tag:blogger.com,1999:blog-2792517408804329062024-03-13T18:17:09.223-05:00Sequence 15Music synthesizers and electronic musicDave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.comBlogger154125tag:blogger.com,1999:blog-279251740880432906.post-30007124004943598752020-05-23T16:59:00.002-05:002020-05-23T17:57:08.835-05:00A few pics of the modularHere's where we are, with the new case. Apologies for the weird angles; it's in a tight space and I was trying to maneuver it so it would not be backlit by a window:<br />
<br />
<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgp6ZUCMnvDsSs4i2ZfYKoikUQOZuCAVfP6_ngjPNcSaOADN2EFGZaApMXixs46srD4tDP8y6X7DrcHej3OIemUuXn2VTwNl3Lph5CuGcwW7WPcupU0W9cHtQqEYNsjA-75puZsTnopU2Q/s1600/20200523_160617.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img alt="Modular" border="0" data-original-height="1600" data-original-width="900" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgp6ZUCMnvDsSs4i2ZfYKoikUQOZuCAVfP6_ngjPNcSaOADN2EFGZaApMXixs46srD4tDP8y6X7DrcHej3OIemUuXn2VTwNl3Lph5CuGcwW7WPcupU0W9cHtQqEYNsjA-75puZsTnopU2Q/s320/20200523_160617.jpg" title="" width="180" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Three vertical rows and two rows set at a 45-degree angle, plus a pseudi-CP row. Approximately 105 units of MU.</td></tr>
</tbody></table>
<br />
<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhkmUWmtgMFEyzqvQ6H7bdq1FXTZW84DEjOgVgYszCbHW6861DdgyxYhkKT7Nv0weuBR9I8PawVpA2RJ_GxSlj-nbnHuO1ax_b4D-is66bVFtXJMD6s0AI5R2MT4LhUUoxziS9gTouIEO4/s1600/20200523_160629.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img alt="Modular, top portion" border="0" data-original-height="900" data-original-width="1600" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhkmUWmtgMFEyzqvQ6H7bdq1FXTZW84DEjOgVgYszCbHW6861DdgyxYhkKT7Nv0weuBR9I8PawVpA2RJ_GxSlj-nbnHuO1ax_b4D-is66bVFtXJMD6s0AI5R2MT4LhUUoxziS9gTouIEO4/s320/20200523_160629.jpg" title="" width="320" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">It's a mixed-format modular, with MU, MOTM, and a few Modern-A. </td></tr>
</tbody></table>
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<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgLfmYmnPwvG8qctd5_POKYyC2VvlzPm5ZBPDShIqknqvYmX-8rxl0ZOWo0s3vs2HvH62M8LKgju_9TX0d0gj_278plxh8Qp8j_z2t_V37-r0kqeknvd_PTdNXDxZT_yubaDYRfqfTIhoc/s1600/20200523_160640.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img alt="Modular, bottom rows" border="0" data-original-height="900" data-original-width="1600" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgLfmYmnPwvG8qctd5_POKYyC2VvlzPm5ZBPDShIqknqvYmX-8rxl0ZOWo0s3vs2HvH62M8LKgju_9TX0d0gj_278plxh8Qp8j_z2t_V37-r0kqeknvd_PTdNXDxZT_yubaDYRfqfTIhoc/s320/20200523_160640.jpg" title="" width="320" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">The angled rows -- mostly still room for improvement. I just got the mounting rail for the bottom row, and the filler panels, installed this week. The empty space above is the CP row; I haven't made a filler panel for it yet.</td></tr>
</tbody></table>
Current lineup:<br />
<br />
<ul>
<li>Top row:</li>
</ul>
<ol>
<li>Synthesis Technology MOTM-650 four-channel MIDI-to-CV interface</li>
<li>Synth Tech MOTM-820 voltage controlled lag processor, built from a kit by me</li>
<li>Synthetic Sound Labs 1130 Ian Fritz voltage controlled ultrasonic oscillator</li>
<li>Synth Tech E350 Morphing Terrarium wavetable oscillator, reformatted to MU by Low-Gain Electronics</li>
<li>Synth Tech E340 Cloud Generator, reformatted to MU by Low-Gain Electronics</li>
<li>SSL 1900 mini oscilloscope, a Dave Jones Design O'Tool reformatted to MU by SSL</li>
<li>SSL 1310 voltage controlled digital delay</li>
<li>Synth Tech MOTM-510 WaveWarper waveform math processor, built from a kit by me</li>
<li>Corsynth C105 Lo-Fi Machine voltage controlled noise generator and downsampler</li>
<li>Synthesizers.com Q130 clipper / rectifier</li>
<li>Encore Electronics frequency shifter</li>
<li>Catgirl Synth CGS57 voltage controlled switched-capacitor filter/oscillator, with panel by Bridechamber; built from a kit and modifications by me</li>
<li>Cynthia StereOSpace stereo field manipulator</li>
</ol>
<div>
<ul>
<li>Second row:</li>
</ul>
<ol>
<li>(3) Synth Tech MOTM-310 micro VCO, built from kits by someone else (I don't know who)</li>
<li>Synthesizers.com Q141 oscillator aid, has behind-the-panel wiring to the adjacent Q016</li>
<li>(2) Synthesizers.com Q016 VCO</li>
<li>Synthesizers.com Q161 oscillator mixer, has behind-the-panel wiring to the adjacent Q016</li>
<li>Synth Tech MOTM-890 5-channel micro mixer, built from a kit by me</li>
<li>STG Soundlabs Post Lawsuit 4075-style VCF</li>
<li>STG Soundlabs Sea Devils VCF</li>
<li>Rob Hordijk Design voltage controlled phase shifter / filter</li>
<li>SSL 1021 Nyle Steiner VCF</li>
<li>Synth Tech MOTM-440 discrete OTA VCF, built from a kit by me</li>
<li>Synth Tech MOTM-410, triple resonant filter; built from a kit and modifications by me</li>
<li>Cynthia Steiner Synthacon filter, with Modcan-A panel reformatted to 1/4" jacks by me</li>
<li>Synthesizers.com Q107 multimode VCF</li>
</ol>
<div>
<ul>
<li>Third row:</li>
</ul>
<ol>
<li>Free State FX Macro Digital Oscillator -- a Mutable Instruments Braids reformatted to MU by FSFX</li>
<li>Synth Tech MOTM-101 noise generator / sample & hold, built from a kit by me</li>
<li>SSL 1250 quad LFO</li>
<li>Synth Tech MOTM-120 voltage controlled LFO, built from. a kit by me</li>
<li>Encore Electronics universal event generator</li>
<li>Synthesizers.com Q108 dual VCA</li>
<li>(2) Synthesizers, com ADSR envelope generators</li>
<li>Synthesizers.com Q179 Envelope++ complex envelope generator</li>
<li>Synth Tech MOTM-190 dual VCA / ring modulator, built from a kit by me</li>
<li>Synth ech MOTM-890 5-channel micro mixer, built from a kit by me</li>
<li>Synthesizers.com Q108 dual VCA</li>
</ol>
<div>
<ul>
<li>CP row:</li>
</ul>
<ol>
<li>Dual 4-way / single 8-way passive multiple, built by me</li>
<li>Gate / trigger generator, built by me</li>
</ol>
<div>
<ul>
<li>Fourth row (slanted):</li>
</ul>
<ol>
<li>Oakley Sound Systems EFG envelope follower / gate generator, built by Krisp1</li>
<li>STG Soundlabs integer divider</li>
<li>Synthesizers.com Q128 dual voltage controlled switch</li>
<li>Synthesizers.com Q119 3-row X 8-step analog sequencer</li>
<li>Synthesizers.com Q123 Standards control voltage generator</li>
<li>Synthesizers.com Q125 control voltage processor</li>
<li>Corsynth C107 dual VCA / mixer</li>
</ol>
<div>
<ul>
<li>Fifth row (slanted):</li>
</ul>
<ol>
<li>Synthesizers.com Q155 Curver non-linear amplifier</li>
</ol>
<div>
<ul>
<li>Not currently installed:</li>
</ul>
<ol>
<li>STG Soundlabs Mankato VCF (the SSM 2164 IC has failed)</li>
<li>Ken Stone / Cynthia Super Psycho LFO (circuit board is de-laminating)</li>
<li>Second Synthesizers.com Q128 dual VC switch (has a short somewhere between +15V and ground)</li>
<li>Several kits in various stages of construction</li>
</ol>
</div>
</div>
</div>
</div>
</div>
</div>
Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-45146376438667955592018-03-24T13:36:00.002-05:002018-03-24T13:36:39.430-05:00InstitutionsThere have been several institutions which have been important in the development of electronic music in the 20th century. Here are brief descriptions of a few of them.<br />
<br />
<h2>
Bell Telephone Laboratories</h2>
<div>
<br /></div>
<div>
Bell Labs, as it was usually known, was established in 1925, as several pieces of the corporate amalgamation known as the Bell System decided to consolidate their research and development efforts. The Labs, created as a joint entity between AT&T and its captive manufacturing company, Western Electric, set up shop in a building in lower Manhattan in New York City. As the labs grew, it began expanding into New Jersey (where land was cheap at the time), and then eventually to a handful of locations around the eastern and central United States, including notably the Chicago area.<br />
<br />
