
Microphone Technique (presumed): Spaced Pair or Blumlein
Microphone Placement (presumed): Maybe within 5 feet, but the microphone’s distance appears
to change for some tracks
Preamp: It is likely that the engineers used the preamps on a custom RCA mixing board9
A/D Converter: Not Available during this time period
Processing (presumed): Unknown, but it is possible that equalization was used to reduce mechanical noise (i.e., noise created by recording equipment)
Facilitators: Lewis Layton, recording engineer; Peter Delheim, producer and editor
Guitar Favorites, 1996
The information documented for this Album comes from the previously noted questionnaire, which was emailed to and answered by Norbert Kraft (engineer), and that includes any direct quotations.10
Artist: Norbert Kraft
Label: Naxos Records
6 Chávez, William. 2015. JulianBreamGuitar.com. Accessed Octobober 13, 2019. www.julianbreamguitar.com. 7 Roy, James V. 2014. scottymoore.net. Accessed October 17, 2019. www.scottymoore.net/studio_rca.html. 8 Roy, James V. 2014. scottymoore.net. Accessed October 17, 2019. www.scottymoore.net/studio_rca.html. 9 Roy, James V. 2014. scottymoore.net. Accessed October 17, 2019. www.scottymoore.net/studio_rca.html. 10 Kraft, Norbert, interview by Philip Logan. 2019. Email Questionnaire (September 11 and 12). 7 Significance: Norbert Kraft is an award-winning guitarist and recording engineer, including first
prize in the 1985 Andrés Segovia International Guitar Competition and a Grammy Award
for his recording of the Parker Quartet playing Ligeti’s String Quartets 1 and 2. Kraft is
currently the artistic director for Naxos Record’s Guitar Collection, which he established.
Guitar Favourites is a self-recorded CD.
Recording Date: August 19–23, 1996
Recording Location: St. John Chrysostom Church, Newmarket, Ontario, Canada
Room Design and Acoustics: According to Norbert Kraft, St. John Chrysostom is “a mid-large
church [that seats] approximately 500 persons.” A modern design with splayed walls
(“quarter pie”), and the far wall is a “5-segment quasi circle.” The side walls are 70 feet
long, and the ceiling is 60 feet (altar) at its highest point and18 feet (far wall) at its lowest
point. “The side walls are made of semi-smooth clay brick with rounded mortar between
the bricks, and the floors are currently tile, although before the year 2000, there was a fair
amount of carpet on the altar and audience floor.” Reverb time is 3.5 seconds, and
performance area (altar) is 18 feet from the nearest wall. “This allows… freedom to be
near the source instrument and still contain a great deal of acoustic resonance.” Norbert
mentioned that he uses the sacristy, which is adjacent to the performance area, as an
isolated monitoring room when recording at this location.
Acoustic Treatment (presumed): None
Guitar: Norbert Kraft used three different guitars for this album, all of which were built by
Paulino Bernabe. The tops are either spruce or cedar, and I presume that they are fan-
braced traditional builds.
Recording Device: Digital Audio Tape (DAT)
8 DAW: Pro Tools
Microphone: (2) AKG C12 (tube, multipattern) and (2) Neumann KM 140 (condenser, cardiod)
Microphone Technique: Both microphone pairs are positioned in A/B. The AKG C12 was likely
set to an omnidirectional or cardiod polar pattern, but that is presumed.
Microphone Placement: AKGs are 30 inches apart, 57 inches high, and 57 inches from the guitar.
Neumanns are approximately 9 feet apart, 10 feet high, and 12 feet from guitar.
Preamp: Neve mixer
A/D Converter: Apogee (20-bit)
Processing: POW-R dithering for CD
Facilitators: Norbert Kraft, recording engineer; Bonnie Silver, producer and editor
Dreams of a World, 1999
Artist: Sharon Isbin
Label: TELDEC
Significance: This album won a Grammy award for Best Instrumental Soloist.
Recording Date: Possibly March 1999
Recording Location: Teldec (now Teldex) studio, Berlin, Germany
Room Design and Acoustics: I presume the recording room to be Teldex’s 19th-century hall,
which is 1492 feet.2 Based on photos,11 the ceiling appears to be approximately 30–40
feet high. Pictures of the room show wood floors and walls with irregular paneling, and I
assume the irregular paneling assists in diffusion. Some photos also contain large Gobo
11 Teldex Studio Berlin. n.d. Studios. Accessed June 21, 2019. http://www.teldexstudio.de/studios. 9 traps, which are large broadband absorbers on wheels that are used for isolating sound
sources.
Acoustic Treatment: Unknown
Guitar: Thomas Humphrey 1988 Millennium guitar12
Recording Device (presumed): Computer
DAW (presumed): Pyramix13
Microphones (presumed): Sennheiser MKH-20 (condenser, omnidirectional), Neumann TLM 50
(condenser, cardiod), Neumann KM 83 (condenser, omnidirectional)1415
Microphone Technique: Unknown
Microphone Placement: Unknown
A/D Converter: Unknown
Preamps (presumed): Millennia HV-3D16
Facilitators: Tobias Lehmann, producer and editor; Jens Schünemann, recording engineer
Aire Latino, 2004
Most information included in the documentation of this album has be provided by the
CD’s liner notes.17 The inclusion of the mixing console and monitoring solution is unique, and no other documented album includes this information.
Artist: David Russell
12 Isbin, Sharon. 1999. Dreams of a World. 3984-25736-2. CD. 13 Teldex Studio Berlin. n.d. Control Room 1. Accessed June 21, 2019. http://www.teldexstudio.de/equipment. 14 Teldex Studio Berlin. n.d. Microphones. Accessed June 21, 2019. http://www.teldexstudio.de/microphones. 15 Sengpiel, Alexander, and Eberhard Sengpiel. n.d. Vorlesungs-Unterlagen 5. Accessed July 31, 2019. http://www.sengpielaudio.com/Klavier1.pdf. 16 Teldex Studio Berlin. n.d. Control Room 1. Accessed June 21, 2019. http://www.teldexstudio.de/equipment. 17 Russell, David. 2004. Aire Latino. 089408061226. CD. 10 Label: TELARC
Significance: Aire Latino was awarded a Grammy for Best Instrument Soloist.
Recording Date: April 18–20, 2003
Recording Location: Peggy and Yale Gordon Center for Performing Arts, Owings Mills,
Maryland
Room design and acoustics: Large performance hall with a seating capacity of 550. Stage
dimensions are 50 x 40 x 26 feet. Stage flooring is wood. Audience flooring is partially
carpeted, the seats are padded, and there are thick curtains for the stage.18
Acoustic Treatment: RPG Diffusors and VariScreen Panels
Guitar: Matthias Dammann double-top guitar with D’Addario strings
Recording Device (presumed): Computer
DAW: SADiE
Microphones: Coles 4038 (ribbon, bidirectional), Sennheiser MKH-20 (condenser,
omnidirectional) MKH-30 (condenser, bidirectional)
Microphone Technique: Unknown
Microphone Placement: Unknown
Preamps: Millennia HV-3D
A/D Converter: Lexicon 20/20 AD, custom engineered by Bruce Vark
Monitoring Solution: Monitored through Krell Studio 20-bit DA converter with ADS 1530
monitors which were amplified by Threshold SA/4e Stasis Pure Class A amplifiers
Mixing Console: Panasonic WRS-4412 custom engineered by John Windt
18 Jewish Community Center of Greater Baltimore . n.d. The Gordon Center: Rent Our Space. Accessed June 21, 2019. jcc.org/gordon-center/rental. 11 Processing: EQ on Panasonic console was utilized prior to digital capture.19 POW-R dithering
Facilitators: Rosalind Ilett, producer and editor; Thomas Knab, Recording Engineer; Robert
Woods, executive producer; Erica Brenner, production supervisor
Solo Piazzolla, 2007
The information documented for this Album comes from the questionnaire, which was emailed to and answered by Asgerdur Sigurdardottir (engineer).20
Artist: Manuel Barrueco
Label: Tonar
Significance: Solo Piazolla was nominated for a Grammy award.
Recording Dates: Invierno Porteño Recorded September 20, 2005; Verano Porteño recorded
October 7, 2005; Five Pieces recorded Jan 30 and February 4, 2004; Tango-Études
recorded October 11–13, and November 21, 22, and 27, 2006
Recording Location: Cellar of Manuel’s home
Room Design and Acoustics: ~20x20 feet, possibly concrete floors
Acoustic Treatment: Wide-band sound absorption panels on walls, blankets over wine bottles
Guitar: Robert Ruck No. 58, built in 1972. An experimental design made of Brazilian rosewood
with laminated sides, back braces made of European spruce, and Brazilian conifer for the
top lining. French polish on top with brushed-on violin varnish on the back and sides.
Originally sold for approximately $450.21 I believe the guitar is fan-braced.
19 Russell, Maria and David, interview by Philip Logan. 2019. Email Questionnaire (June 21). 20 Sigurdardottir, Asgerdur, interview by Philip Logan. 2019. Email Questionnaire (October 16) 21 Jackson, Blair. 2018. Clasical Guitar Magazine. August 15. Accessed October 16, 2019. https://classicalguitarmagizine.com/noted-luthier-robert-ruck-builder-of-manuel-barrueco-famous-no-58-passes- away-at-72/. 12 Recording Device: Computer
DAW: Pro Tools
Microphones: (2) DPA 4006 (condenser, omnidirectional)
Microphone Placement: ~5 feet from performer
Microphone Technique: AB
A/D Converter: Apogee Rosetta
Preamp: Avalon 2022
Post-Processing: Sony S-777 reverb
Facilitators: Asgerdur Sigurdardottir, engineer, producer, editor; Ed Tetreault, mastering
Heritage: The Guitar in Venezuela, 2010
The information documented for this Album comes from the questionnaire, which was emailed to and answered by John Taylor (engineer), and that includes any direct quotations.22
Artist: Ricardo Iznaola
Label: Iznaola Guitar Works
Significance: Ricardo Iznaola is an award-winning guitarist and author, and this album contains a
large collection of Venezuelan pieces for the guitar. Also, it was recorded by John
Taylor, a prolific recording engineer who specializes in classical music.
Recording Dates: August 14–16, 2006, “except for Disc 2, tracks 13–15, [which are] taken from
a 1970 studio recording in Caracas.”
Recording Location: St. Andrew’s Church, Toddington, Gloucestershire, England
22 Taylor, John, interview by Philip Logan. 2019. Email Questionnaire (June 25). 13 Room Design and Acoustics: “Complicated shape with many different dimensions, which I’ve
never tried to measure.” Recording was made in the “moderately reverberant area just to
the nave side of the choir stalls… with mics placed close enough to the guitar [so] that
the church acoustic doesn’t dominate the overall sound, just gives a spacious ‘glow’
around the notes.” The church has stone walls covered by plaster, and stone floors with
“rush matting in most of the nave.” The ceilings are high and wooden, “which gives both
space and warmth to the sound.”
Acoustic Treatment (presumed): None
Guitar: 2002 Andrea Tacchi. The guitar is assumed to be a traditional fan-strutted design with a
spruce top, and John Taylor states that it was chosen “in preference to a louder double-
top alternative… for its fine balance and range of colors.”
Recording Device: Computer
DAW: Magix Sequoia
Microphone: “Brüel and Kjaer (now renamed DPA) 4006 omni fitted with UA0777 nose cones.”
Microphone Technique: A/B
Microphone Placement: “66 inches from the bridge, 59 inches above the floor, and offset to the
guitarist’s right, i.e., nearer to the bridge than the fingerboard.”
Preamp: Buzz Audio MA-2.2 (designed specifically for recording acoustic guitar)
A/D Converter: RME Fireface 800
Processing: Sequoia’s native multi-band compression to limit peaks, and to bring overall volume
up 4dB
Facilitators: produced, edited, and mastered by John Taylor
14 Play, 2014
The information documented for this Album comes from the questionnaire, which was emailed to and answered by Alan Bise (producer), and that includes any direct quotations.23
Artist: Jason Vieaux
Label: Azica Records
Significance: Play won a Grammy for Best Classical Instrumental Soloist.
Recording Date: Unknown
Recording Location Location: Shrine Church of St. Stanislaus
Room Design and Acoustics: “The church is enormous, roughly 85 feet by 200 feet.” The
interior design includes ceramic tile floor, plaster walls, and wooden pews, and the area
around the altar is carpeted.
Acoustic Treatment: None
Recording Device: Computer
Guitar: 2005 Gernot Wagner. Double top soundboard with a spruce top layer and cedar bottom
layer.
DAW: Magix Sequoia
Microphone: (2) Sennheiser MKH-20 (condenser, omnidirectional)
Microphone Technique: A/B
Microphone Placement: ~5 feet
Preamp: Millennia Media
A/D Converter: Lavry Engineering Blue
Processing: high pass filter set to ~50Hz, likely POW-R2 dithering
23 Bise, Alan, interview by Philip Logan. 2019. Email Questionnaire (July 11). 15 Facilitators: Bruce Egre, recording engineer; Alan Bise, producer
The Leaves Be Green, 2015
The information documented for this Album comes from the previously noted questionnaire, which was emailed to and answered by John Taylor (engineer), and that includes any direct quotations.24
Artist: VIDA Guitar Quartet
Label: BGS Records
Significance: This album was recorded by John Taylor, a prolific engineer who specializes in
classical music.
Recording Dates: April 9–10 and July 13–14, 2015
Recording Location: Holy Trinity Church, Weston, Hentforshire, UK
Room Design and Acoustics: Estimated to be 35 x 25 x 25 feet with an “irregular shape;” walls
are plaster on thick stone; floor is stone, except for a “large area of wood” that creates a
low-rise stage. “All this give the notes a pleasant bloom, but the lack of a really high
vaulted ceiling means that the reverberation decays fairly quickly.”
Acoustic Treatment (presumed): None
Guitars: Mark Eden plays a 2000 Christopher Dean guitar; Mark Ashford plays a 2007
Christopher Dean guitar; Amanda Cook plays a 2014 Bert Kwakkel Viscivorus guitar;
Chris Stell plays a 2012 Christopher Dean 7-string guitar. All use D’addario strings.25
“All four guitars are traditional fan-strutted spruce tops, built for concert players who are
24 Taylor, John, interview by Philip Logan. 2019. Email Questionnaire (June 25). 25 Classical Guitar Magazine. 2017. Inside England's Vida Guitar Quartet: Stretching Possibilities. February 1. Accessed June 16, 2019. https://classicalguitarmagazine.com/inside-englands-vida-guitar-quartet-stretching- possibilities/. 16 looking for a full dynamic range, but with subtlety, good sustain to make their lines sing
and a wide variety of colors.”
Recording Device: Computer
DAW: Magix Sequoia. “A comprehensive recording, editing and mastering workstation,
sonically transparent, and designed to be fast and flexible for editing classical music from
multiple takes”
Microphones: (2) Schoeps MK2 (condenser, omnidirectional)
Microphone Technique: A/B “angled slightly out to cover width of quartet.”
Microphone Placement: 16 inches apart, 60 inches from saddle (measured from the nearest
microphone to each player—curved formation), 50 inches above the floor
Preamp: Buzz Audio MA-2.2, which are designed specifically for recording acoustic guitar.
A/D Converter: Prism Sound Orpheus
Processing: limiter used on input, with a 4dB maximum GR. Reverb added with Bricasti M7 to
“enhance apparent space”
Facilitators: produced, edited, and mastered by John Taylor
Guitar Recital, 2016
The information documented for this Album comes from the previously noted questionnaire, which was emailed to and answered by Norbert Kraft (engineer), and that includes any direct quotations.26 Although this is the same location and recording engineer as Guitar
Favorites, the carpet has been pulled up from the floors, and Kraft utilizes different
26 Kraft, Norbert, interview by Philip Logan. 2019. Email Questionnaire (September 11 and 12). 17 microphones, converters (higher bitrate), and preamps, as well as a different recording device.
Along with that, the performer, guitar, and placement of microphones are also different.
Artist: Xavier Jara
Label: Naxos Records
Significance: This album was made in partial fulfillment of Xavier Jara’s prize for winning
Guitar Foundation of America’s international guitar competition, and it is part of Naxos’
Laureate Series. It was recorded by the extremely prolific and sought-after engineer,
Norbert Kraft.
Recording Date: January 5–7, 2017
Recording Location: St. John Chrysostom Church, Newmarket, Ontario, Canada
Room Design and Acoustics: According to Norbert Kraft, St. John Chrysostom is “a mid-large
church [that seats] approximately 500 persons.” A modern design with splayed walls
(“quarter pie”), and the far wall is a “5-segment quasi circle.” The side walls are 70 feet
long, and the ceiling is 60 feet (altar) at its highest point and18 feet (far wall) at its lowest
point. “The side walls are made of semi-smooth clay brick with rounded mortar between
the bricks, and the floors are currently tile, although before the year 2000, there was a fair
amount of carpet on the altar and audience floor.” Reverb time is 3.5 seconds, and
performance area (altar) is 18 feet from the nearest wall. “This allows… freedom to be
near the source instrument and still contain a great deal of acoustic resonance.” Norbert
mentioned that he uses the sacristy when recording at this location, which is adjacent to
the performance area, as an isolated monitoring room.
Acoustic Treatment (presumed): None
Guitar: Hugo Cuvilliez. I presume it to be a lattice-braced guitar.
18 Recording Device: SADiE LRX connected to computer
DAW: SADiE
Microphone: Sonodore MPM-81 (tube, multipattern)
Microphone Technique: A/B, microphones set to omnidirectional polar pattern.
Microphone Placement: 22 inches apart, 55 inches high, 59 inches from guitar
Preamp: Horus interface from Merging Technologies
A/D Converter: Horus interface from Merging Technologies
Processing: POW-R dithering for CD
Facilitators: Norbert Kraft, recording engineer, editor, and producer; Bonnie Silver, producer
19 CHAPTER 3
AUDIO ANALYSIS
The images in this chapter are screen shots of Izotope RX 7, which is a comprehensive post-production and audio repair software with great analysis tools, and PAZ Position by Waves, which is a digital tool that measures an audio tracks stereo width. With these, I have collected three kinds of data: imaging (stereo width), waveform statistics (loudness), and spectral
(frequency). These measurements are helpful during the mastering and emulation stages, providing valuable insight when trying to create or match a particular sonic architecture.
Figure 3.1 PAZ Stereo Position Display
Figure 3.1 is an example of PAZ Position by Waves, which is used to measure and display the stereo position and width of an audio signal. Essentially, it displays the position and volume of sound relative to the left and right stereo field. A narrow image will exhibit a concentration of sound near the center vertical line of the stereo graph, such as seen in Julian
Bream’s recording of Lennox Berkeley’s Sonatina, Op. 51 (Figure 3.4), and a wide image will
20 have measurements closer to the left and right boundary lines (before the antiphase area in red),27 such as in the VIDA Guitar Quartet’s recording of Benjamin Britten’s Simple Symphony, Op. 4
(Figure 3.1). For each individual track, I processed large portions of the recording to produce an accumulated result, which provides, in my opinion, a more accurate image of the overall width.
Figure 3.2 Waveform Statistics
The waveform statistics in Figure 3.2 are based on the international loudness standards defined in ITU-R BS. 1770,28 and there are at least three important measurements for our purposes: true peak level (denoted as dBTP), integrated loudness (measured in LUFS), and loudness range (measured in LU). There are other measurements available, but these three values provide the most important information when attempting to match another audio recording’s loudness: maximum peak level, average loudness, and dynamic range. The true peak statistic
(dBTP) oversamples the audio signal at least 4 times to increase precision of peak level
27 Anti-phase and out-of-phase are the same. The red anti-phase section of the Stereo Position Display communicates that there are parts of the audio that are not phase aligned. Quick, inaudible spikes of the signal in anti-phase are typical in most stereo recordings. 28 European Broadcasting Union. 2014. ITU-R 128 BS. 1770. June. Accessed November 26, 2019. https://tech.ebu.ch/docs/r/r128.pdf. 21 measurements. Integrated loudness is the average loudness of the entire audio track, and this statistic is measured in LUFS.29 Loudness range is a general computation of an audio recording’s overall dynamic range, and it is measured in LU.30
Figure 3.3 Spectral Graph
Figure 3.3 is a spectral graph, and it commonly displays real-time, momentary frequency measurements for a given audio source. However, I want to see a measurement of the entire track as one instance, as that provides a visual of the recording’s accumulated frequency measurements. The accumulated measurements display the overall tonal balance of a recording, which I refer to as its frequency curve. To do this, I simply highlight the entire track in Izotope
RX, then the graph automatically displays the combined measurements in non-real time. Seeing all measurements in one instance helps me quickly compare frequency measurements of various audio files. When viewing the frequency curve, I set the graph’s resolution (“FFT size”) quite
29 LUFS (Loudness Units Full Scale) utilize a filter when measuring an audio signal, which makes the measurements more representative of human perception of sound. 30 LU (Loudness Units) are like to decibels, except they do not require air pressure to be measured. 22 low. This highlights the contour of the curve by reducing drastic peaks and nulls measured in the signal.