Bell Labs was charged by its owners to perform research and development related to telephony and telephone switching systems, transmission systems, and end-user devices. But prior to 1984, with AT&T enjoying a monopoly on telephone service through most of the USA and its profits being more or less guaranteed by the federal government, funding was available to branch off into basic research in areas only peripherally related to telephony. Eventually this led to several fundamental scientific and engineering advances, including the invention of the transistor, pioneering work in satellite communications, the development of the C programming language and the Unix operating system, and the discovery of the cosmic microwave background radiation (a key discovery in proving the Big Bang theory of the creation of the universe).<br />
<br />
Musically related, research into finding more efficient ways to transmit the human voice led to the development of the vocoder and voder in the 1930s. After WWII, the Labs engaged in some of the first experiments in digital sound processing, leading to pioneering work in computer music by Max Mathews, and later Hal Alles and Laurie Spiegel. Mathews developed the MUSIC series of music-generating computer programs, from which spun off Csound and CMIX, as well as a host of interface devices allowing a performer to interact with the software in real time. In the mid-1970s, Alles, with input from Spiegel, developed the Bell Labs Digital Synthesizer aka the Alles Machine, one of the first digital devices designed specifically to produce music. The Alles Machine combined concepts in frequency modulation and additive synthesis; it directly influenced the design of the Crumar GDS and the Synergy digital sequencer of the late 1970s, and indirectly contributed concepts to the Yamaha DX7.<br />
<br />
Funding for basic research at the Labs dried up after the court-ordered breakup of AT&T in 1984. Owned by Lucent Technologies after the breakup, the Labs wound down activities not directly related to telecommunications, and began divesting itself of some of its research facilities. Today, what remains of Bell Labs is owned by Nokia; it remains headquartered in its Murray Hill, NJ location where it has been since 1966. A few other locations in New Jersey are still open and a few former Labs facilities have been sold intact to other companies. The rest have been closed and the properties sold. The original Manhattan location has been redeveloped into an arts community and is now a National Historic Landmark.<br />
<div>
<h2>
<br />Columbia-Princeton Electronic Music Center</h2>
</div>
<div>
<br />
Composer and Columbia University professor Vladmir <span style="background-color: white;">Ussachevsky became interested in tape studio techniques in the early 1950s, after the university's music department acquired one of the first Ampex tape recorders. In 1957, he and Milton Babbit, a cohort at Princeton University, applied for a Rockefeller Foundation grant to establish an electronic music studio. Babbit was aware of the RCA Mark II synthesizer, and he convinced RCA to loan it out to Columbia. Starting in 1958, the duo began composing on the Mark II and opened the Columbia-Princeton Electronic Music Center, opening it to other composers such as Edgard Varese and </span><span style="background-color: white; font-family: sans-serif; font-size: 13px;"> </span><a href="https://en.wikipedia.org/wiki/Charles_Wuorinen" style="background: none rgb(255, 255, 255); color: #5a3696; font-family: sans-serif; font-size: 13px;" title="Charles Wuorinen">Charles Wuorinen</a>. The Center's focus, as driven by <span style="background-color: white;">Ussachevsky, was always on "serious music" and modern classical composition.</span><br />
<span style="background-color: white;"><br /></span>
<span style="background-color: white;">By 1970, the Mark II was considered obsolete, and the Center turned to computer music. Led by composer Charles Dodge, the Center began using the University's IBM 360 computer to realize digital compositions using various software packages. All-night computer runs were necessary to produce a few minutes of music. To hear the music, the data was written to digital tape and transferred to another computer which was equipped to a digital-to-analog converter, whose output was recorded on analog tape. All of the conversion equipment was built by Columbia engineers. Dodge released several albums of music that he produced this way, and the Center also saw work from other composers such as Alice Shields and Mario Davidosky. </span><br />
<span style="background-color: white;"><br /></span>
<span style="background-color: white;">But by 1985, </span><span style="background-color: white;">Ussachevsky was in poor health and Babbit's interests had turned away from electronic music. Princeton ended its association with the Center, and the facilities fell into disuse. Brad Garton, the current director, reorganized the Center in 1995, bringing in new equipment and new composers, and renaming it the Columbia Computer Music Center. Today, the Center focuses mainly on teaching. The RCA Mark II is still there, but is said to be in poor repair.</span></div>
<div>
<h2>
<br />San Francisco Tape Music Center</h2>
</div>
<div>
<br />
A group of influential West Coast experimental musicians, including Morton Subotnick, Terry Riley and Pauline Oliveros, formed the San Francisco Tape Music Center collective in 1962. As the name suggests, the original focus was on tape manipulation; the collective had little funding and no equipment other than that individually owned by the members. Using facilities provided by radio station KPFA, they presented live performances of mostly pieces played on conventional instruments combined with manipulated tape. However, around 1964, Donald Buchla joined forces with the Center and began bringing in components and prototypes for his initial modular synthesizers, for the other members to try out and critique. With their feedback, Buchla gradually assembled the pieces of what became the first Buchla 100 series modular synth. The completed synth was premiered by the Center in 1966, with Oliveros, Subotnick, Ramon Sender, and Buchla himself performing.<br />
<br />
The Center did not last long after this. Subotnick tried to fix the Center's perpetually short funding situation by obtaining a grant from Mills College (where he was a professor) in 1967. But a condition of the grant was that the center come under Mills' management. This proved stultifying, so much that over the next two years, all of the original members (including Subotnick himself) departed, taking their equipment with them. By 1969, neither any of the original members nor any equipment remained. But the Center's place in the history of electronic music is secured by its role as the crucible of the Buchla modular synths, as well as advancing the careers particularly of Subotnick, Oliveros, and Terry Riley. Subotnick employed the Buchla modular synth to record the canonical electronic music album <i>Silver Apples of the Moon</i> in 1967. </div>
<div>
<br /></div>
<div>
<h2>
BBC Radiophonic Workshop</h2>
<br />
The British Broadcasting Corporation created the BBC Radiophonic Workshop in 1958, as a studio to create electronic theme and background music, and sound effects, for BBC radio and television programming. BBC studio musicians Daphne Oram and Desmond Briscoe had begun using tape studio techniques to produce some music for BBC dramas, and they convinced the network to consolidate all of its electronic audio production into one facility, the Workshop. Over the next four decades, the Workshop would produce music and effects for countless BBC shows, as well as some non-commercial album releases, and serve as an incubator for musicians and engineers ranging from Delia Derbyshire to Mark Ayers.<br />
<br />
Of all of the multitudes of music productions that the Workshop engaged in, it is probably still known best for one of its earliest efforts -- the original theme to the <i>Doctor Who</i> sci-fi show, produced in 1963. Composer Ron Granier wrote out a score and brought it to Derbyshire to execute. Using the typical tools of a tape studio -- a few tape machines, some audio test equipment, and a collection of found objects that were hit, bowed, shaken, twanged, dropped, or coerced to make noise by any means handy -- Derbyshire assembled the theme, using three separate reels of tape, each containing hundreds of splices, and hand-synced together to produce the master tape. Granier, on first hearing the results of his score, famously said, "Did I write <i>that</i>?" The theme, and other music and effects produced for the show, helped make <i>Doctor Who</i> a hit that is now running (with some breaks) into its fifth decade of production. Although the theme has been re-made numerous times for subsequent seasons, some long-time fans still swear that Derbyshire's original is the best, and to this day the show still uses some of the original sound effects, including the Tardis "engine" sound created by Brian Hodgson. Here you can hear the original theme, all two minutes and twenty seconds of it, along with an early version of the opening video sequence:<br />
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<iframe allowfullscreen="" class="YOUTUBE-iframe-video" data-thumbnail-src="https://i.ytimg.com/vi/75V4ClJZME4/0.jpg" frameborder="0" height="266" src="https://www.youtube.com/embed/75V4ClJZME4?feature=player_embedded" width="320"></iframe></div>
<br />
Near the end of the 1960s, the studio began to introduce synthesizers. EMS founder Peter Zinovieff was an acquaintance of several of the Workshop musicians, and the Workshop became an unofficial beta test site for EMS gear, in the same manner that the San Francisco Tape Music Center had been for Buchla. This caused a split between the older and younger musicians, the former of which had been trained on the tape studio techniques (which were closer to what we would think of as sampling today), and the late-1960s analog synths did not suit them. A number of them, including Derbyshire, left the Workshop between 1968 and 1973. However, the younger members carried on and finally managed to pry some money out of the BBC for equipment investments. Zinovieff twisted the Workshop's arm to buy one of the massive Synthi-100 synth-in-a-desk units, and later on Hodgson (who had returned to become the Workshop director after several years away) persuaded the powers that be to buy one of the first Fairlight CMI units -- which, in a way, brought back some of the old tape studio techniques.<br />
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The Workshop continued its good work up into the 1990s, when the BBC went onto a "full cost accounting" basis, and began comparing the costs of the Workshop to the costs of using outside studios and contractors, a comparison on which the Workshop usually came up short. Subsequently, the BBC began layoffs and moving work out of the Workshop. As synthesizers had become less expensive, an institutional studio no longer had an equipment advantage over smaller outside studios and individual musicians. One of the last jobs given to the Workshop, for which it had unique expertise, was cleaning up the audio on old programming -- removing pops, crackles, noise, and bad-splice burbles.<br />
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On April 1, 1998, forty years to the day after its founding, the BBC Radiophonic Workshop closed. Mark Ayers set about archiving all of the Workshop's tapes and produced material, a task at which he continues today. </div>
<div>
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<div>
<h2>
IRCAM</h2>
</div>
<div>
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<div>
This Paris institute for electronic arts stems from an initiative created by French President Georges Pompidou in 1970. Pompidou asked modern classical composer Pierre Boulez to began assembling a place where French composers would have studio space and equipment to work in composition and recording of electronic music. A main focus of the center would be to pair composers (who would not necessarily be knowledgeable of electronics or computer programming) with engineers and technicians who could help realize the composers' ideas. The center would be named IRCAM, which is an acronym for the French<span style="background-color: white;"> <i style="border: 0px; font-family: "Helvetica Neue", Helvetica, Arial, sans-serif; font-size: 14px; margin: 0px; padding: 0px; vertical-align: baseline;">Institut de Recherche et Coordination Acoustique/Musique</i><span style="border: 0px; font-family: "helvetica neue" , "helvetica" , "arial" , sans-serif; font-size: 14px; margin: 0px; padding: 0px; vertical-align: baseline;">, </span></span>which conveniently translates roughly to the English "Institute for Research Coordination into Acoustics and Music".<br />
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It took Boulez several years to raise sufficient funding to acquire space and equipment. The center finally opened in 1977, and straight away focused on computers and digital synthesis, as well as modern classical composition in general. In the 1980s, Miller Puckette created the first versions of what became Max/MSP at IRCAM, and the center maintains an extensive library of music software which is available for download to registered users. The center has also expanded out into aspects of signal processing for industrial and scientific uses.<br />
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</div>
Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-64419420596011615062018-03-09T19:24:00.000-06:002018-03-09T19:24:33.542-06:00Review: SSL 1900 O'ToolAlways wanted an oscilloscope integrated into your modular? Euro modular users have the Doug Jones Design O'Tool, a handy scope with a little color LCD display mounted in a module. Fortunately for us 5U guys, Doug Slocum took it upon himself to reformat some into MU format, and the result is the Synthetic Sounds Labs Model 1900 O'Tool. Since the days of Keith Emerson, modular users have wanted to find a way to mount an oscilloscope and be able to conveniently route signals to it. Problems with this have always included the size and weight of traditional scopes, their incompatibility with the panel formats and mounting methods that modulars use, and signal compatibility issues. (Who has room for a Tektronix 464 in their case? Or the $$$ for a Tek MDO3000? Me neither.)<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgIclLIkKTkBHxGGnsRarJtGo6JcH7co8r67MQFw_HeywgsIoC27CMqXYehC1huL3nqkDvhVcIM5ikU-7KS5GmZIQiV8x6SlIHvClTENWWPaJe8zQUAIJnF-DGdoOsGZQpNTfb5WU_r8P8/s1600/DSC05088.jpg" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" height="400" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgIclLIkKTkBHxGGnsRarJtGo6JcH7co8r67MQFw_HeywgsIoC27CMqXYehC1huL3nqkDvhVcIM5ikU-7KS5GmZIQiV8x6SlIHvClTENWWPaJe8zQUAIJnF-DGdoOsGZQpNTfb5WU_r8P8/s400/DSC05088.jpg" width="95" /></a>The O'Tool solves these problem neatly, and provides many more capabilities than your average Ebay-special analog Tektronix. The O'Tool consists of a digital signal processing system coupled to a color LCD screen, packaged in a modular synth panel format. It is powered from conventional +/- 15V power, and easily accepts the usual modular synth signal levels and types. No heavy CRT, no high voltages, and no four-figure price tag. Available functions consist of scope screens, voltage measurement, frequency measurement, signal level metering, and spectrum analysis.<br />
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This version of the O'Tool is physically packaged as a 1U wide Dotcom/MU format module. When I received it, I was a bit concerned at first because the screen is pretty small, and my eyesight is not what it used to be. However, the contrast and resolution are excellent, and I've had no trouble reading the screen. The screen does take up as much of the width as could fit without structurally compromising the panel (which would make it difficult to package this in an MOTM-format module). There are six input jacks, a pair for each of the input channels, and a pair for an external trigger signal for the scope modes. Each pair is simply wired together; this allows them to be used to "patch through" a signal that needs to go somewhere else, so you can conveniently insert the O'Tool into a patch without needing a mult.<br />
<br />