The recordings selected for analysis were based on criteria. Most importantly, I wanted tracks that share the same tonic and key (or closely related key), which ended up being A major/minor. This is because the key signature is a good determinate of what pitches
(frequencies) will be emphasized in the spectral measurements, and this aids in the search for dissimilarities among recordings. Next, the piece must cover a large portion of the instrument’s available notes, because that increases the spectrograph’s coverage of the guitar’s frequency range. The last considerations were loudness range and musical characteristics of the composition (fast or slow, dense or sparse, changes in dynamic levels). In the end, since I was most concerned with the tonic and key signature of each track, there are still quite varied musical characteristics between a few of the analyzed recordings. The last recording, which is of Xavier
Jara playing Jeremy Collins’ Elegy, does not fit any of the criteria, but I included it because of its unbelievable sound quality.
The Art of Julian Bream
Artist: Julian Bream
Piece: Sonatina, Op. 51: III. Rondo
Composer: Lennox Berkeley
Key: A
Length: 3:14
Because I was unable to find the exact specifics for The Art of Julian Bream, the processes that formed this album remain a mystery to me. However, there are obvious
23 differences between this track and the others analyzed in this chapter, as well as among the other tracks from this same album. First, there is mechanical noise present, which I assume is introduced by the recording device (likely a tape machine) and preamps. Second, one very peculiar difference I noticed between this track and some others from this album is the amount of reverb. This recording seems to be in different recording space, or it has additional reverb.
Artificial reverb could have been added by employing a reverb chamber or reverb plate, as both techniques were in use during this time. I suppose it is possible that this recording was not created in the same space as some others, but then there is the question of which space was used.
It could also be that the microphones were simply moved farther away from the guitar for this recording, and that allowed the microphone to capture more of the room’s reflections.
Figure 3.4 Bream Stereo Image
Figure 3.4 shows that this particular recording is not very wide, even though the album was advertised as being in “true stereo”—other tracks from this same album are wider. With that in mind, you will find that this is the only recording without signal spikes in the anti-phase area. I think that this suggests a coincidental or very near-coincidental microphone technique. If the presumed microphones used are ribbon (RCA 77-DX), then it is likely that the Blumlein technique was used, which is a coincidental technique (similar to X/Y) that should be free of any
24 phase issues. However, the microphones could be set in A/B (spaced pair) within a very close proximity (around 1 foot), which should be free or mostly free of out-of-phase occurrences. Both techniques used in combination with a ribbon microphone will produce a narrow stereo image.
Figure 3.5 Bream Waveform Statistics
The loudness stats in Figure 3.5 show this recording to have a high true peak level, as well as the highest integrated loudness figure (-19.3 LUFS), which means that this recording is likely the loudest of the group. The loudness range is average at 10.9 LU.
Figure 3.6 Bream Spectrograph
25 The spectrograph measurements in Figure 3.6 are the most intriguing. The high frequency measurements are not accurate because of the existence of mechanical noise, and due to that fact, the frequency curve begins to level around 6kHz. This noise also poses a fidelity issue. It is not overly intrusive, but the signal-to-noise ratio is subpar to today’s standards. The playing, as is familiar with Bream, is sometimes brittle and bright, and along with that, there is a lack of bass response below 200Hz, which is apparent in the sound.
Guitar Favourites
Artist: Norbert Kraft
Piece: Romanze
Composer: Niccolo Paganini
Key: A minor
Length: 4:10
Norbert Kraft’s classical guitar recordings have always struck me as some of the most beautiful guitar recordings in the history of recorded sound. The quality of sound is incredibly rich, and he is always successful at highlighting the player’s natural sound qualities. I believe he has also found what may be the best space for instrumental recordings (St. John Chrysostom
Church, Newmarket, Ontario, Canada) or at the very least, the best space for classical guitar recordings. Listening to his self-recorded album, Guitar Favorites, I hear an early prototype of the sound that would ultimately make him such a sought-after recording engineer.
Figure 3.7 displays a balanced and centered stereo image; however, the image is not very wide. Referring back to the documentation in Chapter Two, Norbert Kraft uses two small- diaphragm cardiod microphones (KM 140) placed 9 feet apart (spaced pair), 10 feet high, and 12
26
Figure 3.7 Kraft Stereo Image feet from the guitar (room mics). I’m not sure of the exact balance between the close mics
(placed 5 feet from the guitar and 2.5 feet apart) and room mics, but the room mics do not seem to add much to the stereo width (the center of the image has much more sound pressure than the sides). This may be due to the fact that the KM 140 microphones are cardiod, not omnidirectional. A cardiod microphone, because of its directionality, is not able to capture reflections from the sides and back of the room (the off-axis area of the microphone’s polar pattern) in the same way an omnidirectional microphone can, which translates to a narrower image. Since the AKG C12 (close mics) is a multi-pattern microphone, it could also be set to a cariod polar pattern. However, I assume that it is set to omni.
The loudness measurements in Figure 3.8 show signal peaks that are perfectly balanced in the left and right channels (-5.59 dBTP), and they are the lowest values out of all other tracks in this chapter. The integrated loudness is also quite low (-25.9 LUFS), but the loudness range is average (11.7 LU). The spectral analyzer in Figure 3.9 displays a high density of sound pressure from ~180Hz to ~600Hz, which gives the guitar body and warmth, but not to the point of reducing clarity. The audio drops significantly below that ~180Hz point, and I am positive that there is a high-pass filter being utilized, especially since Jara’s recording, which was
27
Figure 3.8 Kraft Waveform Statistics recorded in the same room, displays more subharmonic noise. Above the ~600Hz mark, the contour of the audio appears to have a subtle but wide dip that lasts until 6kHz. After that point, there is a flattening of the slope, which may be due to the AKG C12’s frequency response.
Figure 3.9 Kraft Spectrograph
28 The guitars are likely fan-strutted traditional builds, and Kraft’s masterful performing abilities allows them to sing as one would expect. The darkness of the tone quality of the guitar for this specific track (he used three different guitars for the album) makes me believe that this is the cedar-top guitar, and it is a great choice for this piece of music. During the recording, all four mics were sent to the Neve mixer, which was likely used for preamplification, and mixed down to two channels prior to reaching the Apogee converter. The electric signal is converter to 20-bit digital, then it is routed to the DAT (digital audio tape). Overall, the room’s sound quality is warmer and less reflective than in Xavier Jara’s later recordings, and that is likely due to carpet on the altar and audience floor, which was removed after the year 2000.
Dreams of a World
Artist: Sharon Isbin
Piece: Appalachian Dreams, Op. 121: V. Finale
Composer: John Duarte
Key: A mixolydian
Length: 3:39
Since I was unable to contact anyone that could assist me in documenting this recording
(Sharon Isbin was busy with a deadline, and all emails to the recording engineer and producer went unanswered), I have taken an investigative, as well as speculative, approach to understanding what went into producing this sound. There is little information about the recording session in the CD’s liner notes, but I was able to find a two-dimensional diagram of a piano session at Teldex Studios, which might transfer well to the guitar.31
31 Sengpiel, Alexander, and Eberhard Sengpiel. n.d. Vorlesungs-Unterlagen 5. Accessed July 31, 2019. http://www.sengpielaudio.com/Klavier1.pdf. 29 The actual process for this album is mostly unknown to me, but I would like to present a theory. My assumption is that a pair of Sennheiser MKH-20 microphones were placed approximately 2 feet away, center with the guitar; a pair of Neumann TLM microphones were placed off-center, about 5–10 feet from the guitar; and a pair of Neumann KM 83 microphones were used as boundary microphones approximately 20 feet from the instrument. The KM 83s would have been placed in front of a thick piece of plexiglass (boundary). The glass reduces phase interference from late reflections, producing a roomy but clean sound. The CD’s liner notes say that a “special Teledec arrangement” was used, and this theory qualifies as special. The engineers could have used a pair of omnidirectional microphones at about 5 feet from the instrument, which, according the documentation in Chapter Two, turns out to be quite common, but it seems unlikely that a world-class artist would go to a specialized recording studio in a distant country just to have engineers use a simple A/B omni setup.
In the case that I am correct, a balance of the microphones and positions I suggested would produce a very reverberant sound that is also quite present and bright, and that is what I hear. The recording has a lot of reverb, but the source is still close and open with plenty of bass.
The sound is somewhat heavy around G2 (98Hz), but many guitars I have played resonate strongly to or around G. The reason for this is that the guitar’s body acts as a Helmholtz resonator that is reactive to frequencies between F-sharp and A, and this helps the instrument produce low frequencies.32 Because of this, the various G pitches throughout frequency spectrum, regardless of their octave, can receive a natural stress. Personally, I think the there is too much brilliance in the sound that, consequentially, emphasizes nail-related noise in Isbin’s attacks, but that may also be inherent in her sound.
32 The Univerisity of New South Wales. n.d. How Does a Guitar Work? Accessed November 29, 2019. https://newt.phys.unsw.edu.au/music/guitar/guitarintro.html. 30
Figure 3.10 Isbin Stereo Image
Figure 3.10 demonstrates what a wide stereo image looks like. It is nearly identical to the width of Russell’s recording of Danza Brasilera (Figure 3.7), but with a bit more concentration of sound pressure at the center of the stereo image. The waveform statistics in Figure 3.11 show that this recording has relatively balanced peak values that might, as a sum, be higher than any other recording from this chapter, but the integrated loudness is lower than Russell or Vieaux’s recordings (-21.2 LUFS) with an overall dynamic range that is greater (11.1 LU).
Figure 3.11 Isbin Waveform Statistics
31
Figure 3.12 Isbin Spectrograph
The spectrograph in Figure 3.12 displays a very linear slope from ~250Hz to ~17kHz, but there is a slight dip from ~600–800Hz. Frequencies from 200Hz and below are displayed as having a quick downward slow, with exception to the high resonance at 98Hz. This is expected because of the typical resonance of the guitar, as well as the composition’s mixolydian quality.
You will see similarities with other pieces that utilize G2, such as Vieaux’s recording of Jongo, which also has a mixolydian quality. Also, there is a rather narrow peak at 6k, and I believe that this emphasis in the frequency range is due to the nail-related brightness heard in Isbin’s sound.
Aire Latino
Artist: David Russell
Piece: Danza Brasilera
Composer: Jorge Morel
Key: A major
Length: 3:38
32 Although this piece of music, as well as the performance, is not the most dynamic from
Russell’s album, it does fit well with the other pieces in this chapter, in terms of key signature and coverage of the guitar’s range. However, what really sets this recording apart is the punchiness of the guitar’s sound, and that can be partially, if not wholly, attributed to the guitar’s design. This incredibly expensive (some Matthias Damman guitars sell for around $60,000) and masterfully built instrument comes from Germany, and it produces a huge, robust sound. The guitar is a double-top design, which means that the guitar’s soundboard is essentially two thin layers, as opposed to a traditional single layer. I have found that some guitarists who use double- top guitars on the concert stage opt for a fan-strutted, traditional design when recording. If you refer to the documentation of Ricardo Iznaola’s album (Chapter Two), Heritage: The Guitar in
Venezuela, Iznaola chose a traditional fan-strutted design over a double-top because of its musicality. In my experience, the universal opinion is that fan-strutted designs have a more diverse musical palate, as well as more dynamic range, but that may also be subject to other factors, such as the luthier, overall design, and quality of wood.
The spectrograph in Figure 3.13 shows this recording to be a bit rounder than others, which essentially means that there is a higher density of sound pressure from 200–1000Hz. This creates a rounded contour in the midrange frequencies. Then there is a linear downward slope that ends with a very noticeable energy loss from 6kHz to 15kHz. My opinion is that the double- top guitar has had some effect on this outcome, as well as the use of a ribbon microphone, which is generally less sensitive to high frequencies than other microphone types. Another important aspect of this recording is its spaciousness. This is in part due to the large recording space, but also due to what I believe is the microphone technique and placement.
33
Figure 3.13 Russell Spectrograph
First, the space is huge, and really much larger than necessary for the instrument. If the engineers used one stereo pair within 5 feet of Russell, I doubt that they would have captured the sound heard on this track, at least not without adding reverb during post-production. We also know that diffusers and absorbers were on stage with Russell, which were likely used to reduce early reflections and coloration produced by the wood stage, and that the hall had padded seats and carpet that further absorbed the room’s effects.
Second, at least three microphones were used during the recording, giving the engineers the ability to place microphones at different points in the room. One or more pairs of microphones are placed close to the instrument, and another microphone or microphone pair are placed further away to capture the hall’s late reflections. The presumed microphone technique is a close A/B pair with the Coles 4038 (bidirectional) ribbon microphone, and a M/S pair with the
Sennheiser MKH-30 and MKH-20 are placed in the hall’s seating area. This assumption is based on the listed microphones (refer to Chapter Two), spectral graph, and sound characteristic of the recording. I believe this microphone technique and placement would produce a wide and natural image of the guitar, as well as the hall’s dense reverberations. All microphones are mixed
34 together and equalized with the Panasonic WRS-4412 prior to reaching the 2-channel Lexicon
20/20 A/D converter.
Figure 3.14 Russell Waveform Statistics
Looking at the waveform statistics in Figure 13.14, we see that the dBTP and integrated loudness (-20.3 LUFS) are quite high (second to Bream’s), but the loudness range (8.2 LU) is lower than all the other tracks from this chapter. This may be partially due to the compositional style of the piece, but I also think that the guitar’s double-top design, which is generally recognized for being louder and less dynamic, may also be an important factor. PAZ (Figure
3.15) exhibits a very wide and diffused audio recording, and although the graph shows that the image is quite centered, there is, when listening, a very subtle heaviness to the left side of the image.
Looking back at the spectrograph (Figure 3.13), there is one more factor I would like to point out. The upward shift starting around 15kHz is the effects of dithering, which is defined in the glossary. Russell’s recording states that POW-R dithering was used to reduce bit depth from
20 to 16, and the POW-R1 noise curve appears to be the right match for this high-frequency
35
Figure 3.15 Russell Stereo Image anomaly. Dithering effects can also be seen in the spectrograph for Sharon Isbin (Figure 3.12) and the VIDA Guitar Quartet (Figure 3.28).
Before I move on, I would like to mention the kind of reverb present in this recording. It is completely natural, but if you listen, you will notice how dark it is. The seat padding, carpet, on stage absorbers, and size of the room all contributed to this very warm tone, and because of the quality of the reverb, the clarity of the source is not lost. The performance is completely intelligible, even among the thick, lingering reflections of the hall.
Solo Piazzolla
Artist: Manuel Barrueco
Piece: Verano Porteño
Composer: Astor Piazzolla
Key: A minor
Length: 4:22
The first characteristic that I notice in Barrueco’s recording is its intimacy. The wine cellar puts its mark on the recording not just by its resonance, but also by its smaller size, which
36 translates to a dominance of early reflections. This means that the space sounds small, and that provides the sense of intimacy. However, the A/B pair of DPA microphones still generate a noticeable openness in the sound that keeps the recording from exhibiting a confined quality, and that is likely due to the omnidirectional polar pattern of the DPA 4006 microphones. Also, the additional reverb from the Sony S-777 increases spatiality by providing late reflections that likely did not exist in the original sound capture, especially since the room was treated with wide-band absorbers and blankets to “dry [the] sound,” as stated by Asgerder Sigurdardottir, the recording engineer. 33
Figure 3.16 Barrueco Waveform Statistics
As shown by Figure 3.16, this recording is relatively quiet (low dBTP value), but it has more dynamic range than most other tracks from this chapter (loudness range of 14.6 LU). The reason for this range is likely a) Barrueco plays very dynamically, which I think is necessary for this music, and b) the piece is composed with longer sections of quietness than some of the other
33 Sigurdardottir, Asgerdur, interview by Philip Logan. 2019. Email Questionnaire (October 16). 37 pieces from this chapter. Integrated loudness (-25.8 LUFS) is tied with Kraft’s recording of
Romanze as the second lowest value of all the tracks analyzed in this chapter, and that is the figure that correlates most with perceived loudness. Figure 3.17 shows that the recording is mostly centered, but I do see, and audibly perceive, a little more strength to the left side of the stereo image. The concentration of sound energy near the center of the sound image visualizes a source that is not as wide as other recordings from this chapter (similar to Romanze).
Figure 3.17 Baurreco Stereo Image
Figure 3.18 Barrueco Spectrograph
38 The spectrograph in Figure 3.18 exhibits an even but slightly heavy low-end, a very subtle dip from 700–1000Hz, a narrow resonance around 1250Hz and 2500Hz (these resonances may be the effects of having glass wine bottles in the room), then it is relatively flat until a noticeable boost around 10kHz that is followed by a roll off at ~20kHz. Also notice that there is a strong dip below 100Hz, which looks to be from a high-pass filter (low cut). Although not always necessary, a high pass filter around 50Hz can remove a lot of unwanted subsonic noise and rumble, which is especially necessary if you are recording too close to a busy road or machinery (e.g., HVAC).
Heritage: The Guitar in Venezuela
Artist: Ricardo Iznaola
Piece: Yacambú
Composer: Antonio Lauro
Key: A major
Length: 3:17
To my ear, and looking at the spectrograph in Figure 3.19, I hear similarities in the low- end response (around 100Hz) between this recording and the VIDA Guitar Quartet’s recording of “Boisterous Bourée” (Figure 3.28). Similarities should be expected since both churches share similar building materials (stone, plaster, and wood), are both medium sized halls, and John
Taylor, the recording engineer for Iznaola and VIDA, used a similar mic technique and placement. However, there are more differences than commonalities, and the first that I notice is the high-frequency presence around 10kHz exhibited by the flattening of the slope in the
39
Figure 3.19 Iznaola Spectrograph spectrograph, which comes from, I believe, the UA0777 nose cone on the DPA 4006. This accessory makes the microphone even less directional, but it also produces a rather resonant boost around 10kHz.34 I think that this boost highlights the brightness of Iznaola’s nails, which is not pleasing. The UA0777 nose cone is used to minimizes the high frequency roll-off of sound captured off-axis, which elicits further spaciousness, and the high ceiling produces reflections that are more delayed and longer decaying than those in VIDA’s recording of “Boisterous
Bourée.” Both of these factors make this recording sound more spacious, even though Bourée was treated with a Bricasti M7 (a very popular hardware reverb unit). With that in mind, the long-decaying reverberations do not reduce the clarity of Iznaola’s performance. John Taylor stated that he placed the mics close enough to the guitar so “that the church doesn’t dominate the overall sound.”35 Figure 3.20 is the stereo image for this recording, and it the image appears to be
34 DPA Microphones. n.d. 4006 User Manual . Accessed December 1, 2019. https://www2.spsc.tugraz.at/add_material/audiotechnik/manuals/50_Mikrofone/DPA/dpa_4006_manual.pdf. 35 Taylor, John, interview by Philip Logan. 2019. Email Questionnaire (June 25). 40 more diffused and wider than Verano Porteño (Figure 3.17), which uses the same microphones, as well as a similar microphone technique and placement.
Figure 3.20 Iznaola Stereo Image
Figure 3.21 Iznaola Waveform Statistics
The integrated loudness in Figure 3.21 is on the lower side (-22 LUFS), but there is less dynamic range (loudness range of 9 LU). This number may be influenced by the use of a multi- band compressor, but in any case, the recording’s loudness levels are dynamic enough. Listening to the album, I can say that the compression is inaudible. However, zooming into the waveform,
41 I was able to confirm its existence. The squaring off of the peak in Figure 3.22 is an effect of compression.
Figure 3.22 Iznaola Waveform Compression
Play
Artist: Jason Vieaux
Piece: Jongo
Composer: Paulo Bellinati
Key: A mixolydian
Length: 4:55
Jongo is the first track featured on Jason Vieaux’s Grammy-winning album, Play. The sound of the guitar on this album is quite different from the others, and that caused a negative
42 first impression. However, after listening to the album multiple times, I now appreciate the recording for its unique sound quality. The very fact that the sound of the guitar is unique may be one reason why this recording won a Grammy.
It is important to consider the effect a key signature may have on spectrograph measurements, such as an emphasis of notes (frequencies) specific to that key. Jongo has a mixolydian flavor to it, which highlights the seventh scale degree (G). Looking at the spectrograph in Figure 3.23, you should notice a peak at 98Hz and 196Hz (G2 and G3), as well as a stronger peak at 787Hz (G5), and this could be directly correlated to the mode of the piece.
However, the wide and strong dip at ~680Hz (F/F#) could be a factor of the guitar’s actual sound characteristic, especially since a similar reduction pervades the rest of the album—there is a sense of a hollowness to the sound of Vieaux’s guitar. With that said, the strong dip at ~680Hz, in my opinion, opens and smooths the sound, making the overall tonality of the instrument less punchy than I expect from a double-top design. The slight high-frequency bump around 20kHz is likely a product of the percussive section of the piece when the strings are struck against the
Figure 3.23 Vieaux Spectrograph
43
Figure 3.24 Vieaux Waveform Statistics metal frets of the guitar’s neck. Although a high-pass filter was used, there is still audible noise around 20Hz, which sounds like vehicles driving near the church.
Looking at the waveform statistics in Figure 3.24, we see a loudness range of 10.3 LU and an integrated loudness of 20.7 LUFS. Even though the track is louder than most of the other analyzed tracks in this chapter (second to Russell’s), there is still plenty of dynamic range, which, as mentioned before, can be correlated to the piece’s compositional nature, not just the player or recording process. What is especially interesting about the waveform statistics is the
+0.02dBTP on the right channel, and as you can see in Figure 3.24, this has caused 5 or more
“possibly clipped samples.” I do not believe that these distortions are audible, and 5 samples is an extremely small number when you consider the fact that that there are 44,100 samples every second of the recording. Also, I believe it happens in the percussive section, so any distortions caused by these clipped samples would likely blend in too well to be distinguishable. A simple solution would be to add a limiter, and that would have eliminated the digital clips.
44
Figure 3.25 Vieaux Stereo Image
The stereo image analyzer in Figure 3.25 measured a concentration of sound on the right side of the image, which means that this recording is unquestionably off-center. Of course, not all of the recordings I have analyzed have a perfectly centered sound, but in this recording, the right side of the image has taken precedence. I notice this unevenness change throughout the album—some tracks sound more centered, others do not. Maybe the microphones were moved at some point, or maybe Vieaux’s chair moved. I’m not really sure what caused this shift, but when listening with headphones, it is noticeable. As a last note, the percussive section is rather interesting because the short sounds allow you to hear the length and quality of the reverb in the church.
The Leaves Be Green
Artist: Vida Guitar Quartet
Piece: Simple Symphony, Op. 4: I. “Boisterous Bourrée”
Composer: Benjamin Britten
Key: A minor
Length: 3:24
45
Figure 3.26 VIDA Stereo Width
In this recording, four guitars circle an A/B microphone pair, making the stereo image wider than the other analyzed recordings, at least in terms of balanced loudness levels from the left-most point to the right-most point, which is shown in Figure 3.26. The angled dip at the center of the graph shows that none of the performers were positioned in the center of the stereo field. The waveform statistics in Figure 3.27 shows that this track, in comparison to the others included in this chapter, has a low dBTP level and integrated loudness, but it has one of the higher loudness ranges (13.1 LU). A limiter is used to control sound level peaks, but it is not used to the point of audibility. I cannot spot any compression of transients when zooming into the waveform at the recording’s loudest peak.
The spectrograph in figure 3.28 displays a mostly linear frequency curve, although there is an upward slope around 15kHz, which is likely from dithering. There is a strong roll-off that starts around 50Hz, which is likely due to a high-pass filter. Also, the ~100Hz range, which is around G2, is comparatively low, especially since there is an extended range guitar in this group.
The spectrograph shows very similar results for multiple tracks from this album, making it is safe to assume that this is consistent. This would mean that the room may not have a strong response
46
Figure 3.27 VIDA Waveform Statistics
Figure 3.28 VIDA Spectrograph to this frequency area, that the microphones or placement of microphones affected the capture of this frequency area, or that the guitars lack the ability to produce a lot of sound pressure in the low-frequency range. However, if the engineer thought that the bass response was truly lacking, equalization (a low-frequency boost) could have been applied. As a last note, there is audible bird song coming from outside the church, which I hear in several tracks throughout the album.
47 Guitar Recital
Artist: Xavier Jara
Piece: Elegy
Composer: Jeremy D. Collins
Key: E major
Length: 8:55
There are not enough words to describe the beauty of this recording. From the guitar to the interpretation to the microphones, everything about this is, in my opinion, artistic perfection.
Like Romanze, this recording was engineered by Norbert Kraft, but it is, in some ways, unlike the other recordings from this chapter. The composition is mostly in the key of E, although it does end a chromatic mediant above the original key (A-flat), and it is rather long and quiet.
This album was recorded in the same church as Romanze, and the sanctuary’s ~3 second reverb decay compliments this piece very well. I believe that the Hugo Cuvilliez guitar is lattice- braced, and the Sonodore microphones used in this recording are tube. Figure 3.29 shows the recording to be off-center, but my ears tell me that the source is centered—the heaviness of the right side of the image happens during the rasgueado section. The stereo image is relatively narrow (similar to Verano Porteño), but the venue’s size and reverberations still provide a natural openness in the sound. Personally, I find this stereo width more appealing; the sound is more unified.
Interestingly, Figure 13.30 shows that this recording has the lowest integrated loudness
(27.5 LUFS) and the most measurable dynamic range (17.7 LU). This composition is mostly quiet and lush, but the composer contrasts that with a very climactic (and loud) rasgueado section. It is a recipe for an exquisite musical moment that evokes a lot of emotion. The spectral
48
Figure 3.29 Jara Stereo Image
Figure 3.30 Jara Waveform Statistics graph in figure 3.31, there is a linear slope that only slightly emphasizes 6.5kHz and 10kHz (this may be due to the rasgueado section). However, the microphone’s frequency response graph, provided by Sondore’s website, shows a wide-band increase in response at 10kHz. There is quite a bit of subsonic sound pressure, which may be attributed to air moving through the room, and looking at the spectrograph, I presume that Norbert Kraft did not use a high-pass filter.
Overall, the balance is perfect, and the sound has a lot of clarity and pop—it is like a fog being lifted when compared to some of the other recordings analyzed in this chapter. On the
49
Figure 3.31 Jara Spectrograph
Sonodore website, Kraft is quoted saying that the MPM-81 microphones, used for this recording, are “superbly detailed… richly musical… without compression so often associated with tube mics of the past…,” and that they live in the “sonic family of the [Neumann] M149, [Neumann]
U47, or [AKG] C12.” He also says that these microphones are “the very best instruments in the sound engineers’ toolbox,”36 and after listening to this recording a number of times, I would have to agree.
Sonic Comparisons
I have picked the same musical selection from two separate albums—Cavatina, composed by Stanley Myers for the motion picture film The Deer Hunter—to display sonic comparisons. One track is from Vieaux’s album, Play, and the other is from Kraft’s album,
Guitar Favourites. I volume matched the recordings, and for emulation purposes, which will be discussed in Chapter Six, I also used the “EQ Match” feature in Izotope’s RX7 to quickly render similar frequency curves, influencing Norbert Kraft’s sonic architecture with Jason Vieaux’s.
36 Sonodore, http://www.rensheijnis.com/mpm81.htm, (accessed Novemeber 11, 2019) 50
a) Vieaux reference
b) Kraft level matched
c) Kraft EQ matched
Figure 3.32 Kraft to Vieaux Emulation
51 There are three images in Figure 3.32. From top to bottom, we have Vieaux’s reference recording (a), which is followed by Kraft’s level matched recording (b), then Kraft’s EQ matched recording (c). The result of the EQ Match feature is subtle, as these recordings already have sonic similarities, but it does rebalance Kraft’s frequency curve to some degree. This is not a full transformation of Kraft’s sound into Vieaux’s, but more of a compromise between the two.
If I were to manually perform this task with emulation in mind, I would make bigger EQ adjustments, boosting and cutting frequencies to bring out or subdue specific sonic qualities
(e.g., bass: ~100–200Hz; lower body (fullness): ~250–500Hz; upper body (boxiness): ~500Hz–
800Hz; punch: ~800Hz–1.5kHz; clarity: ~2–5kHz; brilliance: ~5kHz–10kHz; air and presence:
~10kHz and higher).
It is important to note that an increase of 3 decibels is typically noticeable (6 decibels is considered a doubling of sound pressure, and so 3 decibels is a 50 percent change). The changes to Kraft’s audio, although subtle and difficult to see (the x values of the spectrograph span ~140 decibels), reduce clarity and brilliance. Consequentially, this makes the guitar sound heavier, even with the noticeable reduction at 200Hz. Because of the subtleness, it is difficult to choose which is better, but I do think the change has a slight warming quality, which is nice for this piece.
Next, we have a side-by-side comparison of stylistically similar pieces, but these recordings have noticeably different sonic architectures (their frequency curves are not similar). I have level and EQ matched Rebolico from David Russell’s Aire Latino to Carora from Ricardo
Iznaola’s Heritage: The Guitar in Venezuela. Images in Figure 3.33 from top to bottom:
Iznaola’s reference recording (a), Russell’s level-matched recording (b), and Russell’s EQ- matched recording (c).
52
a) Iznaola reference
b) Russell level matched
c) Russell EQ matched
Figure 3.33 Russell to Iznaola Emulation
53 The changes between “b” and Figure “c” in Figure 3.33 are immediately noticeable.
There is a 1–2 decibel increase from 180Hz to 300Hz, a wide 1 decibel decrease from 400Hz to
4,000Hz, and a major increase (varying from 2–8 decibels) in frequency bands ranging from
5kHz to 20kHz, essentially leveling out the high end (e.g., there is a change of approximately 7–
8 decibels at 10kHz between “b” and “c”). The change in those higher frequency bands, because it is a major change, has actually made Russell’s recording sound more like Iznaola’s, which consequentially includes an obvious increase in the presence of nail-related articulation and noise. Essentially, the frequency curve of “c” is similar to “a.” Personally, I whole-heartedly love the change. My perception was that Russell’s recording was missing some of the brilliance and presence that I look for in a great guitar sound, and I think that is obvious when you look at the high-frequency dip present in the original spectral graph (“b”), prior to equalization. So, in this case, the EQ Match algorithm has not only created a similar tonal balance by increasing the brilliance and presence, but it has successfully emulated some important qualities found in
Iznaola’s recording that were not originally very present in Russell’s, such as the emphasis of nail-related articulation and tone.
As a last word, using the intelligent equalizer in Izotope RX is not always ideal. There are many things that contribute to a frequency curve, including non-musical noises or artifacts, and intelligent equalizers, as far as I am aware, are unable to distinguish between musical and non- musical sounds. The A.I. EQ may boost or decrease frequencies based on those non-musical, unwanted sounds, of which you should be aware. In these sonic comparisons, I used the software as a quick and easy way to introduce a concept, and since these recordings were mostly free of audible mechanical or other unwanted noises, it worked fine for my purposes. In Chapter Six, I use a combination of manual and A.I. techniques.
54 Conclusion
The waveform, spectral, and stereo width analysis show us that no recording is alike, and the differences among them are tied to each recording’s individual process. By documenting and analyzing a recording that you consider to be great, you can step into that process, and this will allow you to understand how a sound might be designed. In this chapter, I used the analysis alongside the documentation from Chapter Two to better understand the factors that produced a specific result. Once those factors are understood, I believe that reproducing a similar result is always possible. If the result is not as similar as you would like, there are steps that you can take to further replicate your target, which I exhibited with Izotope’s EQ Match. For more information regarding digital emulation read Chapter Six.
55 CHAPTER 4
THE STUDIO
With each passing year, advancements in audio technologies make owning high-quality recording instruments more accessible to consumers. This means that any person with a need or passion for recording sound can obtain great products at a relatively nominal entry fee; however, it should be noted that simply acquiring recording gear will not get you the results you desire.
Although experience with the equipment and how it works is the first step to producing a great sound, understanding acoustics and how sound moves in a room is an even more complicated, yet necessary, next step. This chapter explains what is necessary for a classical guitarist to create a well-suited home recording studio, which includes an overview of equipment and spaces, modal resonances, effective methods for manipulating your monitoring and recording space with acoustic treatments, and investment considerations for mobile and home studios.
The Anatomy of a Modern Digital Recording System
For any person to record an acoustic signal into a digital system, they need the following items: microphone(s), preamp, A/D (analog to digital) converter,37Capture device (e.g., a computer). The microphone uses a diaphragm, along with other internal parts, to create an electrical translation of physical sound waves, and the preamplifier amplifies that electrical signal. The A/D (analog to digital) converter turns electrical signal (voltage) into a digital format
(binary code). The capture device can simply be a computer with a hard drive, where the audio is captured via software, such as Logic Pro or Pro Tools. While you can find A/D and D/A converters as single-purpose units, many consumers and professionals alike, including myself,
37 Analog to Digital converters transform electrical signal to digital signal. 56 have avoided buying converters separately, and instead purchase an audio interface, which will typically include A/D and D/A converters, preamps, and more within the same device. Of course, with all these items you will need proper connection cables: XLR, TRS, USB, etc.
For monitoring recorded audio, here are the items you will need: D/A (digital to analog) converter,38 speaker and/or headphone amplifier, monitors (speakers and/or headphones).
The all-in-one audio interface should already have the output (digital to analog) converters, as well as an amplifier specific to headphones. Otherwise, these parts will need to be purchased separately. Amplifiers for passive studio monitors will need to be purchased separately, or you have the option of buying active monitors with amplifiers inside of each speaker enclosure.
Instead of purchasing all items separately, as was done in decades past, you have the option, as long as you already own a computer, of buying just three things: a microphone, audio interface, and a powered monitor.39 The audio interface connects to your computer via USB,
Firewire, Ethernet, PCIe, or Thunderbolt, and then sound is distributed to your monitors via an
XLR or 1/4” TRS connecter. This is an easy and relatively affordable route, and it’s one that even professionals use. Also, if you intend to be mobile, I recommend buying portable equipment, such as a laptop instead of a desktop computer, a small audio interface, a pair of open-back reference-quality headphones, and a pair of small diaphragm condenser microphones.
The Space
Generally, there are at least two spaces involved in the modern recording process: the live room (recording room) and the control room (monitoring room). The live room is the performance space, and the control room is for recording, editing, mixing, and mastering. For the
38 Digital to Analog converters transform digital signal to electrical signal. 39 Powered and active are the terms used for a monitor with a dedicated amplifier(s) inside of the speaker enclosure, whereas a passive or unpowered monitor requires an external amplifier. 57 live room, my recommendation is that you find a hall that is fitting for the instrument. Loud instruments may require a large room while softer instruments, such as the classical guitar, can be recorded in smaller spaces. Bright sounding spaces may compliment darker sounding instruments, as well as darker spaces for brighter instruments. You can read more about recording spaces, as well as microphones and their placement, in Chapter Five.
The room that you intend to use for monitoring your recording can vary in size, as long as you have acoustically treated the room well. However, it should be relatively isolated from external noise (this can be difficult if you are near a busy road, or if you have loud neighbors).
The main issue encountered by any person endeavoring to record an album is, first and foremost, locating a great sounding, isolated live room, but once that has been accomplished, those looking to edit, mix, and master their own recordings should put good effort into establishing a proper control room. The importance of the control room’s ability to produce a flat, uncolored sound cannot be overstated, and an acoustically treated control room can second as a live room for guitarists.
The Control Room
The control room for a classical guitar recording, would ideally have at least eight feet of length between parallel walls (square rooms are not recommended),40 and it would have an isolated room for loud machinery, such as desktop computers and amplifiers. If your control room doubles as a live room, it is even more important that you address noise pollution. The reason why you need a room with at least 8 feet between all surfaces is modal resonance. This phenomenon occurs when a wave’s length is equal to double the length of two parallel walls
(8x2=16 feet). Having at least sixteen feet will ensure that the fundamental modal resonance of
40 Having a square room means that you will have three fundamental modal resonances that are coincidental, which will combine to cause extreme resonances. 58 your control room occurs beneath the lowest note common to the guitar, which I am calling D2 at ~72Hz (wavelength of ~15.7 feet). This should eliminate issues with the room’s fundamental mode, which can be very problematic in a room intended for critical listening. The equation to find an apt room length works as follows: the speed of sound in room temperature (~1,130ft/s) divided by 70 (the arbitrary frequency I chose that is beneath 72Hz). The sum is ~16feet (8ftx2).
The extra 2 Hertz of frequency space may be enough to reduce issues with D2. However, having a room with an even lower fundamental mode would be beneficial.
Room Acoustics and Modal Resonances
The Master Handbook of Acoustics states that “the two ends of a closed pipe… can be likened to two opposing walls of a room.”41 The moment a sound source is produced in an enclosed room with surfaces that are parallel, three closed pipes are created. If the length, width, and height of that room are equal in value, such as a perfectly square room, that is the worst scenario for any space intended for sound, as it creates three identical pipes that will resonate to the same frequencies (coinciding resonances), which combine to create extreme modal resonances. A modal resonance, also called a standing wave, stationary wave, and eigentone, is where a room resonates at a specific pitch in the same way a pipe resonates to a pitch, and when one modal resonance is produced, each multiple of that frequency is also affected (e.g., a 50Hz resonance would affect 100Hz, 150Hz, 200Hz, and so on, which correlates to the harmonic sequence of that fundamental frequency).
Within each of these standing waves, there is a node and an anti-node. A node is the part of a sound wave with no displacement, meaning that no sound is present, whereas an anti-node is the point of maximum displacement, meaning that the sound level is at its maximum amplitude
41 Everest, F. Alton. Pohlman, Ken C. 2014. Master Handbook of Acoustics, Sixth Edition. Columbus, OH: McGraw-Hill Education. Page 230 59 (1 and -1 of a wave, also called the areas of compression and rarefaction). For example, if you produced a sine wave that was set to a room’s fundamental mode, you could find areas in that space where the signal would be completely inaudible (node), as well as points where it would be extremely loud (anti-node). To hear this for yourself, produce the correct fundamental mode relative to your wall’s length with a sine wave generator, then move around the room. What you will find is that the sound level is consistently loudest at the wall (anti-node) and inaudible at the center of the room (node). This is because half the soundwave fits perfectly between your walls, reflecting in a way that makes it appear to not move, which is why modal resonances are also known as standing waves. To find the fundamental room mode of parallel walls, the equation is
565/L (half the speed of sound at room temperature divided by the distance between parallel walls).42
Each set of parallel walls in a space will resonate to its own fundamental mode, and these are called axial modes. Axial modes require the most attention when acoustically modifying your space as they produce the loudest resonances. However, modes caused by four walls (front, back, and sides), which are called tangential modes, and modes formed by all six sides of the room
(addition of floor and ceiling), which are called oblique modes, should also be considered. These modes are less severe (-3dB for tangential modes, and -6dB for oblique modes), but they still affect the sound of your space. Also, they can further elevate axial modes if their frequencies coincide. To find your room’s modes, the room-mode equation first stated by [Lord] Rayleigh43 is as follows:
42 This is a simplified version of 1,130/2L. We can cut the speed of sound in half, and that cancels the need to multiply the length by two. 43 Everest, F. Alton. Pohlman, Ken C. 2014. Master Handbook of Acoustics, Sixth Edition. Columbus, OH: McGraw-Hill Education. Page 237 60
� � � � + + 2 � � �
c = speed of sound p, q, r = integers L, W, H = room length, width, and height
To find the first (fundamental) axial mode, enter 1 for p, and 0 for both q and r (1, 0, 0). To find the first tangential mode, enter 1 for p, 1 for q, and 0 for r (1, 1, 0). To find the first oblique mode, enter 1 for p, q, and r (1,1,1).
Table 4.1 is a mode calculation chart nearly identical to the one found in Master
Handbook of Acoustics, page 239. I believe that you should find the frequency of every axial mode up to their room’s Schroeder frequency,44 which is a good starting point for understanding the frequency response of your control room or recording space. Alton Everest states that
“…many timbral defects are traceable to axial modes.”45 However, if I were to calculate each modal resonance of my current control room (including tangential and oblique), I would do so with a computerized spreadsheet that already has the relevant equations programmed into its cells, then I could just fill in the room’s dimensions. The cells will compute all possible sound deficiencies you may have in your space. There are many companies and individuals that have these available online. There is a meticulously designed spreadsheet by John Brandt, an accomplished acoustician, which you can download for free at his website, https://www.jhbrandt.net.
44 The Schroeder frequency is the point where frequencies in a room transition from moving as sound waves to light rays (waveguide to ray), and rays are best controlled with diffusive materials. This transition is dependent on the volume and reverb time (RT) of the room, but a typical carpeted room in a house can be somewhere in the area of 250–300Hz. Frequencies below Schroeder are to be treated with absorptive materials. 45 Everest, F. Alton. Pohlman, Ken C. 2014. Master Handbook of Acoustics, Sixth Edition. Columbus, OH: McGraw-Hill Education. Page 260 61 Table 4.1 Mode Calculation Chart
Mode Number Integer Axial Tangential Oblique
1 1,0,0 x
2 0,1,0 x
3 1,1,0 x
4 0,0,1 x
5 1,0,1 x
6 0,1,1 x
7 2,0,0 x
8 1,1,1 x
9 0,2,0 x
10 2,1,0 x
11 1,2,0 x
12 2,0,1 x
13 0,2,1 x
14 2,1,1 x
15 1,2,1 x
16 2,2,0 x
17 3,0,0 x
18 0,0,2 x
19 3,1,0 x
20 0,3,0 x
62 Measurement Microphones and Room Response
Another way to calculate your room response is with a measurement microphone and measurement software. There are quite a few options out there, including free software like REW
(Room EQ Wizard). Measurement microphones have many price points, and many well-made measurement microphones, such as those produced by Earthworks and DPA, are also used by professional engineers to record instruments. These microphones can give you a very accurate analysis of your room’s response. The technique is simple:
1) Set a reference monitor in a corner of the space.