Underneath the screen is a row of four small pushbuttons. The leftmost one selects the operating mode and screen to be displayed. Pressing the mode button repeatedly cycles through the screens. The other three buttons are "soft keys" whose functions vary depending on the selected screen. Each screen has a small legend at the bottom showing what the soft buttons do in that screen. The available screens are:<br />
<ol>
<li>Single channel voltage/time scope, displays channel 1 only</li>
<li>Dual-channel voltage/time scope, with the two channel signals overlaid. Channel 1 is displayed in red, and channel 2 in green.</li>
<li>Dual-channel voltage/time scope, split screen. Channel 1 is displayed in the top half, and channel 2 in the bottom half.</li>
<li>Bar-graph averaging voltage display, described further below.</li>
<li>VU/peak level meter</li>
<li>Spectrum analyzer</li>
<li>X-Y oscilloscope</li>
<li>Frequency counter</li>
<li>Digital voltmeter </li>
</ol>
<h2>
Scope Mode Screens</h2>
In the first three screens, the three buttons allow the user to select the time per horizontal division, the displayed voltage range, and the trigger source and mode. Pressing each button cycles through the available values (which can be a bit tedious in the case of the time/div setting since there are many possible values). The screen is underlaid with a division grid, shown in dark blue, which appears behind the signal traces. The available time per division settings range from 100 microseconds per division to 5 seconds per division. There are six horizontal divisions across the screen, so at the maximum setting, the time for one screen sweep is 30 seconds. The voltage range differs from oscilloscope convention, and from the time range, in that it applies to the entire vertical span instead of per grid division. Available ranges are<br />
<br />
<br />
<ul>
<li>Plus/minus 5V DC</li>
<li>Plus/minus 10V DC</li>
<li>Plus/minus 10V AC (DC signals/offsets are filtered out)</li>
<li>0-5V DC</li>
<li>0-10V AC</li>
</ul>
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Here are some screen shots of the three scope mode screens. (All of the photos from here through the end of this post were photographed from the actual screen. I cropped the shots, and added some contrast enhancement in order to get rid of room light reflecting off of the screen; otherwise, the photos are unretouched. The somewhat fuzzy look is caused by magnification of the photos, and the fact that I had to use the camera's digital zoom because I don't have a proper macro lens. Bear in mind that these photos are larger than the actual screen. All of the waveforms are from a Synthesizers.com Q106 VCO.)<br />
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This is the single-channel mode, showing a sine wave:<br />
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The dual-channel stacked mode, showing two waveforms from the same VCO. Channel 1 is shown in red and channel 2 is in green. Here, channel 1 (the sine wave) is chosen as the trigger signal.<br />
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The dual-channel layered mode, with the same two waveforms.<br />
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The same screen, but with channel 2 (the sawtooth wave) chosen as the the trigger channel </div>
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The trigger can be set to trigger on either of the two input channels, or on the signal connected to the external trigger input jacks. It can also be set to no-trigger mode, in which the scope free runs.<br />
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<h3>
Triggers and Triggering Modes</h3>
<div>
The concept of triggering, for a scope in general, can be a bit difficult to understand at first. The reason that scopes have triggered modes is to make the waveform "stand still" on the display. Considering what would happen if the scope was free running; that is, if it scanned continuously. Unless the waveform you are trying to display happens to be divisible by the scan rate, the wave won't be stationary on the screen; it will begin in a different place in its cycle on each scan, resulting in a display that jumps around.</div>
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<div>
To solve this problem, a scope has some sort of detection of a certain part or feature and then generates a trigger signal, not unlike the trigger signals that we use in our synths. The trigger causes one horizontal scan to happen; after that scan is completed, the scope waits until it sees the trigger again, and then it does another scan and updates the display, etc. By doing this, the scan always starts at a chosen point in the signal cycle, so that the displayed waveform remains stationary and you can actually look at it. <br />
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The O'Tool can use either input channel as the source to the trigger detector, or it can use the signal at the "Trigger" input There are five trigger modes:<br />
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<ul>
<li>Trigger 1. Uses channel 1 as the trigger source. If the ±5V or one of the ±10V ranges are selected, the trigger is generated when the signal crosses the horizontal axis in the positive-going direction. If the 0-5V or 0-10V range is selected, the trigger is generated when the signal crosses 1.25V in the positive-going direction.</li>
<li>Trigger 2. Same as trigger 1 except that it uses channel 2 as the trigger source.</li>
<li>Ext 0V uses the external trigger input as the trigger source. The trigger is generated when the signal crosses the horizontal axis in the positive-going direction.</li>
<li>Ext 1V is the same except that the tigger is generated when the signal crosses 1.25V in the positive-going direction.</li>
<li>No Trigger is a free-running mode; the scan runs all of the time, unsynchronized to the input signals. </li>
</ul>
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<div>
The trigger selection allows you to select either channel to be fed to the trigger generator. You can even do this in the single-channel mode; you can display channel 1 and trigger off of channel 2. The display may look different depending on which channel you trigger from. Consider the screen shots of the two stacked-mode screens above. In the top one, the trigger is on channel 1, so it triggers when the sine wave crosses the X axis going up. The nature of the Q106 VCO (as with most sawtooth-core VCOs) is that the positive peak of the sine wave is where the positive peak of the sawtooth wave is. So the top half starts with the sine wave heading up from zero towards its peak, while the sawtooth display starts with the last 90 degrees of its cycle. In the second photo, we switch the trigger to channel 2. Now we are triggering on the positive-going zero crossing of the sawtooth, which is pretty much instantaneous. So we see the display start with both of the waveforms descending from their peaks. </div>
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<h2>
Level Displays</h2>
</div>
The bar graph display is interesting but kind of hard to describe. Basically, what it does is show how much time -- what percentage of the cycle -- a signal spends at a given voltage level. The more the signal is at that a given level, the brighter the bar will be at that level. In the shot below, channel 1 is a square wave and channel 2 is a sawtooth. The square wave, of course, alternates sharply between the positive and negative peaks; hence the two discrete bars. The sawtooth falls linearly and so all of the voltage steps get the same saturation, resulting in a spread of evenly lit bars. (Not sure why the top one is a bit dimmer; may have to investigate how the sawtooth waveform is looking coming out of that VCO.) The display range can be adjusted, and "fast" or "slow" averaging can be selected.<br />
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The VU and peak level meters do what you expect them to: display the average and peak voltage level of an alternating signal. The display shows VU and peak levels for both of the input channels; the VU displays are grouped on the left, and the peak displays on the right. The VU indicators appear to be a true RMS measurement, as they display identically to the peak levels when a sine wave is input. I cannot say, however, that the ballistics of a true VU meter are emulated properly; I don't have any means to measure it. There are three selectable scale modes, which effect what reference level is used for the meters, and how the scale on the peak meters is displayed. <br />