2) Set your measurement microphone in the opposite corner.
3) Play a sine wave that sweeps from 20Hz to 20,000Hz (human threshold of hearing), and
record that sound to your capture device.
4) In your DAW, place a frequency analyzer (most DAWs have them) in the signal chain and
send the recorded signal through the analyzer.
5) Study the frequency measurements of your end result.
If done correctly, you will see that there are areas in the frequency spectrum that are flat, and there are areas where there are significant peaks or nulls in the signal. The peaks and nulls in your room’s response can be severe, or if you have a good response, they can be less severe.
Frequency analyzers provide a visual representation of the affects modal resonances have on the sound in your room, and any peaks or nulls in the room’s response are concerning areas. Those imbalances should be tended to before your space can be properly utilized as a control room. In most cases, addressing a fundamental mode will reduce many other response degeneracies within your space, but subharmonic frequencies, which most fundamental modes are, can be difficult and expensive to address.
63 For software options, I use Sonarworks. Not only does it accurately measure and display your frequency response, but it also tunes the sound coming out of your monitors (calibrates), which combats room modes and speaker deficiencies. Essentially, a very accurate and automatically generated EQ is placed in the signal chain, affecting audio sources before they are transferred to your sound system, which consequentially reduces the effects of modal frequencies. This does not necessarily replace acoustic treatment, but it can instead assist a treated room. If you intend to use calibration software, you should also buy the measurement microphone used with that particular software.
Figure 4.1 is an image of my control space prior to applying correction software, and these measurements have been taken from my listening position. You can see a concerning resonance around 140Hz, which still exists after wide-band treatment has been added to the room
(absorption of this frequency would require a dedicated bass absorber, such as a membrane trap).
This response is decent for a bedroom turned into an office (12.5x12x10), and Sonarworks is able to effectively flatten out the response with zero artifacts. The fundamental mode of this room is around 48Hz, but my speakers are unable to effectively produce that pitch. This means that the audio produced by my speakers remains mostly unaffected by the fundamental mode.
Figure 4.1 Sonarworks Measurement
64 To hear an accurate representation of frequencies from 40–50Hz, I use open-back reference headphones. However, classical guitars cannot produce frequencies that low, so hearing them is mostly unnecessary, except to check for subharmonic noise (e.g., rumble from nearby traffic).
Treating Your Studio
Before you consider treating your studio, it is important to understand the qualities of a good acoustic space. According to Master Handbook of Acoustics, this is a perfectly diffuse sound field:
Even though unattainable, it is instructive to consider the characteristics of a diffuse
sound field. Randall and Ward have suggested the following ideas:46
• The frequency and spatial irregularities obtained from steady-state measurements
must be negligible.
• Beats in the decay characteristic must be negligible.
• Decays must be perfectly exponential (they will appear as straight lines on a
logarithmic scale).
• Reverberation time will be the same at all positions.
• The character of the decay will be essentially the same for all frequencies.
• The character of the decay will be independent of the directional characteristics of
the measurement microphone.47
In translation, a transparent, uncolored space is one that reacts to all frequencies equally.
46 Randall, K.E. and F.L. Ward. “Diffusion of Sound in Small Room,” Proc. Inst. Elect. Engs., 107B, pp. 439–450, Sep., 1960. 47 F. Alton Everest and Ken C. Pohlman. 2014. Master Handbook of Acoustics, Sixth Edition. Columbus, OH: McGraw-Hill Education. Page 127. 65 Acoustically treating a room entails two things: absorption and diffusion. Lower frequencies require absorption, and higher frequencies should be diffused. However, materials commonly used for the absorption of low (125–250Hz) and middle (250–2500Hz) frequencies are typically more efficient at absorbing higher frequencies (above ~2500Hz), which is not ideal if you plan to use the same space to record your guitar as too much absorption of higher frequencies can produce a dull and lifeless sound. For a control room dedicated to playback, high frequency reflections are not important. However, if you would like to use the same space for recording guitar, you should consider more advanced treatments.
If you intend to use your dedicated space solely as a control room, you can get away with absorbing high frequency reflections. All you need to hear is the sound directly from your monitors. For the guitarist with this in mind, I recommend building your own 2-foot by 4-foot broadband absorption panels. The job is quite easy, and it’s fairly inexpensive. You’ll need these materials for each panel:
1) Two 2’ 2” x 2” x ~1” and Two 4’ x 2” x ~1” pieces of wood
2) Wood glue
3) Nails and a hammer
4) Guilford of Maine acoustic fabric (my preferred fabric)
5) Staple gun
6) 4’ x 2’ x 2” acoustic mineral wool (Rockwool)
The process starts by gluing your wood pieces into a rectangular shape, then, after the glue has dried, hammer nails into the contact points to ensure that the wood pieces hold together. Next, cut a piece of fabric (I prefer Guilford of Maine because of its quality and balanced wide-band absorption) and use a staple gun to apply the fabric to what will be the front face of the acoustic
66 panel. Once you have the front facade, put the panel facedown and place mineral wool inside (I prefer Rockwool 80 for good bass absorption). If the mineral wool is snug enough, you’re done, but if not, I recommend applying another piece of fabric on the back to hold the insulation inside.
I advise that you make at least 6-8 broadband absorbers, then place them in key room positions. You need to attack early reflections first. These are the initial reflections that happen when a sound source is created in any environment. Consider your listening position: the earliest reflections will come from the wall directly behind your speakers, the area of the ceiling above your speakers, and the wall points to your left and right. Each spot should have at least one panel, although 2 might be an improvement. Panels can be hung directly on the wall, but you will be rewarded with enhanced middle- and low-frequency absorption if you hang the panels 3–4 inches off the wall. Once those panels are applied, you will want to consider placing a panel on the wall behind your listening position, and if you have larger panels dedicated to bass absorption, I would personally place those there.
That was the easy way to absorb frequencies ranging from approximately 125–20,000Hz.
However, the addition of too many broadband panels, due to their increased efficiency at absorbing high frequencies, will begin to deaden the spatiality and sparkle of the room, which is why you should consider other options if you intend to also record in this room. For me, I am fine with recording in my current living room, which is quite large, and has points where the ceiling is around 15 feet high. This room works well for guitar, and I have even recorded professional quality guitar recordings in that room without acoustic treatment. With that said, I do think a little diffusion or high-frequency absorption could have reduced time spent correcting the few harsh resonances that did exist around 2.5kHz, which is an area that is typically problematic.
67 In the case that you would like to use only low-band bass absorbers (i.e., the absorption will affect a smaller band of frequencies) and diffuse high frequencies, I recommend membrane absorbers, also called diaphragmatic absorbers, for bass absorption. They are panels with a vibrating membrane (also called a diaphragm), which is tuned to a specific range of frequencies.
The membrane will vibrate sympathetically to that particular area, absorbing that energy. Well- made membrane absorbers can currently be purchased from reputable companies for around
$200 per a 2-foot x 2-foot panel, not including shipping, and this method of frequency absorption does not require the frames to be very thick, which is great for adding bass traps to a small room.
Also, these do not replace broadband absorbers in early reflection zones, as those are still important (absorbers in early reflection zones prevent phase issues at your listening position, which improves the clarity and image of sound coming from your monitors).
The easiest way to diffuse sound, in my opinion, is to build panes with convex surfaces,48 also known as polycylindrical diffusers. If built deep enough, they can even double as low frequency absorbers, especially if you insert mineral wool or mineral fiber board, and polycylindrical membrane traps are also possible. However, one especially great way to diffuse high frequencies in a sound-field was developed by Manfred R. Schroeder, known as the quadratic residue diffuser (QDR). Schroeder diffusers are complicated to build, so you may be better off buying than building. However, buying can be costly. There are some 2-foot x 4-foot panels currently advertised online for around $200 each, but well-built quadratic diffusers, which can also double as bass absorbers, will likely start around $1000. These more expensive diffusers are large and made from solid wood. If you do intend to buy the materials and build from home, you should seriously consider finding or buying building plans from companies or acousticians
48 Concave surfaces concentrate sound. 68 that specialize in acoustic treatment. A well-diffused room will sound larger and more open, and if you are really on a tight budget, a large bookshelf filled with books and other items can substitute as a decent diffuser.
When utilizing your control room as a live room, I recommend using the “half-dead, half- live” method. This is a way for you to get the best of both worlds in one room, although I should say that having separate single-purpose rooms will likely provide better results. To do this, build broadband absorption panels and place them in your listening position’s early reflection zones.
At the back half of the room, you will apply diffusers and low-band bass absorbers. For recording, you work at the diffused side of the room (live), and for critical listening, you will be at the side if the room with wide-band absorbers (dead). This is my current configuration, and although I still prefer recording in the living room, which is much larger, the response in my control room allows me to make good recordings.
Listening Position
Your listening position should be away from walls because room modes are strongest at these points. Your monitors should be as far away from you as they are from each other, creating an equidistant triangle, and I prefer a distance of no more than 3 or 4 feet. Monitors should be at ear level, or at a downward angle of no more than 15 degrees. The monitor you purchase should come with setup instructions specific to that monitor’s design, but if not, there may be instructions on the company’s website. Typically, monitors are placed at least 1–2 feet from the front wall, but your monitor’s design may require more or less. If you are using headphones, this does not pertain to you.49
49 If you plan to primarily use headphones, I recommend that you use a pair that are calibrated. 69 Acquisition of Recording Gear
As mentioned previously, certain items are required to record audio. For the needs of most guitarists, your computer's processing requirements are quite low. Many relatively inexpensive desktop and laptop computers can fulfill your needs, so I will not write about that specific area. I am, however, going to provide information about every other aspect of the recording process related to recording the guitar. I want to give you two options to consider: the mobile option and the home studio option.
The Mobile Studio
The mobile studio starts by first having a laptop, although there are other options available, such as iPads and other tablets, as well as dedicated portable recorders. Once you have your laptop or other device, you need to install a DAW. If you have a Mac, I recommend Logic
Pro or Garage Band, but if not, there are tons of options out there, and many of them can be used with Mac OS, Windows, Linux, and even Ubuntu. Each DAW has its strengths and weaknesses, and they sell at a variety of price points. You may even be eligible for academic discounts, so keep that in mind. Here is a list of 9 popular DAWs:50
• Logic Pro (mac only, inexpensive)
• Pro Tools (expensive, most popular DAW)
• Cubase (great for composing and midi arrangement)
• Studio One (I have no experience with this DAW)
• Mixbus (geared towards mixing, but it is cheap and contains great editing features)
• Ableton Live (popular with DJs and electronic artists, great editing features)
50 This is not a comprehensive list of every DAW, and all opinions about their expense is based on current pricing. 70 • Audacity (free, but limited)
• FL Studio (primarily built for electronic artists and producers)
• Magix Sequoia (expensive, but incredible editing and processing features)
The next requirements for a mobile studio include a microphone, interface, and monitor.
I’ll start with the monitor. If you are a recording artist on the go, then I don’t think speakers are for you. My opinion is to use reference-quality headphones, maybe even two different kinds.
There are many brands of headphones, and they all have their unique sound architectures. My recommendation, especially if you plan to only use headphones—I have met classical recording engineers who exclusively use headphones—is to buy a pair that is pre-calibrated. Sonarworks sells pre-calibrated headphones with their software, and the price of these headphones with the necessary software currently ranges from $248–$474. Headphones used for monitoring should be open-back headphones, whereas tracking headphones should be closed.51
To continue with the mobile option, the all-in-one audio interface is ideal. It packs everything you need into a single box, including microphone preamps, A/D and D/A converters, and a headphone amp, and you can get some great quality, mobile audio interfaces for around
$500–$1,000. In no particular order, here are some popular brands that sell high-quality compact interfaces: Apogee, RME, Merging Technologies, Universal Audio, Motu, and Focusrite.
Apogee is Mac only, but most other interfaces will work with any operating system. Your choice should come down to the type of computer and operating system you are using, the types of features you require for your recordings, and, lastly, connectivity (e.g., Thunderbolt vs. Firewire
51 Open-back headphones allow sound to flow out, creating a cleaner, more precise and natural sounding reference. Closed-back headphones keep sound in the ear cup, which works well for monitoring audio while tracking, preventing headphone audio from bleeding into the recording microphone. Closed-ear headphones are also what most consumers use, and therefor act as a great mixing and mastering reference. 71 vs. USB). Also, you may consider auditioning interfaces at a local music retail store, or even take the time to watch a few of the many video reviews and comparisons available on the internet.
Last but not least in the list of gear for the mobile studio: the microphone. This tool, although mentioned last, is a top contender for most important piece of gear. A microphone can substantially alter the sound of a recording, and they come in many different flavors. For the mobile recording setup, I recommend a good pair of factory-matched small-diaphragm condenser microphones. Small-diaphragm microphones produce a sound that has a lot of clarity and detail, and they are very portable. To aid in your search for the right microphone, here are some common small-diaphragm cardiod microphones: Sennhesier e614, Rode NT5, Neumann KM
184, and Audio-Technica AT4041 (my personal favorite in price to performance category). I should note that Neumann KM 184 microphones are incredibly popular for recording guitar, but they also cost twice as much as the other microphones listed. If you are considering omnidirectional microphones, here are some options: Earthworks M23, Audio-Technica
AT4022, Sennheiser MKH-20 (used to record Play), and the very popular and expensive DPA
4006 (used to record Solo Piazzolla and Heritage: The Guitar in Venezuela)
The Home Studio Option
This option will have pieces of gear that are not as portable, and they may not work well for a classical guitarist. One thing to note is that you will need to have an effective control room, and if the control room is not ideal for recording, a separate live room. This option includes more monitoring options, more microphone options, as well as thoughts about external preamps and converters. This option is also costlier, but having a home studio with a good live room and control room, as well as a lot of quality hardware options (converters, preamps, microphones), will enable you to make professional-quality guitar recordings in your pajamas. You may also
72 find other musicians who want to record in your studio, and that makes your investment a potential source of income. Location recordings are always still possible (with a laptop or mobile recorder), but in my experience, musicians prefer recording in a comfortable, stress free environment. Also, location recordings can be expensive (more work for the engineer, and rental fees), so having a good home studio will be a low-cost option to potential clients.
Now, the first thing I would do at this level is buy pair of large-diaphragm microphones.
Compared to a small-diaphragm microphone, I’ve found that large-diaphragm microphones provide a warmer, full-bodied sound, which is a great option for guitar. Although I appreciate the clarity and detail of small-diaphragm microphones, guitarists for whom I have personally engineered recordings prefer the warmth of a large diaphragm over the detail and clarity of a small diaphragm. However, it depends on the player and the desired sound. With guitar, a microphone that is comparatively more sensitive to high frequencies can exaggerate nail-related noise and articulation, as well as string buzz. Also, the character of a large-diaphragm microphone makes the instruments sound larger and richer, which enhances the relatively quiet and intimate sound of the guitar. Neumann released a web article that says this about small and large diaphragm microphones:
There is no better and worse tool, both large and small diaphragm condenser
microphones are great recording tools. It’s all about choosing the right tool for the job.
Small-diaphragm condensers give you an uncolored, neutral, very detailed sound image.
Small-diaphragm microphones are ‘realists.’ Use them for anything that you want to
capture just like it is. Large-diaphragm condensers are part microphone, part instrument.
Their aim is to make the sound source appear bigger, more engaging, more beautiful and
adorable. They will give you that ‘sounds like a record’ feeling. Large-diaphragm
73 microphones are ‘romantics.’ Use them to put vocals and other lead instruments into the
spotlight.52
As far as which large-diaphragm microphone to choose, I can recommend most
Neumanns (TLM 103, U87, etc.), but they are typically expensive. I have thoroughly enjoyed the sound quality I get from the Audio Technica 4047SV microphone, which is round and smooth.
You might also consider an AKG C314 or C414, but there are endless options, which include lesser known microphone builders, such as Sonodore. In any case, you should audition or listen to samples of a microphone before buying. Also, consider ribbon microphones as a possibility. In my experience, ribbon microphones have incredibly pleasing and natural sound quality, and the
Coles 4038 has been one of the most popular ribbon microphones for decades (it was used for
Russell’s Aire Latino recordings).
After microphones are decided upon, consider your monitoring options. I personally use both headphones and speakers. Since we have already discussed headphones, I’ll say a few words about near-field monitors.53 There are, again, a number of options, and choosing the best one may require you to visit an audio equipment retailer. You should be able to audition a few to find the one that fits your taste.
As long as your control room has been acoustically treated, any well-made studio monitors with a response from 70–20,000Hz should get the job done. The human ear can typically adapt to minor timbral changes, such as those between different monitors with similar frequency ranges. However, your ears cannot correct for a monitor that does not produce enough
52 "What is the Difference Between Large and Small Diaphragm Microphones?” www.Neumann.com. Accessed December 07, 2018. https://www.neumann.com/homestudio/en/difference-between-large-and-small-diaphragm- microphones. 53 Near-field monitors are typically smaller than mid-filed or far-field monitors, which means that they are cheaper, meant to be placed nearer to the listening position, and have a reduced bass response. 74 bass, has poor high frequency detail and imaging, or lacks clarity. Apart from the lack of bass, these issues can be the result of one’s untreated listening environment, but they can also be attributed to cheaply or poorly made monitors. I use Genelec 8030a monitors, and I can confidently recommend these for guitarists. I have also used and can recommend JBL or Mackie, but within reason, a well-treated listening environment is, I believe, always more important than what monitor you use.
After the room, microphone, and monitoring solution, your next concern should be preamps. These devices exist to increase an audio signal. Preamps, like microphones, can give your recording a certain sound characteristic, also known as coloration, or it can be transparent.
For a preamp that will be used to boost a classical guitar signal, I prefer a transparent, uncolored sound, which is a common characteristic of transformerless solid-state preamps, but there are many choices, such as tube preamps, solid-state preamps with transformers, and hybrids that include solid-state and tube circuitry. In my opinion, it’s a small job, and the main difference between a cheap preamp and an expensive preamp is really the amount of noise introduced when boosting the audio signal. Good preamps are able to increase a signal very high without the introduction of mechanical noise or other distortions.
A/D and D/A converters are the last thing to consider. If you plan to buy an audio interface, converters are already included. However, having dedicated converters can possibly enhance your overall sound quality and stereo image. A good converter, without getting too technical, can produce a recording with more detail and clarity, as well as width and dynamic range, and the same goes for playback. It’s a simple conversion of binary code to or from an analog source, but there are a lot of other factors that can make quantifiable differences, such as
75 its sampling rate, quantization error,54 physical design, and quality of parts. With that said, buying audio converters separately can be expensive, and it should not be considered unless your budget accounts for it. I believe working at higher sample rates (88.2kHz or 96kHz) can produce a professional sound quality when working with inexpensive converters. You will, of course, down sample at some point before delivery (48kHz for web, and 44.1KHz for CD), but that is trivial.
Currently, I work with a RME UFX, which once sold for $2,300, but you can now buy them pre-owned for much less as new models (RME UFX II) were introduced around 2016. This device is the company’s flagship audio interface, which means it has their best converters, preamps, and more, all in one device. One handy trick that is unique to this device is its ability to record to a USB drive, such as a Lexar, in standalone mode. This means that I can take the device anywhere and record directly onto a thumb drive, so there is no need for a computer.
Some may consider this to be a liability as thumb drives are less reliable than computer drives, but I have been using the same 16GB drive for years with no issues. Although this option may be out of some budgets, the thumb drive feature is an example of considerations you may make before purchasing a recording interface.
Conclusion
There are many considerations to make when building your own recording system. What you decide should be dependent on your financial means. Once you have profited from this investment, you can then justify your need to upgrade equipment. When making any decisions regarding your home studio, question your actual needs. Do you intend to record on location or at home? If at home, should you invest in acoustic treatments? Would it be worth it to have an
54 Quantization errors are time related defects that add clicks and pops to the signal. You can also expect jitter, which is a subtle blurring effect. 76 audio interface that lets you record straight to a USB drive? Would it be wise to consider calibrating your monitoring system? How much of your budget should be spent on microphones?
Once you have made your decisions, search around for the best deals. I have purchased a lot of gear pre-owned, and there have never been any quality issues. I have even been lucky enough to find a pair of microphones at a quarter of their retail value, and they work and sound as good as new, even though they may be cosmetically imperfect. For a high-quality mobile option, expect to spend anywhere between $2,500 to $5,000. For the home studio, you may consider a budget of
$2,500–$10,000. If these figures are not possible, start your journey with cheaper equipment, or consider saving your money until you can afford the cost of quality gear.