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Like many such meters, the display uses color bars to display different regions of the measurement levels. The blue horizontal line indicates the reference signal level (the level that is considered a "100%" signal) for whatever scale mode is in use. Levels below and at the line are displayed using green bars. Above the line, on the peak side, the first three steps are displayed using yellow bars, and levels above that are displayed with red bars. On the VU side, all levels above the line are displayed using red bars.<br />
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This screen has three modes, which effect where the "100%" line is, and how the peak display is scaled. The modes are:<br />
<ul>
<li>+4dBu: In this mode, the VU scale conforms to the standard recording industry definition, in which zero VU = +4 decibel volts RMS, or dBu. This in turn is defined as 1.228 volts RMS. (It's defined at 1000 Hz, but that is not supposed to matter across most of the audio range. I'll have more to say about this further down.) The peak scale displays dBu and the blue line will pass through +4 on that scale. (It always passes through zero on the VU scale.) Red bars on the peak scale start at +8 dVu.</li>
<li>+2.5V: In this mode, zero on the VU scale corresponds to 2.5V RMS. The peak scale will be re-scaled to show voltages up to 10 volts, and the blue line will pass through the 2.5V mark. Red bars on the peak scale start between the 3.5V and 5V marks.</li>
<li>+5V: In this mode, zero on the VU scale corresponds to 5V RMS. The peak scale will be re-scaled to show voltages up to 10 volts, and the blue line will pass through the 5V mark. Red bars on the peak scale start between the 7V mark and the 10V mark.</li>
</ul>
An issue with the VU/peak display is that it does some preliminary high-pass filtering before it processes the signals for display. This is common for VU meters; it prevents a DC offset in the signal from creating a false high reading. However, it prevents the meters from working properly with low-frequency signals. If you want to look at levels from an LFO, use the bar graph display, or one of the scope screens.<br />
<h2>
X-Y Display</h2>
The X-Y display emulates a feature of many of the old analog scopes, in which the X-axis, which is normally controlled by the scope's time base, can instead be driven by an external signal, producing two-dimensional patterns on the scope screen. In this implementation, channel 1 drives the X axis and channel 2 drives the Y axis.<br />
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The old analog scopes depended on the persistence of the display phosphor for the user to be able to perceive the drawn figures. The O'Tool attempts to emulate that with a setting that defines the "persistence" of each dot drawn; the dot is removed from the display after the equivalent of what would be that amount of time has passed, which determines how long each part of the figure remains on the display (which is, of course, also a function of the frequencies of the two waveforms driving the display). To my eye, it doesn't work all that well; the continuously redrawn form is hard to perceive at faster settings, and it quickly fills the entire screen at slower settings. The photo below was taken at a 1/15 second exposure and captures more of the drawn figure (which is made from a triangle wave driving the X axis and a sine wave on the Y axis) than was visible to the eye in real time.<br />
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<h2>
Spectrum Analyzer</h2>
The spectrum analyzer surprised me with how well it works. The update rate is pretty fast, and it seems to not have much of a problem with quantizing noise. It has two display modes, "linear" and "log". In the linear mode, there are four frequency ranges available, with a choice for the upper end of 20, 10, 5, or 2.5 KHz. Vertical scaling is relative, but you can choose from 1x up to 4x. If, in one of the higher vertical magnifications, one of the peaks exceeds the vertical range, the peak displays a red top, as you can see in the shots below.<br />
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This one is with a square wave on channel 1 (top) and a sawtooth on channel 2 (bottom.) On the square wave, you can see the odd-harmonics pattern typical of square waves. <br />
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This one is with a 25% pulse wave on channel 1, and a triangle on channel 2. Notice how little harmonic content the triangle has. <br />
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I have found the log mode to not be as useful in general, because it groups all of the frequencies into octaves and displays one bar per octave. Depending on the range setting, it displays between 5 and 7 octaves. Here's an example; unfortunately, I forgot to write down what waveforms I was using for this shot.<br />
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<h2>
Frequency Counter</h2>
The frequency counter is straightforward and works well. You can select channel 1 only, channel 2 only, or both. There are three elements on the display. At the top is the frequency, in Hz, for each channel. Not having a calibrated frequency source, I can't really speak to whether these are actually precise to two decimal places. In the middle, it displays the closest equal-tempered note for the frequency of each channel, and how far away in cents it is from the ideal equal-tempered value. The bottom portion shows this deviation graphically.<br />
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For the note display, you can select concert A to be either be 440 Hz (the usual standard) or 432 Hz. There is a lot of nonsense surrounding 432 Hz tuning; there's nothing special or magical about it. Prior to the 20th century, orchestras were all over the map as to what standard they tuned to; Bach, Beethoven and Haydn are thought to have used an A of about 422 Hz. Nonetheless, if you want to try something different, and you're using the O'Tool to tune instruments, you can give 432 Hz a try. Note that some polysynths may not be capable of being tuned this far off of 440 Hz.<br />
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<h2>
Voltmeter</h2>
\The last display is a simple voltmeter, displaying the voltage present on each channel. Only DC voltages can be displayed. Keep in mind that the O'Tool is (in this case) being powered from a 15V supply; it most likely cannot display voltages exceeding the supply rails, and trying to do so could potentially damage it. (I haven't tried.) So don't use it to check the supply voltages on your Roland System 700.<br />
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<h2>
Conclusions</h2>
<div>
The O'Tool is a useful thing to have in your setup. And it looks cool.</div>
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<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-48728043393250193972018-02-19T14:31:00.001-06:002018-02-19T14:32:46.306-06:00Review: Synthesizers.com Q119 Analog Sequencer<div style="margin-bottom: 0in;">
The Q119 from Synthesizers,com is a
24-step analog sequencer. If you haven't used an analog sequencer
before and don't know what its purpose is, it's a device that stores
a set of control voltage values, and sends them to an output one
after the other, under the control of a clock signal. As is the case
with many analog sequencers, the “storage” for the control
voltages consists of a set of knobs, each of which selects a control
voltage within a given range. If you've listened to early Tangerine
Dream or any other “Berlin school” electronic music, you've
doubtless heard note sequences produced by an analog sequencer
connected to the control input of a VCO. Repeating control voltage
patterns have a huge variety of other uses, such as controlling
filters, switching between different signals via connections to VCAs,
and even using the output as an audio signal when the clock rate is
high enough.