77 CHAPTER 5
RECORDING AND PROCESSING
In the previous chapter, I discussed the concerns when establishing a home or mobile recording studio, such as recording devices, DAWs, room response, acoustic treatment, and more. This chapter relies on an understanding of that information, but it also contains information more relevant to the recording process, such as microphone placement and technique, the steps to be taken and considerations to be made before you record, as well as the post-recording process.
Knowing how to produce a certain sound architecture requires that you have knowledge of how a microphone and room, as well as other factors in the process, will affect your sound. In my opinion, having technical knowledge about the process of other recordings, which is covered in Chapter Two, will further guide you to a specific sonic quality. Knowing whether your favorite guitar recording was made in a room with wood or tile flooring, whether the microphones were placed near or far from the performer, and which microphones, guitars, and recording devices were used is all relevant information that can lead you to a favored sound.
Room Acoustics
A recorded sound is the combination of the direct source (the sound that is created by the performer) and the reflections created by the space (the return of that sound from a nearby surface). The Acoustical Materials Association says this about sound in a room:
The sound which one hears in any room, whether it be an auditorium, a class room, an
office, or a hospital corridor consists of two parts: (1) the sound which travels directly to
the ear from the point in the room where it originates, and (2) the sound which reaches
the ear after being reflected one or many times from the room surfaces. When sound
78 strikes the surface of any material, a part of its energy is absorbed, and the remainder is
reflected back into the room.55
The reflective and tonal characteristics of a room are determined by its size and building materials, as well as its shape. A large room will give sound waves more traveling distance before being reflected back, the shape will affect the sounds path through the room, the building materials, such as drywall, tile, fabrics, and wood, will have different absorption coefficients, and the density of room boundaries (thin or thick walls) will have different levels of transmission loss, which is the amount of sound allowed to pass through barriers. A combination of these factors, along with the room’s modal resonances, will determine a room’s sonic architecture.
Early reflections of a sound source (reflections that arrive less than 50ms after the source) help us perceive the room’s size, tonal quality, and reflectiveness, and first reflections (very early reflections that reach the ear less than 30ms after the source) are perceived by the brain as part of the original source, which is called temporal fusion. This means that the room’s tonal characteristics are directly imprinted on the sound of the instrument, and, depending on the room, that can translate to a more or less pleasing quality of sound. According to Modern
Recording Techniques: “temporal fusion is not absolute; rather, it depends on the sound’s envelope. Fusion breaks down at 4ms for transient clicks, whereas it can extend beyond 80ms for slowly evolving sounds.”56
Late reflections (those that happen after 50ms) are called reverberations, and for musicians, the quality and density of a room’s reverberations can be the most important factor in deciding whether or not we want to use that room for music performance. A reverberant room
55 Acoustical Materials Association. 1966. The Use of Architectural Materials: Theory and Practice. New York, NY: Acoustical Materials Association. Page 3 56 Huber, David Miles. Runstein, Robert E. 2010. Modern Recording Techniques. Burlington, MA: Elsevier Inc. Page 70 79 typically means that sound is allowed to reflect for two or more seconds before decaying 60 decibels (RT60). The longer a reflection takes to decay, the more high-frequency roll-off occurs in that reflected sound, and as a consequence, long reflections are much darker and warmer than the source. These warm reflections increase the LEV (listener envelopment) of the space, and a room with a strong LEV is what we look for in a good music hall.
Before you decide to use a room for recording, consider the intention of the space. Is the acoustic space intended for speech or music? Rooms intended for speech are built for clarity, whereas halls intended for music are built for warmth and spaciousness. The shape of the room can give you an idea about its intended use. Is the hall rectangular or splayed? Splayed halls bring audience members closer to the stage by extending seating longitudinally, and, according to the Master Handbook of Acoustics, these fan-shaped halls are “not used for music performance.”57 Rectangular shapes are common for music performance because they allow audience members to sit farther from the stage, so more reflections accumulate prior to reaching the listener’s ear. In either case, it’s important to realize that rooms intended for speech will be less reverberant, and they will be especially so in the lower and higher end of the frequency spectrum. Halls or rooms for speech tend to emphasize frequencies that make hearing speech consonants easier, such as from 1kHz to 4kHz. Music halls, on the other hand, are built to be more reverberant, and emphasis is in the lower end of the frequency spectrum. According to the
Master Handbook of Acoustics, “… music requires longer reverberation times at low frequencies.”58
57 Everest, F. Alton. Pohlman, Ken C. 2014. Master Handbook of Acoustics. Columbus, OH: McGraw-Hill Education. Page 490 58ibid., pages 486–487 80 Now, in general, if you are looking for a very reflective room, look for a space with mostly hard and smooth surfaces, such as finished wood, brick, concrete, and tile. If you want the room to be a bit brighter, you’ll want wooden floors and gypsum walls. For a balanced frequency response, consider a room with smooth tile or concrete, and for a darker room, find a space with materials that absorb high frequencies (e.g., carpet, thick drapes, upholstery). If you want a clear source with darker, less interfering reflections, find a hall with a wood stage and padded seats, or consider bringing in your own absorbent panels, such as the ones discussed in the Chapter Four. Personally, I enjoy working in a medium sized hall (50,000–100,000 cubic feet) with wooden floors and padded seats, especially for guitar. My preference is a clear and open source with reverberations that are dark and subdued, as this allows plenty of flexibility in post-production to manipulate the sound.
If the desired space is one that sounds isolated and uncolored, you need a room filled with absorptive materials (the soft fluffy stuff, as I have heard it called). The most common absorber in any room is carpet, and the addition of thick padded seats will help in this endeavor.
Placing the guitar in the center of a large, heavy-carpeted space can produce a dry sound, 59 especially if the microphones are placed close to the source. Using an isolation booth is the far extreme. I also find that the guitar can sound relatively dry in any large music hall with close directional miking (e.g., cardiod polar pattern), especially if the hall is suited for an orchestra, and greatly so if there is an audience in attendance—a human body wearing thick clothing is extremely sound absorbent. Recording the guitar with these variables can produce an unadulterated or mostly unadulterated guitar sound.
59 Dry refers to a sound that is not reverberant or spacious. 81 If the space you intend to use for your live room is in your own home, then you have the ability to manipulate that space as needed, which can be advantageous. Having a relatively large room with high ceilings is a plus, and floors with wood, tile, or some other reflective material can produce a lively and reverberant sound. If the room you want to use is not already a great sounding space, then measure the response with techniques from the previous chapter and decide how you can make it better (i.e., more diffuse, warmer or colder). Do you need bass absorption or high frequency diffusion? Should you consider pulling up your carpet or moving out furniture? The character of a space can come from its imperfections, and creating a space that sounds good is, in my opinion, more important than creating a space that has a perfectly flat response.
For more information about the sound characteristic of materials, consider searching for and reading Sound Absorption Coefficients of Architectural Acoustical Materials (1953) and The
Use of Architectural Acoustical Materials (1965). Both sources are written by The Acoustical
Materials Association, and both journals are public domain. The first source provides the absorption coefficients of many common building materials, as well as many materials used for acoustic treatment. For example, based on their tests, wood has an absorptive coefficient of .15 at
125Hz, .11 at 250Hz, .10 at 500Hz, .07 at 1000 Hz, .06 at 2000 Hz, and .07 at 4000 Hz. This means that wood is more absorptive at lower frequencies than higher frequencies, although it is not very absorptive in general, which is why it is very reflective.
Microphones Types
Microphones and their placement can significantly affect the quality and overall sonic architecture of a recording. In my own list of the most important components in a great recording, first is the player, second is the guitar, third is the acoustic space, and fourth is the
82 microphone and its placement, then comes the preamps, converters, and so on. If the desired space is an isolation booth or room that has very little effect on the guitar’s sound, then the microphone is going to be more sonically important than the recording space.
The type of microphone you use is also important to the resulting sound. Essentially, a microphone’s type is based on the way the microphone converts acoustic energy into electrical energy. Popular microphone types are dynamic, condenser, and ribbon, and each will produce a different sonic quality. Dynamic (moving coil) microphones, like a Shure SM58, convert sound energy into electric energy by displacing a suspended metal coil that wraps around a magnet, which happens when sound energy moves the diaphragm to which the coil is connected. The dynamic microphone has a stiff diaphragm, and it is therefore able to handle high sound pressure levels. However, the stiffness of the diaphragm causes these microphone types to be less capable than others at faithfully reproducing transients and high frequencies. This system of energy conversion is called electromagnetic induction, and ribbon microphones use a similar system, which is why they are considered a type of dynamic microphone.
Ribbon microphones are built on the theory of electromagnetic induction, but the design is different from the Shure SM58. A grooved (corrugated) aluminum diaphragm is suspended in a field of magnetic flux, and when the diaphragm moves, an electric current is induced. Because of the diaphragm’s extreme thinness (approximately 2 microns),60 ribbon microphones are better than any other microphone type at reproducing transients, and their very focused directionality makes off-axis sounds nearly inaudible, which is why they were once extremely popular on movie sets. However, they produce a very weak output signal that requires a lot of pre- amplification. It is also important to note that, due to the thinness of their diaphragms, extreme
60 Huber, David Miles. Runstein, Robert E. 2010. Modern Recording Techniques. Burlington, MA: Elsevier Inc. Page 113 83 care should be taken when operating or traveling with older ribbon microphones, although newly designed ribbons are made with more robust diaphragms.
Condenser microphones, which are the most popular microphones today, convert sound energy to electric energy by way of an electrostatic charge. Two capsules, one being a thin diaphragm and the other being an immovable plate, form a capacitor, also known as a condenser.
When the microphone is powered, a charge is created between those two plates. When the diaphragm reacts to sound pressure, the front plate moves inward and outward in relation to the immovable backplate, increasing and decreasing the capacitance, and inversely changing the voltage. The changes in voltage are converted to an output signal by way of a resistor, and the output signal, because of its low energy level, must be pre-amplified internally, which is why these microphones require phantom power.61 Typically, condensers produce a bright and transient sound, making them good all-around microphones for most situations.
Another microphone type to be discussed are those with internal tube circuitry. Tube microphones are both famous and infamous for their ability to add a certain “color” to the signal.
I do not have much experience with tube microphones, but I have lots of experience with tube amplifiers and preamplifiers and can certainly attest to the color and midrange liveliness that they infuse into the signal. Tubes, when overdriven, will compress and harmonically saturate an audio signal, which can be a very pleasing distortion. Cheap tube microphones, as I have read in many articles and forums, can be notorious for their harshness and noise, but more expensive ones, can be quite clean and tonally rich. In any case, using an external tube preamp with a typical transistor-based condenser microphone might produce similar results.
61 48 volts sent to a microphone via the XLR cable. 84 Each of these microphones, because of design differences, will produce a different sound architecture. Some microphones are designed to be smooth and warm, and others are bright and transient. The method of sound conversion, the thickness and pliability of the diaphragm, and the type of preamplification will all make a difference. With that said, I do not think it is necessary for a you to know the technical aspects of every microphone. You simply need to consider the sound you want, then find a microphone that captures that for you. The best practice is to audition microphones within your price range, watch online video reviews, listen to audio samples, and maybe research the microphones used for some of your favorite recordings.
Generally, dynamic microphones, because of their thick and stiff diaphragms, have a more rugged feel and sound quality. They can be less responsive to high frequencies and transients, producing a warm sound that does not have a lot of natural transients and punch.
Ribbon microphones are known for having a warm sound with a very human-like high-frequency roll-off, and they are also extremely responsive to transients. Ribbon microphones produce a sound that is lively and warm. Condenser microphones are the most versatile. They are designed with quite varied diaphragm thicknesses and sizes, and because of those variations, there are many different sound characteristics associated with these microphones. There are both large- and small-diaphragm condensers. Typically, large diaphragms deliver a warm and smooth response in relation to their small-diaphragm counterparts, which are brighter and more detailed.
Microphone Technique
The polar pattern of a microphone can also make a significant difference. If the polar pattern of the microphone is extremely directional, like a super-cardiod or hyper-cardiod pattern, the microphone will not capture as many room reflections, such as those coming from the back and sides of the acoustic space. However, microphones with omnidirectional polar patterns can
85 record sound from all angles, and that will translate to a more spacious sound image.
Bidirectional polar patterns can act as a middle ground between the sound architecture of the other two polar patterns, or two can be set up as a Blumlein pair, which has a 360-degree pickup pattern. If recording two performers, a bidirectional microphone can be placed between them to record a mono track of two different sound sources with relatively good isolation. Also, bidirectional microphones are required for mid-side recording, which is a popular, although slightly advanced, miking technique.
Which stereo technique you use should be based on the sound you desire. There are three popular techniques used for classical guitar. The most common is the A/B spaced pair. This technique commonly utilizes omnidirectional microphones, and depending on the distance between microphones, this technique can create a narrow or wide stereo image. A/B is a near- coincidental technique, which means that you need to place the microphone’s diaphragms within close proximity (1–2 feet apart), and they should be equidistant from the source. However, if you intend to place the microphones farther apart, meaning that they are no longer near-coincidental, you should consider adhering to the 3:1 rule.
Modern Recording Techniques states that the 3:1 distance rule is used to “reduce [sound] leakage and maintain phase integrity…” and that “… for every unit of distance between a mic and its source, a nearby mic (or mics) should be separated by at least three times that distance.”62
The book also goes on to state that some “err the on the side of caution… by following a 5:1 distance rule.”63 This means that if microphone B is 1 foot from its source, microphone A should be 3 feet from microphone B (three times the distance). If the distance of microphone B
62 Huber, David Miles. Runstein, Robert E. 2010. Modern Recording Techniques. Burlington, MA: Elsevier Inc. page 137 63 ibid. page 137 86 increases to 2 feet from its source, the distance of microphone A from microphone B should increase to 6 feet. This rule is especially useful when recording multiple sources, and some engineers will even place gobos between the sources and/or microphones to further reduce sound leakage. With that said, the 3:1 rule is more of a guideline than an actual rule, and you may find that it is contingent on the microphone’s polar pattern. For example, the increased directionality of microphones with cardiod polar patterns may allow more flexibility than an omnidirectional polar pattern.
The next common technique is X/Y, which uses two small-diaphragm microphones with cardiod polar patterns. In this technique, each microphone is angled 45-degrees inward to create a 90-degree angle sum (right angle). This technique creates a relatively narrow image, but it retains perfect mono compatibility. Variations of this technique include the Blumlein pair, which utilizes bidirectional microphones instead of cardiod microphones. Both techniques are considered coincidental because their diaphragms overlap, so soundwaves reach the microphone at the same exact time and place.
The last popular technique is M/S (mid-side). A cardiod microphone is pointed directly at the source, and a sideways facing bidirectional microphone is placed coincidentally (diaphragms overlap). The cardiod microphone is the mono component (mid), and the bidirectional microphone is the stereo component (side). Since a single microphone is used to capture the stereo component (left and right sides of the stereo image), you will need to separate that source into two signals. I recommend watching a video or reading an article prior to attempting this technique. It’s not incredibly difficult to learn and utilize, and it has the benefit of producing a very wide image that is perfectly mono compatible. Essentially, the signal from the bidirectional microphone is doubled, and those two signals are positioned out of phase with each other. The
87 two signals are then panned to opposite sides of the stereo field (left and right). This allows you to hear the left and right sides of the acoustic space as two distinct signals, even though they were originally captured by one microphone.
Microphone Placement
The placement of your microphone is also extremely effective in changing the sonic quality of your recording. Placing a microphone close will capture more of the source and less of the room, and placing it further away will capture more of the room (more of the reflections).
Another consideration is the tonality of the instrument. Angling a microphone at the bridge of the instrument can increase body and fullness, and angling it more to the neck can increase bass and treble. When placing microphones close to the source, you should be cautious. Proximity Effect is a low-frequency emphasis that occurs when a directional microphone (cardiod or bidirectional) is placed too close.
If you imagine your microphones as a set of ears, you can start to better determine your sound image. Consider two guitarists playing a duet side-by-side. If you placed a set of closely spaced stereo microphones between the players, the left guitarist will be on the far-left side of the recorded image, and the right guitarist will be on the far-right side of the image. As you move those closely spaced microphones farther away, the sound of the two players narrows. Moving the microphones apart will increase stereo width, but it will also further concentrate the sound from those two guitarists into a more unified source. At this point, the way to increase the perceived distance between players would be to spread them apart.
Recording Preparation
The best practice, especially when recording on location, is to be prepared, and possibly over prepared. Before traveling, you should have the following items:
88 1. A sound capture device, such as a portable recorder or laptop computer with recording
software
2. A microphone preamp(s)
3. A/D and D/A converter
4. Headphones and/or speakers with necessary amplification
5. A microphone (two paired microphones for stereo capture)
6. All cabling, which includes XLR cables for microphones. You may need extra XLR or TRS
cables if using speakers for monitoring on location. Also, if your speakers require an external
power amplifier, you will need speaker cables or wires, which are different from standard
XLR and TRS cables
7. An appropriate number of microphone stands and shock mounts
8. Consider a stereo bar for A/B or X/Y setups, because they are incredibly convenient
The simplest method would be to use a portable recorder and one microphone. However, for the sake of flexibility, I recommend using a laptop for location recordings, an audio interface with two preamps and a headphone amplifier, closed-back headphones for monitoring, and two microphones that are either factory matched or the same model. Having different mics can work, but you should understand how those two microphones will blend.
Once you have accounted for your needs, decide where you will record. If deciding on a location other than your home, you must consider rental fees, isolation from exterior noise (e.g., traffic), interior noise (air conditioners and fans should be turned off), whether or not recording equipment can stay in place over consecutive days, and whether or not you can make minor alterations to the space, such as moving around furniture. Of course, creating a recording space
89 in your home will allow the most convenience, flexibility, and comfort, but you may already know of a building with great acoustics that you would like to use.
When setting up a recording session on your device or DAW, record at the highest bit rate possible (likely 24), and also consider recording at a sample rate of at least double your intended delivery rate (88.2kHz for a 44.1kHz final product). Increased bit rates give you more dynamic range and resolution, and increased sample rates, in my opinion, produce a more detailed recording with extended depth and spaciousness. However, the benefits of higher sample rates are controversial, and many feel high sample rates are unnecessary with modern converters.
Older converters created unwanted artifacts when filtering out frequencies above Nyquist’s sampling theorem,64 so increasing the sample rate meant that the filtering of frequencies, and the creation of artifacts, happened higher in the spectrum, above what was audible.
I personally use a sample rate of either 88.2kHz and 96kHz because digital plugins perform better at these rates, although they also require more processing power from the computer. Also, many third-party plugins, even some stock DAW plugins, are able to oversample, so they can work at two or four times your sample rate, giving you the benefits of higher rates on select plugins while recording and working in lower rates. I highly advise higher sample rates or oversampling for plugins that add harmonics to your sound, which will avoid the aliasing that occurs when those plugins creates harmonic content above your digital systems available frequency range (22kHz when working at a sample rate of 44.1kHz). When a plugin creates harmonics above your available digital limitation, those harmonics will fold back into the audible spectrum, creating noise and unwanted artifacts. Typically, analog modeled compressors and equalizers can and will do this, as well as any type of harmonic saturator or distortion effect.
64 The sample rate must be two times the audible frequency range. 20,000Hz multiplied by 2 is 40,000. 44.1kHz became the standard when the compact disc originated. 90 Recording
When you arrive at your destination, find the spot in the room where you want to setup your microphone(s). It’s easier to do this if someone is there to play for you. To hear the timing and length of reverberations, clap or make some other percussive sound. To hear frequency response, you could loudly vocalize a sweeping pitch from low to very high, or even short chromatic pitches, actively listening for any frequency that is drastically more resonant than others. Once you have located a good spot, sit in front of the mics and do some test recordings, moving your chair or the microphones a few feet each time to decide which placement makes your guitar sound best.
Once you find your placement, you may begin the recording session. However, if you are using a laptop, it’s important to ensure that the laptop’s cooling fans do not run loudly during the recording session, especially if the laptop is near your microphones. Many of today’s basic laptops are powerful enough to record audio without activating your computer’s cooling system, but you should find out prior to recording. If they are loud, you will need to move the laptop away, positioning it out of the microphones pickup range (if you must, move it to another room, but make sure your cables are long enough to reach). If you are then worried about turning your
DAW’s record button off and on (i.e., the laptop is too far away), consider letting the recorder run the entire session, then clap or make a loud sound between each take. Otherwise, you have the option of connecting a device, such as a phone or tablet, to wirelessly control your DAW, if your DAW has that feature available.65
You may want to plug in-ear or closed-back headphones into a metronome, as that will help you keep your tempos consistent on each take. If you do this, record a test take to ensure
65 Wireless connections may require a router. 91 that the click from the metronome is not bleeding into the microphone(s). You may also want your separate takes to be perfectly synchronized, even during sections of musical acceleration and deceleration (i.e., you have very specific performance tempos). To do this, consider creating a custom click track prior to the recording session, then either listen to the click via your DAW, or export the click and put it on your phone or other handheld device. Using a click track may make the performance a little robotic, but the editing process will be much easier—it can be difficult to composite multiple takes when tempos are not consistent.