</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
Like all Synthesizers.com products, the
Q119 is formatted in the MU (Dotcom) format, which means it uses 1/4”
jacks for all signal connections, and the standard Dotcom six-pin
MTA-100 connector for power. (It does draw from the +5V power; the
power supply must supply that voltage in order for the Q119 to
function.) At a width of 8U, it is one of the physically largest
modules that Synthesizers.com offers. The panel is divided into
three basic sections: The section on the left has the clock controls
and the various option switches that change the way the sequencer
works. The middle and largest section consists of the 24 step
controls, each having a control voltage tuning knob and an LED
indicator. The section on the right is the output section, with the
row outputs, and the master outputs with their offset and lag
controls.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEixxdlgolVeGxCsn1wRy5UJcGpeZhJ4lqYVsIrRjO38DLjhwUnuB4EngrQTv8XnktZzlZiwOnlFTOUTB0p9lnc1ZIxrV-haUwzPSdgb-OkEx_CxpWTrGz8GXjngC-9XL_kR-T3W5v2jyFY/s1600/DSC05057.JPG" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEixxdlgolVeGxCsn1wRy5UJcGpeZhJ4lqYVsIrRjO38DLjhwUnuB4EngrQTv8XnktZzlZiwOnlFTOUTB0p9lnc1ZIxrV-haUwzPSdgb-OkEx_CxpWTrGz8GXjngC-9XL_kR-T3W5v2jyFY/s320/DSC05057.JPG" width="320" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Synthesizers.com Q119 analog sequencer, with a single-width Q128 A-B switch shown next to it for size comparison.</td></tr>
</tbody></table>
<h2 style="margin-bottom: 0in; text-align: left;">
</h2>
<h2 style="margin-bottom: 0in; text-align: left;">
Clock Rate, Start/Stop, and Cycle Controls</h2>
<div>
<br /></div>
<table cellpadding="0" cellspacing="0" class="tr-caption-container" style="float: right; margin-left: 1em; text-align: right;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhaM0FKJD8Ceal1mAmiwLh5Cf1a8uLcosCB3WCrxzUjv9vqyeJ1E8lUl7-LRj0-BRgjDS_3OIAddppLdlSZF310ucrSYI_nRoHOShKXn7NbSLXMDrw98g6J8tOkuhwSFvUw9S3I2OjrZpg/s1600/DSC05060.jpg" imageanchor="1" style="clear: right; font-size: 12.8px; margin-bottom: 1em; margin-left: auto; margin-right: auto; text-align: center;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhaM0FKJD8Ceal1mAmiwLh5Cf1a8uLcosCB3WCrxzUjv9vqyeJ1E8lUl7-LRj0-BRgjDS_3OIAddppLdlSZF310ucrSYI_nRoHOShKXn7NbSLXMDrw98g6J8tOkuhwSFvUw9S3I2OjrZpg/s320/DSC05060.jpg" width="180" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Cycle option switches at the top,<br />
clock controls at center,<br />
start/run/stop controls at bottom</td></tr>
</tbody></table>
<div style="margin-bottom: 0in; text-align: left;">
In the clock section, the most
prominent controls are the oscillator frequency (RATE) knob and the GATE
WIDTH knob. The RATE knob and the adjacent RANGE switch control the rate of
the internal clock. With the knob full CCW and the RANGE switch on
LOW, the slowest available rate is about 3 Hz, which to me is not
slow enough. If you want slower, you have to use an external clock,
The fastest available rate, with the RANGE switch on HIGH, is about
320 Hz. To the left of this knob is the external clock input and the
SOURCE switch. As you might guess, when the SOURCE switch is in
EXTERNAL, the internal clock is disconnected and the sequencer is
driven by a clock signal received at the external clock input. This
input should be a pulse wave (although the sequencer will square it
up if it isn't), and the sequencer advances on the leading edge.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
When the internal clock is being used,
the GATE WIDTH control determines the “on” time of the gate
outputs, as a duty cycle percentage (which means that as the
frequency gets faster, the gate on time gets shorter).
Unfortunately, the one on my Q119 does not work (I bought this unit
used); it produces gates that are about 1 ms wide regardless of what
I set the knob at. Fortunately, when an external clock is used, the
gate on time follows the pulse width of the external clock; the GATE
WIDTH control is ignored. This means that if you are driving the
Q119 with a VCO that has pulse width modulation, you can change the
gate “on” time by adjusting the VCO's pulse width, or better yet,
make the gate “on” time voltage controlled by feeding a control
voltage to the VCO's pulse width input.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
The start/stop controls at the bottom of the clock section consist of four pushbuttons and three associated input jacks (one for each button except SET END). The START button, when pressed, causes the sequencer to start; it then runs continuously (unless the the SINGLE / CONTINUOUS switch is in the SINGLE position), until the STOP button is pressed. The GO button causes the sequencer to run as long as the button is held; when the button is released, it stops. The jacks under the START and STOP buttons accept trigger signals; receiving a signal on one of these jacks has exactly the same effect as pushing the associated button. The jack under the GO button accepts a gate input; the sequencer will run as long as the gate signal is high. The SET END button, we'll cover in a minute.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<h3 style="margin-bottom: 0in;">
How fast will it run?</h3>
<div>
With an external clock, I tested mine to see how fast it would run, and it made it up to 920 Hz; faster than that, and the sequencer freezes. (Synthesizers.com's documentation only says that it will run “up to” 1 Khz.) This means that you can, in effect, use the Q119 as a sort of function generator at low audio rates; at this speed, a full 24-step sequence will cycle at about 38 Hz, and faster if you make the sequence shorter. There is no limit on the slowest rate; you can unplug the cord from the external clock jack, and the sequencer will simply wait until you plug it back in. When the sequencer is stopped, pressing the MANUAL STEP button next to the RATE knob causes the sequencer to advance one step. This is normally used to tune steps when setting up a sequence, but it can be used to “clock” the sequencer manually.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<h3>
Cycle options</h3>
<div style="margin-bottom: 0in;">
The four switches across the top select
various options for the sequencer's operation. The MODE switch,
I'll cover in the next section where we go over the step controls.