Editing
Editing can be easy or difficult, but it depends on your knowledge of the DAW and its editing features, whether or not you know what to listen for when editing, and whether or not you know what tools to use. Recording a professional album usually requires a lot of editing, and compositing audio can be slow, meticulous, and rather boring. You should take breaks, and you should not edit an entire album in one day. With that said, performers who consistently play well, and who do so at consistent tempos, make the editing process far less cumbersome, as there will be fewer edits that all elide seamlessly.
The main editing tools with which you should be concerned are the cut (or splice) tool, which may have a scissor or blade icon (or whatever else the programmers of the software decided), and the crossfade tool. Essentially, you cut out the pieces you want to join and position them together while being very attentive to the rhythmic pulse of the performance, and then, if necessary, use the crossfade tool to ensure that the transition is smooth. There should be no audible pops, clicks, double attacks of a single note, or extreme dissimilarities (differences in volume, tone quality, tempo, etc.). If there are noticeable differences, you may need to consider automation.
92 Automation is a way of programming your DAW to make adjustments at specific points in the audio timeline, such as moving volume faders, turning on and off equalizers, as well as other useful tricks. However, if the take you are trying to transition to has extreme tone quality, tuning, or tempo inconsistencies, automation may not help. Subtle differences, like slight volume changes, can be easily fixed with automation. If you do have to automate, which should not be often, ensure that adjustments are inaudible. If you cannot fix the issue with automation or the use of another take, you will likely need to rerecord that section of the music.
Mastering
Mastering was once the transfer of a master recording to a distributable format, such as from the master tape to vinyl. Now, the goal is to increase quality and cohesion of that recording(s), which is sometimes referred to as “sweetening,” as well as to make it compatible with a myriad of distribution services and consumer playback devices. The mastering process can include phase correction, noise reduction, equalization, compression, additional reverb, stereo width adjustments, and limiting/maximizing. Some say that no processing is needed in a classical recording, and if your recording is good enough, you may not need to do anything.
However, should you decide to process your recordings in any way, this section describes the post-processing workflow that I typically use when working with a classical music recording, and I hope that the discussion is able to help you find your own workflow.
Phase Alignment
If your recording suffers from phase misalignments that make it incompatible with monophonic playback systems (i.e., the audio is soft and the tone quality sounds somewhat degraded), you should consider taking corrective measures. When working in your DAW, I advise that you check phase relationships with a phase-correlation meter before you call the
93 recording(s) finished. Phase-correlation meters visualize your phase relationship, and for most correlation meters that I have used, this is the user interface: “1” (or a green indicator) signifies an in-phase monophonic signal; a signal that measures between “0” and “1” is considered a phase-aligned stereo image; “-1” (or a red indicator) signifies an out of phase monophonic signal, and a signal that measures between “0” and “-1” is likely a stereo signal that is moving in and out of phase.
Noise Reduction
Noise reduction should be used prior to any audio manipulation tool. Effects like compression and reverb can increase the audibility and sustain of unwanted noise. However, what sounds are considered noise can sometimes be subjective. Typically, the low-end rumble from an air conditioner would be considered noise, but the clicks from nails moving across the strings of a guitar might not be considered noise. In either case, both can be processed in a way that makes those noises mostly inaudible. For general room noises, like the wash of white noise that comes from moving air or the soft hum of nearby electronics, an average noise reduction plugin can work, although I use Izotope RX in standalone mode because of the high-quality result. When using a noise reduction plugin, simply find a section in the audio file where only the room noise is audible (nothing else), highlight that section and use the “learn" function to map out contour of that noise pattern, then set the threshold and quality of the reduction, as well as the number of decibels that noise should be reduced. How this works is that any sound above the threshold is untouched, and any sound below is reduced as many decibels as you choose. It’s important to check your reduction in the softer sections of the performance to ensure that the music remains unaffected. Sub-harmonic noise can also be reduced with a simple high pass filter set to around 50Hz.
94 Next, consider accidental noises, such as nail clicks, string buzz, and any other unwanted sound that may not have been produced intentionally. With this type of reduction, I sometimes use what is called spectral repair. The analogy someone once used to explain this technique to me was that spectral repair removes noises in the same way photo editing software might remove a skin blemish. The software program can find a similar sample of audio to replace the unwanted sample that you have highlighted, just like a photo editor can find a similar fragment of skin to replace a highlighted blemish. However, I typically use spectral repair in manual mode, and I attenuate most unwanted sounds to inaudible levels instead of replacing them—replacements should be handled in the editing phase, if possible.
Although noise reduction plugins are very common, spectral repair is less so, and unless you are using an expensive software, such as Sequoia by Magix or Pyramix by Merging
Technologies, you will probably need to buy a third-party plugin or specialized software. Once you have your plugin or software ready, simply find the perpetrator, highlight it, set the strength of attenuation, then let the software do the rest. Most of the time this works well; however, noises that happen at the exact instance the player plucks a string, such as a “string buzz,” are difficult, if not impossible, to remove. This is because string buzzes are many times missed notes. The guitarist produced a buzz instead of a note, which is a left-hand mistake. Removing the buzz creates an instance of silence. To effectively remove a buzz, manually replace that note with one from a repeating section of the music, or even another take. Right-hand nail clicks usually happen during preparation, just before the note is struck, so those are typically easy to attenuate. Foot taps and other short noises, depending on their timing, can also be trivial to remove.
95 Equalization
Equalization originated as a noise-reduction tool. According to Andre Millard, the writer of America on Record: A History of Recorded Sound, “Engineers at Bell Labs began to attenuate the strength of signals at chosen frequencies, basically adjusting volume of different sounds by increasing or decreasing the amplitude of chosen bands of frequencies.”66 Today, equalizers can be used as noise reducers, but I generally use them to as a tool for reducing harsh and overly resonant frequencies, as well as for tonal rebalancing. Once I have edited and noise reduced my files, I put them into a new session timeline inside my preferred DAW, then I begin what I consider to be the main part of the mastering process with equalization.
The first equalizer, which I place on the stereo channel (affecting both left and right channels identically), is for surgical removal of overly resonant frequencies. I position EQ points at the offending frequencies and reduce them when they occur. Usually, I do this with automation (reduce the frequency at the occurrence of the unwanted resonance, then returning the EQ point to its inactive position), but you could set the equalizer and leave it, which would save time and possibly create similar results. Whichever way you chose, just ensure that you are not losing any important frequency information in the process. Surgical EQ is always subtractive, and you will generally want to use a small bandwidth (also called “Q,” which is short for quality factor) for this step.
Once I am done with subtractive equalization, I move to the master bus. Here, I begin adjusting the overall sonic architecture of the recording with a tone shaping equalizer. If I feel the recording(s) has too much high-end and not enough body or bass, I will roll-off the high frequencies with a high-shelf or low-pass filter starting around 3–4kHz and raise the bass with a
66 Millard, Andre. 2005. America on Record: A History of Recorded Sound. Second Edition. page 277 96 low-shelf or wide bell filter around 130–180Hz. If needed, I reduce unwanted noise below what is within the guitar’s response with a low shelf, which I will set at around 50Hz, and I turn to linear-phase equalizers if I feel the need to alter the left and right channels separately.67 Some
DAWs have built-in linear-phase equalizers that allow separate treatment of left and right channels, others do not. If your DAW does not allow the left and right channels to be independently altered with linear-phase EQ, consider finding a third-party linear-phase equalizer that does. Equalizers inherently alter the natural phase of the audio track, but linear-phase equalizers ensure that the phase remains unchanged.
Compression
Compression, like equalization, was developed for the film industry, and they used it because “soundmen were faced with the problem of recording a volume range of 60 to 80 decibels on the sound stage with a medium that could accommodate a range of only 40 to 50 decibels.”68 Today, however, compressors are used for the certain sonic quality they add, as well as for reducing dynamic range. Many classical musicians I have known tend to scoff at or deny the need for compression, but I find that it can be useful. The issue is that those who are not familiar with the effects of compression tend to not hear what is happening to the source, and nascent sound engineers who are also not familiar with the effect may over use or incorrectly set the compressor.
There are many types of compressors available today, such as optical, solid-state, and tube, and each has its own purpose and character. A compressor can be used to smooth the
67 Linear-phase equalizers allow you to separately alter left and right channels without disrupting phase coherencies. This means that you can reduce the brightness of the right channel and remove an unwanted resonance in the left channel without affecting your sound image or mono compatibility. This might be necessary if you used unmatched microphones as a stereo pair, but you would like for them to sound more cohesive. 68 Millard, Andre. 2005. America on Record: A History of Recorded Sound. Second Edition. page 277 97 dynamic level of a performance, which is also known as “glue,” and that is a great way to make a recording session of one piece with 20 or more edits sound more cohesive. You could even use it to make an entire album sound more unified. Compression can also be used to add harmonic coloration, increase transients to add more attack, or reduce transients to reduce attack, and all of these techniques can be used in a way that transparently improves the recording.
With that said, my process is to find the loudest transient in the entire recording—if I’m mastering multiple recordings, I may use the loudest transient of all the recordings—set the compressor to a 2:1 ratio. Then I adjust the attack to 30–50 milliseconds, adjust the release to around 0.5 to 1 second (dependent on tempo), and move the threshold down until the loudest transient is being reduced by no more than approximately 2–3 decibels. This provides transparent dynamic control, and analog compressors, as well as analog modeled digital compressors, can also add some harmonic content to the sound (what one might call “coloration,” “mojo,” or
“juice”) which is a great way to add character to an overly clean and neutral recording.
Reverb
If I have captured a good room sound, I may not use any reverb. However, I do add some amount of reverb around 90 percent of the time, and that is usually to add an elongated reverb tail.69 The elongated tail (late reflection) will make the space sound bigger and more enveloping, and it can make the heavily edited performance sound more blended (i.e., glued). If the music is fast and articulate, extra late reflections may not be ideal, but as the engineer of your own record, you can decide what is best. Many reverb plugins have low- and high-pass features, which I use to keep late reflections from affecting frequencies above ~6,000Hz and frequencies below
~250Hz. The 250Hz high-pass keeps the reverb from getting too muddy (sustain of low
69 Tail refers to the reverbs sustain. 98 frequencies can cause unwanted overlap of harmonically fundamental tones), and the 6,000Hz low pass keeps the reverb from affecting unwanted high-frequency noise, such as nail clicks and string buzz. You may also want to use pre-delay70 to give those additional reflections more time before their occurrence, which will help you retain clarity of the source.
Limiting/Maximizing
Last in the mastering process is maximizing, which is a brickwall limiter paired with a gain feature. The job at this point is to bring volume levels up or down to nationally or globally set standards, which is dependent on your medium. Each digital retailer and streaming service has a standard volume level, measured in LUFS and dBTP, and you should do a little research to find out what levels you need for the digital service you intend to use. If you do not do this, the site may convert your file for you, which can potentially degrade the quality of your audio.
Recent internet publishings show that Amazon and Spotify require an integrated loudness of -14
LUFS and a peak level of -1.0 dBTP (this peak level limit is enforced to avoid inter-sample peaks when digital audio is converted to analog audio), which is quite loud for classical music. If your audio is over that set standard, the streaming services normalization71 algorithm will reduce it, and if it is below, the algorithm may or may not increase your LUFS and dBTP. Hopefully the limiter/maximizer you are using shows audio levels in LUFS and dBTP, but if not, you will need to add a loudness meter at the end of your mastering chain so that you correctly view loudness levels.
70 Pre-delay is commonly used in reverb and other effect processors. It delays the desired effect, and that creates a space in time between the dry sound and the effected sound. 71 Normalization is the adjustment of audio files to have similar loudness values, and this makes a music library sound more uniform. The reason is to keep consumers from having to continually increase and decrease the volume of their audio players. 99 If you are solely using CDs to distribute your recordings, you can keep your audio volume at whatever level you prefer (take into account that those listening to your music may listen in a moving car). However, I usually reduce the threshold of the limiter to barely touch the highest peak of the audio, and then set the output ceiling to around -0.3 dBTP—CDs require that your output level (output ceiling) be no higher than -0.1 dBTP. Once your threshold is set, just lower your ceiling to be within that standard. Also realize that limiters and maximizers will aggressively compress your audio if the threshold is set too far below your highest peak, which is something you will want to steer away from in classical music. In my opinion, the limiter, which is essentially a compressor set to a ratio of 10:1 or higher, should not be reducing more than a decibel, although I try to keep it below that figure. For digital distribution, I stick very closely with the standards set by streaming services, so my WAV, AIF, MP3, and FLAC files do not exceed an integrated loudness of -14 LUFS and peak loudness of -1dBTP. Do not feel as though you need to push the integrated loudness of your audio to -14 as that will require way too much compression for a classical recording.
Exporting
Regardless of what sample rate and bit depth you recorded in, there are set standards for this, too. CDs require a WAV or AIF file at a sample rate of 44.1kHz and a bit depth of 16. Files exported for CD should be exported with those settings. Online streaming services are more flexible, as they convert the files to what they need, regardless of the file type, sample rate, or bit depth you upload. Most streaming services will tell you which file type, sample rate, and bit depth they prefer, and it should be as easy as a google search. Spotify’s FAQ page currently says that they accept FLAC and WAV format, but that they prefer FLAC. Most all streaming services accept bit depths up to 24, and sample rates above 44.1kHz (48kHz, 88.2kHz, 96kHz) may be
100 fine. Larger bit depths and samples rates enhance sound quality, even when compressed to MP3 or AAC, but you must initially record and keep these settings throughout the process.
Dithering
Dithering is the addition of a low volume, random noise signal to remove the harmonic distortion caused by quantization error. Although seemingly complex, dithering is an important process that can ensure comparable sound quality when audio files are converted to lower resolutions, such as from 24 bits to 16 bits for CD. During the conversion process, the least significant bits of a high-resolution signal are rounded so that the signal fits into the capacity allowed by the lower resolution bit depth, and that cyclical rounding of the analog signal is what produces quantization errors, also known as truncation distortion. By adding a random noise signal, the quantization errors are randomized, and that breaks the distortion pattern. Essentially, harmonic distortions are replaced with noise, which is a smoother and far less abrasive sound, so the original analog signal appears unaffected by the conversion process.72
Izotope, inc. uses the following analogy to explain dithering: hold your hand up in front of your eyes to obstruct part of your vision (truncation distortion). Now, quickly and repeatedly wave your hand back and forth (dithering). Waving the hand allows you to see everything in front of you, as if there is no visual hindrance. 73 Modern Recording Techniques says this about dithering:
The concept of dithering relies on the fact that noise is random. By adding small
amounts of randomization into the quantization process, there is an increased probability
that the D/A converter will be able to guess the least significant bit of a low-level signal
72 Izotope, inc. 2012. Intoduction to Dithering. October 1. Accessed November 17, 2019. https://youtu.be/vVNzylf9sGo. 73 Izotope, inc. 2012. Intoduction to Dithering. October 1. Accessed November 17, 2019. https://youtu.be/vVNzylf9sGo. 101 more accurately. This is due to the fact that the noise shapes the detected sample in such
a way that the sample-and-hold (S/H) circuitry can determine the original analog value
with greater precision.74
Those with experience in digital video or photography might understand dithering quite well, as it is the process of adding small amounts of grain (random noise) into an image to break up color banding and other artifacts that visually manifest in the image when compressing to smaller file sizes.
Dithering can be done in your DAW during the exportation process (a.k.a. rendering or bouncing); however, you can dither after the fact, as well as down-convert bit and sample rates. I normally use Izoptope RX in standalone to dither and down-convert bit rates, mainly because
Izotope RX has, in my opinion, higher quality options. With that said, the options in your DAW should be perfectly fine in most cases, but you should not dither unless you are reducing the bit resolution of your audio, such as moving from 24 bits to 16 bits.
Some think that dithering should always be used, and I think that is incorrect. However, for many recordings, random noise may already exist within the signal. The wash of white noise produced by moving air might create enough noise to suffice for dithering purposes. Cheap preamps or microphones, as well as many types of digital or analog processes, can also add noise to your signal. It’s really up to you as the engineer. If you reduce the resolution of a track and notice truncation distortion (you should become familiar with this sound), then consider dithering, but if not, you are fine to proceed without it. Also, 24-bit audio is slowly becoming the new standard as CDs are now outdated and internet speeds continue to increase, so there may be a time when dithering will also become obsolete and unnecessary.
74 Huber, David Miles. Runstein, Robert E. 2010. Modern Recording Techniques. Burlington, MA: Elsevier Inc. 102 Distribution
Here are some common options for distribution: sell digital or hard copies directly through a website; use online services like Tunecore or CD Baby, which will sell digital and hard copies, as well as get your music on streaming services; or sign a deal with a record label (e.g.,
Naxos, Tonar, Azica, and Teldex). In the end, your main concerns are who keeps the rights to your recordings,75 as these can decide who makes what percentage off your music’s sales, and what will make you the most money. Selling the record yourself guarantees that you keep the most profit, but it limits your ability to sell to a wide audience. Using a distribution service like
Tunecore or CD Baby allows you to sell more copies, as well as get in with online retailers and steaming services, but they charge annual fees for their services.
Signing a deal with a record company is also possible, especially if you already have the product ready for distribution, but you will have to sacrifice some or all of your rights to the music and/or recordings, which means that you are losing a percentage of your royalties. If you think that you have a product that everyone will want, this option could make you the most profit. Typically, a good record label will have a lot of available resources, and they may use those for a signing bonus, up front living expenses while you wait for the record to catch on with consumers, and whatever costs you can claim for cutting the record (cost of equipment, rental of recording space, and so on). Record labels may also provide you with a concert tour, television special, or some other event intended to advertise your record. Of course, this comes at a cost, and they may recoup all of their investments before you start receiving a respectable portion of the profits.76
75 Because classical musicians typically play the music of others, the only rights they have are the rights to the recording, which is what record labels will want if they make a deal with you. 76 Passman, Donal S. 2015. All You Need to Know About the Music Business. New York, NY: Simon & Schuster. Page 442 103 Conclusion
Before you record, consider the sound you want, then spend time deciding how you might achieve that particular sound (e.g.., what room, microphone type, and microphone placement). Before you begin recording, do some tests to find out if you need to make any adjustments. You do not want to get home after a long day of recording on location and find out that there is a harsh resonance throughout, or that your metronome was audible, or that you accidentally recorded in mono. Once you finish the recording, take your time with the editing. If you do not, you will make mistakes, and you may not even notice them until your printed copy reaches the first customer. Also, if you are uncomfortable with mastering, there are plenty of professional mastering engineers ready to give you the professional polish that you desire, but if you do intend to master the audio yourself, be sure that you are working on a very precise sound reproduction system.
104 CHAPTER 6
SONIC EMULATION
In my experience, there are two ways to emulate the sound of a recording. The first is to try to match your desired sound during the recording process, and the second is to try to match that sound during post-processing. Both provide certain challenges, and a really good emulation will require aspects of each method. However, before you make any attempt at emulating a recording, it is important to acquire knowledge of the recording process and resulting sonic architecture of your reference recording. Chapter Two, which documented many of the factors that created great guitar recordings, is how you begin. Find out which microphones were used, which preamps, converters, and so on, then see if you can find identical or near identical recording tools, which includes finding a microphone with a similar frequency response and design—response charts are usually published on the internet or in manuals. Place your microphones at similar distances and angles from the performer, use a similar type of guitar, make sure the room you use is similarly designed, and use absorbers and diffusers if necessary.
Chapter Three analyzes the measurable and audible result of finished recordings, as well as introduces concepts of emulation. If you find that your recording is still too different from the one you are trying to emulate, and it’s not related to aspects of how the piece was performed, then you can analyze your recording and make a comparison. Does your recording have too much low-end, or does it lack brilliance? Is it quieter or much louder? Is it punchy or boxy enough, or does it lack fullness? Is there any audible compression, saturation, or additional reverb? And, whether intentionally or not, which components were used to add these effects?
These are all questions that require answers before you begin the emulation process. Matching a recording may be as simple as a wide-band bump or cut at a certain frequency range, or the 105 addition of a digital reverb. Emulation may also require sending your recording through a vintage tape machine, as well as other hardware devices—a good digital emulation of those devices may suffice.
Ultimately, you need to do your homework, and you should understand that emulating a desired sound does not mean creating a complete match. Your recording does not need to be, nor will it be, indistinguishable. That is not the goal. Emulation, in my opinion, is a way to find your sound, as much as it is finding someone else’s. You are working to create a guitar recording that sounds the way you desire, regardless of whether that is extremely realistic or creatively unrealistic.
Finding the Sound
One of my favorite recent finds is a company called mix:analog. They have a website that allows you to connect with real hardware, including classic tape machines, a Fairchild 670 tube compressor, an Elysia Museq, which is currently my favorite tone-shaping equalizer, and many more audio goodies. You create an account, upload your files, buy tokens, and schedule a time to use the hardware. There is a web-based GUI that allows you to change settings with your mouse, then you bounce the track when your settings are finalized. Simply sending your recording through their Studer A812 (updated A80) or Telefunken M15 tape machine can add an interesting vintage quality, and topping that off with the subtle addition of an old tube compressor, like the Fairchild 670 (or a modern clone), would reinforce that characteristic. There are also many digital emulations of old preamps, compressors, and tape machines available for purchase, and digital emulations are, these days, nearly identical to their hardware counterpart, which means that you can very easily add vintage coloration to any recording.
106 A lot goes into obtaining a “classic” sound, but it’s important to note that most early recordings, especially those after the 1960s, would have likely been recorded onto tape. As stated in America on Record, “When the Beatles first entered Abbey Road recording studios in 1962, the transition from disc to tape recording was complete.”77 However, “tape recorders were first installed in control rooms around 1947,” which was because they made “longer recordings [than disc lathes], and it was easy to edit them.”78 With that information, your first stop in obtaining a vintage sound, such as one recognized with Julian Bream or older John Williams recordings, would be to process your audio through a tape machine, and maybe try to find the same one that was used on whatever reference track you are modeling. It is also important to note that most active circuitry before the advent of solid-sate (transistor-based) amplification, such as the circuitry found in most new microphones, would have likely used a vacuum tube, which famously adds its own harmonic fingerprint, so you may consider using tube microphones or preamps. You could even feed your audio through a tube-based hardware device, such as the
Fairchild 670 compressor, to acquire similar results.
Emulating a modern recording, however, tends to be centered around less coloration. I regularly hear musicians and music enthusiasts use a golden age mythology in reference to their distaste of modern classical recordings, stating that audio recordings are not as good as they used to be because of all the processing. This idea could not be further from the truth. If you take another look at Chapter One, you will see that many of those engineers are placing great microphones in front of great players that are performing in great rooms, and that is it. There are few accounts of digital reverb or equalization, and only one engineer even mentions
77 Millard, Andre. 2005. America on Record: A History of Recorded Sound. New York: Cambridge University Press. Page 295 78 ibid., page 289 107 compression. My opinion is that, presently, recording equipment has become so advanced that engineers are closer to the truth than ever before, and for those that grew up listening to vinyl and tape, this neutral, uncolored sound is displeasing.
Finding the sound you desire is about determining what you want. Maybe that is the ultra-clean, digital recordings of today, or maybe you want a bit of both worlds. I find myself frequently returning to the audio recording of Xavier Jara playing Jeremy Collins’ Elegy not just because the composition and performance are so well done, but because of the sound of the
Sonodore MPM-81 tube microphones that Norbert Kraft used. These microphones, which are modern creations that are built around an archaic technology, reproduce sound with immense detail, but also with a unique richness and opulence that evokes a feeling of nostalgia.
Before Recording
Before you record, document the process of the recording you wish to emulate. This may require emailing the producer or engineer of the selected recording, or if the recording is old
(e.g., from the 1960s), you may be required to research the popular techniques and equipment of that era. Once you have that information, decide how to realistically match those variables. Can you afford or find the same or similar preamps or microphones? Are you able to record in a similar space with a similar auditory design? Do you have access to vintage gear, like tape machines and tube compressors, or are there reasonably priced analog clones or digital recreations on the market? Depending on the recorded sound you want to achieve, there may be more or fewer factors to consider, but the truth is that, even if you don’t get as close as you would like during the recording session, the flexibility that we have in the digital age allows you to make almost anything happen. My belief is that you are able to record a completely isolated
108 guitar and still achieve similar results to whatever sound you seek to emulate, which I attempted and documented later in this chapter.
During the Recording Session
During the recording session, do some test recordings and listen to them before you pick a final microphone placement or seating position. You want to produce a similar balance between the source and room, as well as roughly matching the tonal balance (angling a microphone at the bridge of a guitar produces a sound with more body and fullness, and angling it at the neck fret produces a sound with more bass and brilliance). If you know that reverb was added in post, consider recording a less reverberant sound than that of your reference, and if you know that reverb was not added, try to produce a similar balance on location. Be sure to record small samples and compare them to your reference recording. Find the sample that sounds the closest and place everything back in that position. If you are still receiving results that are not as close as you would like, remember that work can, and likely will, be done during the post- processing stage.
Emulation in Post
The last step of the emulation process happens around the same time as the mastering process. This is where you use your analyzed reference tracks to further simulate auditory conditions, such as a similar frequency curve, reverb type and amount, as well as integrated loudness, dBTP, and so on. When comparing your tracks, it’s important that you match sound levels prior to making any decisions. According to the Fletcher-Munson curve,79 louder sounds typically appear to have more bass and treble than quieter sounds, so if you are listening to your recording at a lower sound level than the track to which you are referencing, your track will
79 This curve, which has been made obsolete by ISO 226:2003, depicts the human ear’s natural balance of sound at different decibel levels. 109 appear comparatively weak at those frequencies. You may also consider using software crafted specifically for referencing (“A/B-ing”) your music with your reference track(s). I am only aware of a few: Reference, Metric AB, and MCompare, but there are likely more plugins, as well as stand-alone software, designed to handle this task.
Like mentioned in the post-processing guideline in Chapter Four, there is a specific workflow that I recommend for every project: surgical EQ, tone-shaping EQ, compression
(compression can happen before tone-shaping EQ), reverb, limiting/maximizing, and then dithering, if needed.80 If you are planning to match your sound to a recording that utilized vintage gear, such as tape machines, analog preamps with transformers, and/or tubes, you may consider applying those effects prior to your mastering workflow or after surgical equalization.
The idea is to impose any characteristics that have yet to be achieved into the source. There are many digital emulations of classic studio gear on the market to help you in this endeavor, but you can also use professional recording studios or other services, such as mix:analog, that have a variety of real analog gear available.
Digital Emulation
Let’s look at what a digital emulation can do for a recorded sound. Typically, I trial plugins before I buy them, and I regularly test those plugins with a software called Plugin
Doctor. This software allows me to see what the plugin is doing, like changes in the phase relationship, harmonic content, and frequency curve. I will now include images from that software to display the changes produced by several third-party plugins that I own, such as a couple digital equalizers, an emulation of an analog equalizer, a tape machine, a tube compressor, and more, which will all be fed with a 1kHz sine wave. For each plugin, you will
80 It is also important to note that you can choose your own order, skip steps, or completely avoid any processing. 110 find 4 images: the first is a harmonic analysis (top left), which is followed by phase analysis (top right), imposed frequency curve (bottom left), and dynamic changes (bottom right). All plugins performed at a sample rate of 96kHz (two times 48,000Hz).
Figure 6.1 Unaffected Signal
Figure 6.1 is an unaffected 1kHz sine wave. It is apparent that, apart from some inaudible noise located in the harmonic analysis, the 1kHz signal remains pure and unchanged.
Figure 6.2 is the BX_Digital V3 (digital EQ). I used a bell filter at 500Hz with a 7 decibel increase and a Q factor of 1. The harmonic analysis (top left) visualizes an increase in noise, especially at 500Hz, but that noise is inaudible. The phase analysis (top right) displays slight
111 phase shifts before and after 500Hz (500Hz remains phase aligned), the Frequency (bottom right) shows the change at 500Hz, and the dynamic level (bottom right) is unaffected.
Figure 6.2 BX_Digital V3
Figure 6.3 is the Izotope Ozone 7 (digital EQ). I used a linear-phase bell filter at 500Hz with a 7 decibel increase and a Q factor of 1. The linear phase plugin has introduced some strange noise patterns in the harmonic content, but this, too, is inaudible. The phase, of course, remains unchanged, which is the point of using a linear phase equalizer. The frequency curve shows the adjustment at 500Hz, which also displays a much narrower bell filter than Figure 6.2, although the settings are identical. There is no change in dynamics.
112
Figure 6.3 Izotope Ozone 7 Equalizer
The linear-phase plugin in Figure 6.3 has introduced a strange noise pattern in the harmonic content, but this, too, is inaudible. The phase, of course, remains unchanged, which is the point of using a linear phase equalizer, and the frequency curve shows the adjustment at
500Hz, which also displays a much narrower bell filter, although these settings are identical to the previous plugin. There is no change in dynamics.
Figure 6.4 is the Brainworx Elysia Museq equalizer (digital emulation). Bell filter at
500Hz with an increase of 7 decibels at a Q factor of 1.3 (fixed at this factor). Now that I have started using plugins modeled after analog circuitry, we will begin to see more drastic alterations to the measurements, all of which are characteristics inherent to the analog circuit after which the
113
Figure 6.4 Brainworx Elysia Museq
plugin was modeled. In the harmonic analysis, we see additional noise at 500Hz (similar to
BX_Digital V3), but we also see the formation of two new harmonics. The first is a second-order harmonic, which is an octave higher (2kHz) than the original signal (1kHz), and the second is a third-order harmonic, which is an octave and a fifth higher than the original. Second order harmonics reinforce the source, and third-order harmonics add to the source. The phase analysis shows a slight shift before and after the 500Hz point. The frequency graph exhibits the widest bell filter, and it also shows a high frequency cutoff near 35kHz. I believe this cutoff acts as an anti-aliasing filter, ensuring no fold back of harmonics created above Nyquist, but I believe it is rather unnecessary. It may otherwise be a characteristic of the hardware. The dynamic level
114 remains essentially untouched, although there is a gain increase that I was unable to correct, which I did in all previous plugins.
Figure 6.5 is the Brainworx Millennia TCL-2 optical compressor (digital emulation). I set the threshold at three-o-clock, attack at 50ms, release at .75sec, and ratio at 2:1. The plugin is in tube mode (there is an optional J-FET transistor mode).
Figure 6.5 Brainworx Millennia TCL-2
The Millennia TCL-2 is a tube and J-FET compressor, and I believe it was originally intended to be used as a transparent option for mastering engineers. As you can see in the harmonic analysis, there is very little coloration, and that coloration is the introduction of a
115 second-order harmonic, which will reinforce and fatten the source. The phase and frequency graphs both exhibit slight changes below 5Hz, which are inaudible, and the dynamics are showing roughly 5 decibels of compression with a soft knee. I pushed the threshold lower than I normally would in order to more easily exhibit the changes produced by this digital emulation.
With all analog modeled plugins, the more you push the them, the more they color your sound with their characteristics. These settings, with a higher threshold (less compression), would produce a very transparent compression that should still add a bit of richness to the source.
Figure 6.6 Brainworx Millennia TCL-2 J-FET
116 Figure 6.6 is the Brainworx Millennia TCL-2 optical compressor (digital emulation), but it is set in J-FET mode. In my opinion, this option is an airy and natural sounding compression; the previous setting has a distinct thickening quality. You can see the addition of a third-order harmonic, as well as a reduction of the second-order harmonic, but the phase and frequency curves remain the same. The dynamics show the same 5 decibel compression, but with an even softer knee. Overall, this option sounds more discreet.
Figure 6.7 Kush Omega Transformer: Model N
Figure 6.7 is the Kush Omega Transformer: Model N (digital emulation). The intensity is set to 12-o-clock. This plugin is an emulation of a vintage Neve preamp, although Kush’s
117 website does not detail which one they are using. Looking at the harmonic analysis, there are many additional harmonics based off the third-order harmonic of the original signal. The phase analysis depicts a large shift (up to 360 degrees positive and nearly 360 degrees negative) that correlates to the frequency curve, which contains a high and low pass filter in those areas of phase shift. As long as both left and right channels of the stereo signal are treated identically, these phase shifts will not affect the phase alignment. The dynamics show approximately 3 decibels of compression was added, which starts at approximately -12 decibels from maximum amplitude.
Figure 6.8 Brainworx Black Box HG-2
118 Figure 6.8 is the Brainworx Black Box HG-2 tube saturator (digital emulation). All settings are at 12-o-clock. The “air” option is off, and the “saturation” is on. This is currently my favorite digital emulation. The Black Box HG-2 is perfect for adding subtle tube saturation to nearly anything. The harmonic analysis shows the addition of second- and third-order harmonics.
The phase shift is subtle, and it correlates to the faint, yet audible, shift in the frequency curve
(the Black Box makes the signal brighter, but not harsh). The dynamics depict some strange anomalies at the lower, inaudible area, as well as roughly 4–5 decibels of compression, which starts when the audio level passes approximately -13 decibels from max amplitude.
Figure 6.9 Waves J37 Tape Machine
119 Figure 6.9 is the J37 Tape Machine by Waves (digital emulation). Settings: 815, input at -
9 and output at +8.6, nom bias, modeled tracks at 2+3, wow and flutter rate at 12-o-clock, noise and sat at 0, and no delay. Because this is a tape emulation, the effects of this plugin oscillate as they would in a real tape machine, continually altering the way the signal is affected. The harmonic analysis shows additional harmonics based on the third order, as well a noise floor that is quite high, although still not to the point of being audible (you can turn up the “noise” setting to make it audible). The phase graph displays sharp changes at ranges that are inaudible (below
20Hz and above 20,000Hz), then smoother, more natural shifts in the audible range. The frequency graph also displays some sharp effects. As you can see in the frequency graph, the plugin treats the left and right signal with slight variations that remain in phase. This will add width to a stereo source, although not much. The dynamics remain intact, but pushing more signal into the plugin, as well as increasing the “saturation” setting, will cause some compression and more dramatic harmonic coloration.
Post-Emulation in Practice
The first example is of a double-top guitar recorded in a large living room. The AT 4041 microphones were in A/B, and they were roughly 2 feet from the performer and 5 feet apart. The room had thin Berber carpet, and the size of the room was approximately 20 x 13 x 15 feet. The original intention of the recording was to record an isolated source that could be processed to sound like it was in a small music hall, but I have now reprocessed this recording to sound similar to David Russell in Aire Latino. From this album, I chose his recording of Allegretto from Morel’s Sonatina as my reference, and the recording that I will manipulate is the Allegro from Bach’s Prelude, Fugue, and Allegro (BWV 998) transcribed to D major. Stylistically these are very different compositions, but the key signature, dynamic and melodic range, and tempo,
120 were close enough for my purpose. In the end, the results were strikingly similar. I did not somehow make this anonymous guitarist sound like David Russell playing a Matthias Dammann double-top, but I was able to create a very similar tonal balance, as well as similar room acoustics and midrange detail. I even used a similar dithering process. However, obvious differences, such as a lack of nail-related articulation in the subject’s sound, precluded me from getting as close as I would have liked. Here are images detailing the evolution:
a) Russell reference recording
b) Subject pre-manipulated
Figure 6.10 Russell Emulation Pre-manipulated
121
a) Subject pre-manipulation
b) Subject after manual manipulation.
Figure 6.11 Russell Emulation Manual Manipulation
For the subject’s manual manipulation in Figure 6.11, I used the spectrograph, as well as my ear, to manually manipulate the recording to sound similar to Russell’s. What can’t be seen is that I also added digital reverb to further emulate Russell’s sound. Here is a list of the alterations:
122 • In Logic Pro, four separate instances of “channel EQ” were opened. The first was used to
surgically remove one unwanted resonance at 415Hz. The Q factor was set to 7.30, and 5
decibels were removed.
• The next three instances were used for tonal rebalancing. In the first of those three, I reduced
215Hz by 6 decibels at a Q of 1.20; 750Hz by 5 decibels at a Q of 2.60; 1300Hz by 2.5
decibels; and 7600Hz by 3 decibels at a Q of 0.64. In the second instance, I added 2 decibels
to 600Hz at a Q of 4, and 3 decibels to 1040Hz at a Q of 3.2. I reduced 1500Hz by 2 decibels
at a Q of .25 (very wide), and 9300Hz by 2 decibels at a Q of 1.3. The last instance contains a
1 decibel boost at 260Hz with a Q of .37, and a 1.5 dB boost at 400Hz with a Q of 1.3. I also
reduced 1600Hz by 1.5 decibels at a Q of .2, as well as 2000Hz by 2dB at a Q of 1.9.
Figure 6.12 EQ Set for Russell Emulation
123 Figure 6.12 (continued)
The last three equalizers may seem arbitrary, but each was added with the purpose of getting closer to Russell’s tonal quality and frequency curve. Some alterations required more narrow shifts, and others required wide shifts. Each EQ point was evaluated by studying its effect on the frequency curve, as well as its audible effect, and I have to say that I was rather impressed with the end result.
After equalization, I used convolution and algorithmic reverb to create an acoustic space similar to the one featured in Russell’s recording. All reverbs were placed on an effect send
(“bus send” in Logic Pro), not directly onto the stereo track. The first one was Liquid Sonics’
Seventh Heaven reverb with the “Studio A” algorithm, and it was used to add early reflections, creating more space between the guitar and the microphone. The next reverb was Wave’s IR-L,
124 and I used the Apollo Theater impulse response, which I thought added a similar warmth and
LEV (Listener Envelopment) to that of the late reflections in Russell’s recording. That impulse response was gently equalized to further imitate Russell’s room sound, then I sent the product of those two plugins through another reverb (Seventh Heaven with the “Sandors Hall” algorithm) to slightly increase the reverb tail and thickness, which I pre-delayed by 15ms.
I then bounced the file, put it in Izotope RX for closer evaluation, and decided to utilize the EQ matching option (Figure 6.13), which pushed me slightly closer to Russell’s sound. The
EQ matching software is great at easily creating very complex EQ curves, and it successfully refined my adjustments. The more noticeable shifts are at 200Hz and ~10.5kHz.
a) Subject with manual manipulation
Figure 6.13 Manual Manipulation Versus Izotope’s EQ Match
125 b) Subject after EQ Match
Figure 6.13 (continued)
a) Russell reference
Figure 6.14 Russell Emulation Result
126 b) Subject’s resulting curve
c) A review of the subject’s untouched frequency curve
Figure 6.14 (continued)
The last thing I could do is alter the subject’s stereo width, and there are many plugins available that do that well. Listening to my emulation against Russell’s, the width is close enough that I do not care to add this process. Also, keep in mind that some widening plugins can
127 also decrease width, if need be, and if you are emulating an old recording that is in mono, most
DAWs have built-in plugins that will easily monetize a stereo recording.
This next emulation uses the same subject, but a new reference and piece. I picked
Scarlatti’s K.11 Sonata performed by Julian Bream on his 1959 album, The Art of Julian Bream, which was released by RCA in 1960. Since our subject’s audio is already recorded, we are, as stated previously, emulating exclusively in post-processing. This is not ideal, but the results are a testament to the flexibility available with digital processing. For Russell’s emulation, I gave a detailed account of my work. For this emulation, I will simply include images of all processes, then I will write a short explanation of my process at the end. I used Ableton Live for this emulation, which was decided for the simple reason that I wanted to utilize a different DAW, and
I also incorporated more third-party plugins than previously.
Here are the plugins added directly to the stereo channel:
Figure 6.15 Ableton EQ with Oversampling Activated
Figure 6.16 Soundtoys Radiator
Figure 6.16 Soundtoys Radiator
128
Figure 6.17 BX_Digital V3-1
Figure 6.18 BX_Digital V3-2
129
Figure 6.19 BX_Digital V3-3
Figure 6.20 BX_Cleansweep Pro
130 The following plugins were placed on an effect send (“return track” in Ableton Live)
Figure 6.21 Liquid Sonics’ Seventh Heaven for Early Reflections (Return A)
Figure 6.22 Ableton EQ to Shape Tone of Reverb (Return A)
Figure 6.23 Seventh Heaven for Late Reflections (Return B)
131 The next two plugins were placed on the main stereo out (also known as the master bus)
Figure 6.24 Waves J37
Figure 6.25 Waves Kramer Tape
132 A few words about the plugins utilized: The first stock EQ was used to remove unwanted resonances (Figure 6.15); Radiator added the tube grit and dirt of an Altec 1567A Mixer
Amplifier from the early 1960s (Figure 6.16); Figures 6.17–6.19 are the BX_Digital EQ, which I find to be very pleasing and transparent, and they were used to make the frequency curve align more with Bream’s (the third instance was used in mid/side mode to reduce ~400Hz in the middle of the image, effectively making the guitar sound more open); and BX_Cleansweep reduced high frequencies (Figure 6.20). The reverb components added early and late reflections, and the attempt was to emulate Bream’s space, to which I believe I was close (Figures 6.21–
6.23). The J37 and Kramer tape machines are not the correct hardware devices used to track
Bream (Figures 6.24 and 6.25), and neither is the Altec 1567A Mixing Amplifier (Figure 6.16).
From my research, I think it is likely that Bream was recorded onto an Ampex tape machine with a ribbon microphone, which was amplified by a custom mixing amplifier built by RCA (refer to
Chapter Two). However, all of these plugins together, at those settings, provided a convincingly similar sonic quality.