The voltage range OUTPUTS switch sets the minimum and maximum range
of the step tuning knobs. When the switch is in the -5 / +5 mode,
turning a step knob full CCW causes tha step to output -5V, and full
CW outputs +5V; the 12 o'clock position outputs 0V. When the switch
is in the 0 / +5 mode, full CCW on the step knob outputs 0V. (The 12
o'clock position doesn't output 2.5V; I'll say more about this
later.)
</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
When the CYCLE switch is in the SINGLE
position, the sequencer always stops on the last step in the
sequence. To make it run again, a START operation has to be
performed again. In the CONTINUOUS position, as you might expect,
the sequencer runs in a continuous loop until you stop it. (Note
that when the “hidden” random mode is selected, this switch is
ignored; the sequencer always runs continuously until stopped.). The
SEQUENCE switch, when in the UP/DOWN position, causes the sequencer
to reverse direction when it reaches the last step in the sequence,
and again when it gets back to step 1. If the configured length of
the sequence is 6 steps, then after step 6 the next steps will be 5,
4, and so on, back to 1. At that point it will again change
direction and count through 2, 3, etc. When the up/down mode is
selected, and the CYCLE switch is in the SINGLE position, the
sequencer stops when it returns to step 1.
</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
The SET END button serves two purposes.
Its primary function is to allow you to set the desired length of a
sequence. This is done by pressing the SET END button once and
releasing it; the LED for either step 1 or the current end step will begin to flash rapidly.
Repeatedly press the SET END button to advance the end step (you have to do it quickly); when it
reaches the step you want, stop pressing the button. After a second
or two, the flashing will stop, and then that step will be the final
step in the sequence. This is effective for all sequence modes -- up,
up/down, and random. Note that when you switch the sequencer to 3x8
mode, it will automatically set step 8 as the end step. When you
switch back to 1x24 mode, step 8 will remain the end step, and you
will have to use SET END to reset it to a longer sequence if you
want. (Or cycle the power.) <br />
<br />
The SET END button is used with
the MANUAL STEP button to select two "hidden" modes of the
sequencer. The normal start mode is the "reset" mode; in
this mode, any time the sequencer starts, it first resets to step 1.
Pressing MANUAL STEP while pressing and holding SET END selects the
"continue" mode. In this mode, when the sequencer starts,
it resumes with the step after the one it stopped on. Doing the
opposite of that – pressing SET END while pressing and holding
MANUAL STEP -- sets the cycle mode to the random mode. In this mode,
each time the sequencer advances, it selects a step at random.
Although I haven't attempted to do an analysis of the distribution,
it seems to be pretty uniform. One thing to note is that the code
presents the same step from being selected twice in a row. This is a
nice feature when generating random notes; in a random-note sequence,
it tends to be jarring to the listener to hear the same note sound
twice. The CYCLE and SEQUENCE switches have no effect when the random
mode is engaged; the sequencer runs continuously until stopped.
Either of these hidden modes may be disengaged by repeating the
button sequence for that mode, or by cycling the power. </div>
<div style="margin-bottom: 0in;">
<br /></div>
<h2>
Step Controls</h2>
The
heart of the Q119 is in the 24 step blocks, which are organized in
three rows of 8 steps each. Each step block consists of a single
knob, which is used to select the output voltage for that step, and a
red LED that indicates when the block is active. To improve finger
room for the knobs, the odd-numbered steps have the knob on top and
the LED on bottom, while the even-numbered steps are the reverse.
This results in a rather amusing pattern of lights moving in a
zig-zag when the sequencer is running, which some performers object
to, but I think it actually improves recognition of which step is
active. The LEDs also function with the SET END button in selecting
which step is to be the last step in the sequence. Changing the
setting of a knob will be reflected immediately in the output if the
sequencer is on that step (the step's LED is lit), whether running or
stopped.
<br />
<div style="margin-bottom: 0in;">
<br /></div>
<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhU4Xslum4-JpC6N3Lvb90Vey9fKhCoYZPDvxhyRwsKYka_vyDqgnpvC_7hCzP8bJGNJn0OQ5tfe2lnPAIFoujhOlWUUmvQsAbteD9fykwpBqkwW9L-5CtEvzQhRBtGsmIwR-Pd2aSIRJE/s1600/DSC05061.JPG" imageanchor="1" style="font-size: 12.8px; margin-left: auto; margin-right: auto; text-align: center;"><img border="0" height="180" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhU4Xslum4-JpC6N3Lvb90Vey9fKhCoYZPDvxhyRwsKYka_vyDqgnpvC_7hCzP8bJGNJn0OQ5tfe2lnPAIFoujhOlWUUmvQsAbteD9fykwpBqkwW9L-5CtEvzQhRBtGsmIwR-Pd2aSIRJE/s320/DSC05061.JPG" width="320" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Q119 step controls and LEDs, with row outputs on the right.</td></tr>
</tbody></table>
<div style="margin-bottom: 0in;">
</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
The organization of the step blocks
into three rows is not merely a visual presentation. The Q119 has
two operating modes, known as “1x24” and “3x8”, and selected
by the MODE switch. In the 1x24 mode, the sequencer drives a single
sequence of up to 24 steps long, using the three rows in series.
When the sequence runs, it will proceed across the top row until it
reaches step 8, then resume on the second row at step 9, going to 16
and then jumping to the third row at step 17. At step 24, it jumps
back to the first row and step 1. In the 3x8 mode, the sequencer
drives the three rows in parallel, producing three sets of control
voltages at the three BANK outputs. The first step is steps 1/9/17,
then it proceeds to 2/10/18, and so on, up to 8/16/24, at which point
it returns to 1/9/17. The LEDs for the proper steps in each row will
light simultaneously, as opposed to the 1x24 mode, in which only one
LED is lit at a time. (In either mode, the SET END button can be
used to make the sequence shorter than the maximum, if desired.)