In the end, our subject’s recording found its way into the same sound arena as Bream’s, but there were qualities that eluded my efforts. It may have been the fact that the guitar of our subject is a double-top, while Bream is playing a classic Herman Hauser. Bream also plays with a lot of color, which changes from very brittle to very sweet, and our subject was much more conservative in his use of tone color (that is impossible to digitally recreate). I utilized a lot of plugins that all made small differences, and all these little differences, when combined, played their part in reaching my sonic goal. Here is the resulting frequency curve:
In Figure 6.26, the top image is the Bream reference, and the bottom is the subject’s untouched audio.
133
a) Bream reference
b) Subject’s pre-manipulated frequency curve
Figure 6.26 Bream Emulation Pre-manipulation
134 In Figure 6.27, top image is the subject without manipulation, and bottom is the manual manipulation.
a) Subject’s pre-manipulated frequency curve
b) Subject with manual manipulation
Figure 6.27 Bream Emulation Manual Manipulation 135 In Figure 6.28, the top image is manual manipulation, and the bottom is extra processing from
EQ match in RX.
a) Subject with manual manipulation
b) Subject after EQ Match
Figure 6.28 Bream Emulation with EQ Match
136 In Figure 6.29, we have Bream’s reference, the subject’s end result, and the subject’s untouched audio.
a) Bream reference
b) Subject end result
Figure 6.29 Bream Emulation Result
137
c) Subject pre-manipulated curve
Figure 6.29 (continued)
The interpretations of the music are wildly different. However, the adjustments to the frequency curve, additional tube and tape coloration, as well as the fabricated noise, have all manifested in a natural way, and the subject’s audio does now have a very vintage quality. In this regard, I would call it a success. There are some subtle qualities in our subject’s guitar tone that, after emulation, sound far more like Bream’s, but getting any closer would, at the very least, require the use of a similar guitar.
Conclusion
Emulation begins at the start of your recording process. It begins with a documentation and analysis of the recording you want to emulate. To fully replicate a sound, you need every factor to be identical. The farther you diverge from the variables that created your reference recording, the less your recording will emulate that sonic architecture. However, I think it is
138 important for a recording to have its own sound. You can use the emulation process to put your recording in the same sonic arena, but I do not believe it is actually ideal to fully emulate someone else.
As discussed in this chapter, we are in a time where anything is possible. You have the ability to impose era-specific characteristics to any modern recording, and you may even be able to do it inside your computer with a few plugins, as opposed to finding and operating a physical hardware device. With that said, if you do want to use real hardware devices, services like mix:analog make it easier than ever to do just that. In the end, the most important factors are that you are happy with the recording, and that your consumers happy with it, too.
139 APPENDIX A
GLOSSARY OF TERMS
An expansive glossary of recording terms provided by Sound on Sound is available online. It was used as a reference.81
A/B – A pair of microphones placed parallel to each other. Also, this term is used as a verb for
comparing two separate recordings.
Active Device– A hardware device that requires an electrical current to operate.
Aliasing – An effect that occurs when audio frequencies that surpass a digital systems Nyquist
limit are created. These sounds return into the audible range at an alias frequency,
creating unwanted distortions in the signal.
Ambient Field – The sound field of a room that is dominated by reverberations of the source.
Analog to Digital Converter – The component of a recording chain that transforms an electrical
signal to a digital signal.
Audio Interface – A physical device that allows its user to communicate with a computer. The
audio interface is responsible sending and receiving audio signals.
Automation – pre-programmed adjustment of parameters.
Bell Filter – An equalization filter that forms the shape of a bell when used to increase or
decrease a frequency range. A bell filter is also called a peak filter.
81 Sound on Sound. n.d. Glossary of Technical Terms. Accessed Novemeber 1, 2019. https://www.soundonsound.com/sound-advice/glossary-technical-terms. 140 Bidirectional Pattern – A microphone pickup pattern that captures sound directly in front and
directly behind the microphone, producing a polar pattern that is similar to the figure-
eight shape.
Bit – The basic unit of measurement in a digital system. A bit can be either “0” or “1.”
Bus – In this document, a bus is an auxiliary effects channel to which sound sources are routed.
Cold – Refers to a sound that is tonally bright, emphasizing the high frequency range.
Cardiod Pattern – A microphone pickup pattern comparable to that of a heart ideograph. This is a
directional pattern with an on-axis pickup range of ~130 degrees. Super-cardiod
microphones have a pickup angle of ~115 degrees. Hyper-cardiod microphones have a
pickup angle of ~105 degrees.
Coincident Pair – A pair of microphones with their diaphragms mounted as close to each other as
possible so that sound reaches them at the same time. Many use coincident pair to
describe an X/Y or Blumlein configuration, but any pair of microphones with
overlapping diaphragms could be considered a coincidental pair.
Compression – The reduction of a signal once it surpasses a threshold.
Condenser Microphone – A microphone that converts soundwaves into electrical current by way
of a capacitor.
Diaphragm – This part of a microphone vibrates when it comes in contact with soundwaves,
which is how the microphone converts the sound source into an electrical current.
Digital Audio Workstation (DAW) – A software program or hardware device used for recording,
organizing, and mixing audio.
Digital to Analog Converter – A D/A converter transforms a digital signal into an analog signal.
141 Dithering – The addition of noise in a digital signal when converting to lower bit depths. The
reduction of resolution causes quantization errors, and the dithering process helps to
prevent and mask those distortions.
Dry – A source without ambience, reverberation, or any kind of signal processing.
Dynamic microphone – A microphone that uses a metal coil in a magnetic field to induce an
electric signal.
Dynamic range – The difference in decibels between the loudest and softest sound level.
Early Reflections –The first reflections of a sound source.
Effect Send – Also called an auxiliary channel or bus. Effects are placed on these channels, then
audio signals are sent to them to for processing (e.g., reverb or compression).
Equalizer – A signal processor that increases or attenuates specific frequency bands.
Fidelity – The precision of a reproduced sound source.
Fletcher-Munson Curve – A curve that depicts the sensitivity of the human ear to various
frequencies. It is also called the equal loudness curve.
Frequency Response Curve– The contour of an audio components sensitivity to certain
frequency bands within its transmittable range.
Gain – The increase of a signal’s sound level.
Gobo – A movable sound absorption panel.
Harmonic Distortion – Additional harmonic information that was not present in the original
source.
High-Pass Filter – The attenuation of frequencies below a designated frequency point.
In Phase – When coincident sound sources have identical polarities.
142 Isolated – This typically refers to a room that does not allow internal or external noise to
permeate its barriers. It can also refer to an unadulterated sound source, such as one
recorded in a heavily treated recording booth.
Jitter – A blurring effect in the audio signal that happens during A/D conversion.
Knee – How gradual or immediate (soft knee versus hard knee) compression is introduced when
a sound source passes a compressor’s threshold.
Level – The signal strength.
Limiter – A compressor with a hard knee and high ratio used to avoid signal clipping.
Line Level – The normal level of an audio signal when working in a studio. Mic level is below
line level, which is why pre-amplification is required for this signal level, and speaker
level is above line level, which is why amplification is also required for this signal level
required.
Loudness – The human perception of a sound’s intensity.
Loudness Range (LRA) – Loudness range (LRA) is a general measurement of an audio
waveform’s dynamic range. It was defined in ITU-R 128 BS. 1770. The loudness range,
which is measured in loudness units (LU), is the difference in the lowest integrated level
and the highest integrated level. It implements a gating feature that removes low-level
sound (e.g., air conditioner noise or other unimportant sounds) and very short, loud sound
events that are inconsequential to the audio’s perceived loudness.
Low-Pass Filter – The attenuation of frequencies above a designated point.
Masking – The phenomenon where louder sounds inhibit the ears ability to hear softer sounds in
the same frequency range.
143 Master Bus – Also called main out and master send, this is the audio channel that is sent to your
main monitors.
Mid-Side – A miking technique that uses a cardiod and bidirectional microphone as a coincident
pair. The cardiod microphone is angled at the source, and the bidirectional microphone is
angled to face the left and right sides.
Near-Coincident Pair – A pair of microphones with their diaphragms placed in close proximity,
such as when using an ORTF or closely spaced A/B microphone placement technique.
Noise Floor – The amount of noise existing below the signal.
Normalize – The process of increasing a signal’s level to a specific limit.
Nyquist Frequency – A theory by Harry Nyquist that states that the highest digitally reproducible
frequency is dependent on the sample rate. A Nyquist frequency of 20,000 Hertz requires
a digital sample rate of at least 40,000 Hertz.
Off-Axis – The area of the microphone’s pickup pattern that is less sensitive to sound. Typically,
this means that the sound source is approaching the microphone from an angle that does
not coincide with the diaphragms physical motion.
Omnidirectional Pattern – A microphone pickup pattern that captures sound equally or near-
equally at all sides.
On-Axis – The area of a microphone’s pickup pattern that is most sensitive to sound. Typically,
the sound source is approaching the microphone at an angle that does coincide with the
diaphragms physical motion.
ORTF – A microphone placement technique developed by the Office de Radiodiffusion
Télévision Française at Radio France. In this technique, two cardiod microphones are
144 placed approximately half a foot apart (17cm) with their diaphragms angled 110-degrees
outward.
Out of Phase – Coincident soundwaves that are offset by ~180 degrees (half wave cycle or half
wavelength), which is typically due to timing issues.
Pan – The manual positioning of a sound element within the stereo field.
Passive Device– A device that does not utilize an electric current.
Phantom Power – Power provided to microphones and other components via an audio cable.
Phase Cancellation – An effect that happens when similar soundwaves that are ~180 degrees out
of phase coincide, reducing their amplitude.
Plate Reverb – Artificial reverb produced by vibrating a metal plate with a transducer.
Plugin – Inside a digital audio workstation, a plugin is a piece of code that processes audio.
Polar Pattern – A microphone’s pickup sensitivity in relation to the angle of incidence of a
sound.
Post-Production – Sometimes referred to as post-processing, these are the processes that occur
after recording.
Power Amplifier – A hardware device that increases a line level audio signal before being sent to
a passive speaker.
Preamplifier – Also called a preamp, increases a low-level signal, such as from a microphone or
electric guitar, to a line level.
Pre-Delay – A parameter on a signal processing device that delays the processed sound from the
unprocessed (dry) sound.
Precedence Effect – Also called the Haas Effect, this psychoacoustic phenomenon happens when
the human brain localizes a sound source based on differences in timing. As an example,
145 a sound that is delayed in reaching the right ear, as long as it is within 30ms, will cause
the human brain to localize that sound at a leftward position.
Presence – Increasing high and upper-mid frequencies to make an instrument sound closer to the
listener or more present in a mix.
Pulse Code Modulation – The translation of analog signal into binary numbers.
Proximity Effect – A natural boost in bass frequencies when a directional microphone is placed
very close to the source.
Pure Tone – A simple sine wave with no overtones.
Quality Factor – When using an equalizer, this the bandwidth of frequencies that will be
affected.
Quantization – In analog to digital conversion, this is the process of fitting (rounding) an analog
signal into a set of digital points.
Quantization Error – Errors that occur when the analog signal does not fit within the available
digital values. This causes unpleasant distortions.
Reverb Time – The amount of time required for a sound source to drop 60 decibels.
Ribbon Microphone – A microphone that captures sound by way of a thin metal strip placed
between magnets.
RMS – The average level of a waveform.
Sample Rate – The number of digital samples taken per a second.
Schroeder Frequency – The point at which sound transitions from waver characteristic
(omnidirectional) to ray characteristics (directional).
Shelf Filter – An equalization filter that has a shelf-like shape. This filter is used to attenuate or
increase frequencies above or below a designated point.
146 Solid State – Transistor based electronics.
Transducer – An energy converting device, such as a speaker or microphone.
Transient – At the beginning of a sound, this is the initial peak in the waveform that contains the
percussive part of the sound, such as the attack of a guitar string.
Transmission Loss – The loss of sound energy that results when a room barrier allows
soundwaves to pass through.
Temporal Fusion – The psychoacoustic phenomenon where the brain combines early reflections
with the direct sound.
Truncation – The removal of data to make a higher bitrate audio file fit into a lower bitrate
format, such as from 24 to 16 for CD compatible resolution.
Vacuum Tube – A vacuumed diode that controls the flow of electrons.
Waveform – The visual graph of a sound wave.
Wet – A signal that is reverberant or affected by other processing effects.
X/Y – A coincident pair of directional microphones turned inwards to create a 90-degree angle.
Bidirectional microphones used in a similar manner are called a Blumlein pair.
Room Sound – The ambience and reverberation of a space.
Sound Image – The audible field where sound sources are localized.
Stereo Width – How wide or narrow a stereo sound image appears.
Warm – A sound that is tonally dark, emphasizing low frequencies.
147 APPENDIX B
IRB APPROVAL
Florida State University Office of the Vice President For Research Institutional Review Board Human Subjects Office [email protected]/850-644-8673
EXEMPTION MEMORANDUM
Date: 6/6/2019
To: Philip Logan Department: MUSIC SCHOOL
From: Florida State University Institutional Review Board (IRB)
Re: Use of Human Subjects in Research
Recording the Classical Guitar: A Documentation and Sound Analysis of Great Classical Guitar Recordings with a Technical Guide for Sonic Emulation
The application that you submitted to this office in regard to the use of human subjects in the proposal referenced above have been reviewed by the Secretary, the Chair, and one member of the Human Subjects Committee. The proposed research protocol is Exempt from human subjects regulations as described in per 45 CFR § 46.101(b)3.
The Human Subjects Committee has not evaluated your proposal for scientific merit, except to weigh the risk to the human participants and the aspects of the proposal related to potential risk and benefit. This memorandum does not replace any departmental or other approvals that may be required.
The Committee expects that all relevant subject protection measures and ethical standards will be followed, as outlined in your proposal. No continuing review is required unless the nature of the project changes and it would affect the project exemption status.
You are advised that any change in protocol for this project that would affect the exemption status must be reviewed and approved by the Committee prior to implementation of the proposed change in the protocol. A protocol change/amendment form is required to be submitted for approval by the Committee. In addition, federal regulations require that the Principal Investigator promptly report, in writing any unanticipated problems or adverse events involving risks to research subjects or others.
By copy of this memorandum, the Chair of your department and/or your major professor is
148 reminded that he/she is responsible for being informed concerning research projects involving human subjects in the department, and should review protocols as often as needed to ensure that the project is being conducted in compliance with our institution and with DHHS regulations.
This institution has an Assurance on file with the Office for Human Research Protections. The Assurance Number is FWA00000168/IRB number IRB00000446.
Cc: Bruce Holzman, Advisor HSC No. 2019.26906
149 APPENDIX C
RESEARCH INFORMATION SHEET
FLORIDA STATE UNIVERSITY
RESEARCH INFORMATION SHEET
Recording the Classical Guitar: A Documentation and Sound Analysis of Great Classical Guitar Recordings with a Guide for Sonic Emulation
You are being contacted to voluntarily participate in a research study because of your involvement in the recording process of significant classical guitar recordings. The information gathered from you is intended to aid in a treatise on recording techniques for the guitar, which covers practical ways to record in a contemporary format, as well as modern methods that can be utilized to emulate a favored sonic architecture. If you choose to participate, you will be asked to answer questions about recordings in which you have participated. The questionnaire should take no more than 30 minutes to complete, and data collected by phone will be recorded.
The information you provide will be included in the treatise, and you will be publicly identified. The information provided will also work in conjunction with a computer analysis of the recording(s), such as the average contour of audible frequencies, loudness, and stereo imaging, and the combined data of all recordings included in the treatise will assist with creating a methodology to aid guitarists in synthesizing a favored sonic architecture.
Returning the questionnaire will be considered a valid form of consent. If you have any questions, please contact Philip Logan or Bruce Holzman: (redacted)
If you have any questions or concerns about your rights as a research participant, or regarding the study and would like to talk to someone other than the researcher(s), you are encouraged to contact the FSU IRB at telephone number 850-644-7900. You may also contact this office by email at [email protected], or by writing or in person at 2010 Levy Street, Research Building B, Suite 276, FSU Human Subjects Committee, Tallahassee, FL 32306-2742.
150 APPENDIX D
DISCOGRAPHY
Barrueco, Manuel. 2007. Solo Piazzolla. Tonar Music 20360-12852. CD.
Bream, Julian. 2011. The Art of Julian Bream. él, Cherry Red Records 013929321137. CD.
Isbin, Sharon. 1999. Dreams of a World. Teldec Classics International 3984-25736-2. CD.
Iznaola, Ricardo. 2010. Heritage: The Guitar in Venezuela. IGW 22880-81. CD.
Jara, Xavier. 2017. Guitar Recital (GFA 2016). Naxos Records 47313-37977. CD.
Kraft, Norbert. 1997. Guitar Favourites. HNH International 730099-499927. CD.
Russell, David. 2004. Aire Latino. 089408061226. CD.
VIDA guitar quartet. 2015. The Leaves Be Green. BGS Records 34158-50403. CD.
Vieaux, Jason. 2014. Play. Azica Records 8786-71287-2. CD.
151 REFERENCES
Acoustical Materials Association. 1953. Sound Absorption Coefficients of Architectural Acoustical Materials . New York, NY: Acoustical Materials Association.
—. 1966. The Use of Architectural Materials: Theory and Practice. Ney York, NY: Acoustical Materials Association.
Bise, Alan, interview by Philip Logan. 2019. Email Questionnaire (July 11).
Chávez, William. 2015. JulianBreamGuitar.com. Accessed Octobober 13, 2019. www.julianbreamguitar.com.
Classical Guitar Magazine. 2017. Inside England's Vida Guitar Quartet: Stretching Possibilities. February 1. Accessed June 16, 2019. https://classicalguitarmagazine.com/inside- englands-vida-guitar-quartet-stretching-possibilities/.
DPA Microphones. n.d. 4006 User Manual . Accessed December 1, 2019. https://www2.spsc.tugraz.at/add_material/audiotechnik/manuals/50_Mikrofone/DPA/dpa _4006_manual.pdf.
European Broadcasting Union. 2014. ITU-R 128 BS. 1770. June. Accessed November 26, 2019. https://tech.ebu.ch/docs/r/r128.pdf.
Everest, F. Alton, and Ken. C Pohlman. 2014. Master Handbook of Acoustics. Columbus, OH: McGraw-Hill Education.
Huber, David Miles, and Robert E. Runstein. 2010. Modern Recording Techniques. Burlington, MA: Elsevier Inc.
Izotope, Inc. 2012. Intoduction to Dithering. Izotope. October 1. Accessed November 17, 2019. https://youtu.be/vVNzylf9sGo.
Jackson, Blair. 2018. Clasical Guitar Magazine. August 15. Accessed October 16, 2019. https://classicalguitarmagizine.com/noted-luthier-robert-ruck-builder-of-manuel- barrueco-famous-no-58-passes-away-at-72/.
Jewish Community Center of Greater Baltimore . n.d. The Gordon Center: Rent Our Space. Accessed June 21, 2019. jcc.org/gordon-center/rental.
Kraft, Norbert, interview by Philip Logan. 2019. Email Questionnaire (September 11 and 12).
Millard, Andre. 2005. America on Record . New York, NY: Cambridge University Press.
Nuemann. 2015. Nuemann.com. Accessed Decemeber 7, 2018. www.neumann.com/homestudio/en/difference-between-large-and-small-diaphragm- microphones.
152 Passman, Donal S. 2015. All You Need to Know About the Music Business. New York, NY: Simon & Schuster.
Roy, James V. 2014. scottymoore.net. Accessed October 17, 2019. www.scottymoore.net/studio_rca.html.
Russell, Maria and David, interview by Philip Logan. 2019. Email Questionnaire (June 21).
Sengpiel, Alexander, and Eberhard Sengpiel. n.d. Vorlesungs-Unterlagen 5. Accessed July 31, 2019. http://www.sengpielaudio.com/Klavier1.pdf.
Sigurdardottir, Asgerdur, interview by Philip Logan. 2019. Email Questionnaire (October 16).
Sound on Sound. n.d. Glossary of Technical Terms. Accessed Novemeber 1, 2019. https://www.soundonsound.com/sound-advice/glossary-technical-terms.
Taylor, John, interview by Philip Logan. 2019. Email Questionnaire (June 25).
Teldex Studio. n.d. Teldex Studio Berlin. Accessed June 21, 2019. http://www.teldexstudio.de.
The Univerisity of New South Wales. n.d. How Does a Guitar Work? Accessed November 29, 2019. https://newt.phys.unsw.edu.au/music/guitar/guitarintro.html.
153 BIOGRAPHICAL SKETCH
Philip Logan is a versatile musical artist with professional experience as a classical and electric guitarist, recording engineer, mixing and mastering engineer, live sound mixer, cinematographer, and photographer. He has composed music for various professional ensembles, including the world music group Omnimusica, and has composer credits and music placements in the film and television industry. He has studied with many world-renowned classical guitarists and pedagogues, such as Stephen Robinson of Stetson University, Stephen Aron of Oberlin
College and The University of Akron, and Bruce Holzman of Florida State University, and he has multiple years of experience teaching music at the high school and university level.
154