</div>
<div style="margin-bottom: 0in;">
<br /></div>
<h3 style="margin-bottom: 0in;">
Control voltages</h3>
<div>
The control voltage knobs are not
linear with respect to output voltage. With the OUTPUTS switch in
the -5/+5 position, one might expect that the zero position (12
o'clock) is 0 volts, and each major hash mark is a difference of one
volt. The first statement is true, but the second is not. From 0 to
+1 on the indexing is a difference of about 0.6V. The steps get
larger moving further away from the zero position, finally reaching
plus or minus 5V at the +5 and -5 positions respectively. With the
OUTPUTS switch in the 0/+5 position, something similar happens: the
full CCW position (-5 in the indexing) is 0V; -4 is about 0.3V, -3 is
about 0.7V, and so on. In both modes, the steps get larger as you
move farther away from 0V. This is something of a benefit if you can
use the ADD offset control (further down) so that you can keep most
of the steps near the 0V position, which makes it easier to make fine
adjustments. However, it is confusing if you expect to be able to
look at the indexing and dial up a desired voltage; that isn't
straightforward. If you need a specific voltage, it is best to check
it with a voltmeter. If you are running the output into a VCO and
trying to tune notes, it is usually better to either let the sequence
run and tune it by ear, or if that doesn't work for you, single-step
the sequencer with the MANUAL STEP button and check each note against
a tuner. </div>
<div style="margin-bottom: 0in;">
<br />
<table cellpadding="0" cellspacing="0" class="tr-caption-container" style="float: right; margin-left: 1em; text-align: right;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgtnw4s_SiSng29jofg2q2NhVASKNPP-jUX9ORX1Lr1Bo624jmuZCMx4CEDs1b-A8M3IpxL-3b7MrxDVg7rWvzjEyr7XtO5lKbYgia-XvG76ZY0tru9cogqe6PRTG6zLKYNZQapo3cLC-w/s1600/DSC05062.jpg" imageanchor="1" style="clear: right; margin-bottom: 1em; margin-left: auto; margin-right: auto;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgtnw4s_SiSng29jofg2q2NhVASKNPP-jUX9ORX1Lr1Bo624jmuZCMx4CEDs1b-A8M3IpxL-3b7MrxDVg7rWvzjEyr7XtO5lKbYgia-XvG76ZY0tru9cogqe6PRTG6zLKYNZQapo3cLC-w/s320/DSC05062.jpg" width="180" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Output section, with row (bank)<br />
outputs on the left, and the master<br />
outputs on the right.</td></tr>
</tbody></table>
</div>
<h2>
Outputs</h2>
The output section contains the master outputs, a
set of row outputs for each row (labeled BANK 1/2/3), a knob for
adding lag (portamento), and a knob and jack for adding an offset
voltage to the master output. The master output is usually used when
the sequencer is operating in the 1x24 configuration. The master
OUTPUT jack outputs the voltage from the currently active step. The
GATE jack outputs a gate which rises when the sequencer advances to
the next step, and falls some time after, as determined by the GATE
WIDTH knob in the control section (or the pulse width of the external
clock, if an external clock is being used). The LED next to the GATE
jack lights when the gate is active. If the MODE switch is in the
3x8 mode, the master OUTPUT jack will have the sum of the active
steps from each row. This isn't usually what you want, but it does
have creative possibilities. Note that the OUTPUT jack is active all
of the time, including when the sequencer is stopped. The GATE
output remains low when the sequencer is stopped.<br />
<br />
The row
output jacks are active when the corresponding row is active. When
the sequencer is in the 3x8 configuration, the top-row OUTPUT jack
outputs the voltage selected from the currently active step in that
row, and the other two row OUTPUT jacks perform the same function for
their rows. All of the GATE jacks pulse together in this mode. In the
1x24 mode, the row output jacks are only active for the row that
contains the currently active step. When the current step is not in
that row, the OUTPUT jack outputs the minimum voltage (0V or -5V
depending on the OUTPUTS switch setting), and the GATE jack remains
low.<br />
<div>
<br />
<h3>
Master output modifiers</h3>
The GLIDE and ADD knobs only effect the control voltage master output.
The GLIDE is a conventional lag processor that acts on the control
voltage output. The ADD knob adds an offset voltage to whatever voltage is present at the master output; this has a number of obvious uses, such as transposing sequenced notes, or bringing them in tune with another instrument. If a cable
is plugged into the ADD INPUT jack, that is also added to the master output. To sum it up, the voltage at the master output consists of:<br />
<div style="margin-bottom: 0in;">
</div>
<ul>
<li> The current step control voltage (or the sum of the three steps, in the 3x8 mode)</li>
<li> The ADD knob voltage</li>
<li> The signal present at the ADD INPUT jack</li>
</ul>
<h3 style="margin-bottom: 0in;">
Interaction with another sequencer</h3>
<div>
The DONE OUTPUT jack sends a trigger
signal at the time that the sequencer advances from the last step
back to step 1 (or would have, except for the CYCLE switch being in
the ONCE position). This allows you to operate two (or more!) Q119s
in a round-robin fashion, by setting their cycle switches to ONCE,
and then patching the DONE output of one into the START input of the
next. When the first one finishes, it will start the second one, etc.
By careful adding of the outputs, you can create sequences of 48 or
more steps. (You could take the master OUTPUT jack of one Q119 to the
ADD INPUT of the next one to combine the control voltages, but you'd
need some external module to combine the gates.)</div>
<br />
<h3 style="margin-bottom: 0in;">
Interaction with other modules</h3>
<div>
Some performers who use an analog
sequencer to produce note sequences find it easier to set up the
sequencer when they can run the outputs through a quantizer.
Synthesizers.com offers a quantizer, the Q171, which has features
designed to make it complementary to its sequencers. In particular,
it has three quantization channels, so that you can quantize all
three rows when using the 3x8 mode, and it has gate inputs to force
quantization to only occur on the note gates, which can help avoid
the “dithering” problem (where the quantizer jumps back and forth
between adjacent notes). However, other quantizers could certainly
be used. </div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
Output selector switches, such as the Q962, have potential uses with the Q119. The DONE OUTPUT can possibly be used to cycle between different bus selections or outputs, for various purposes.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
</div>
<h2>
Conclusions </h2>
<div>
It seems a bit unfair to describe the Q119 as an
“entry level” sequencer, since it is a quite capable module. It
is not as full featured as, say, the Moon Modular 569, the GRP R24,
or Synthesizers.com's own Q960. Then again, it also costs a lot less
than those others; the direct-sale price of $560 USD is a bargain in
the world of analog sequencers, which generally tend to be expensive.
(Moon's direct-sale export price, excluding VAT, for the 569 is
E1258.77, which at the exchange write on this date, 7 Feb 2018, works
out to $1545.41 USD.) The main thing that those sequencers have that
the Q119 lacks is flexibility; they typically have features like
individual gate outputs for each step and reset trigger inputs. Then
again, they sometimes require either additional aid modules or fancy
patching to perform functions that the Q119 has built in. So yes,
the Q119 is a good choice for someone who has no experience with
analog sequencing and wants to get practice with it, but it's a
module that will continue to be useful in your case even after you
purchase one of the higher-end sequencers.</div>
<br />
<div style="margin-bottom: 0in;">
<br /></div>
<h2 style="margin-bottom: 0in;">
Demonstration videos</h2>
<div style="margin-bottom: 0in;">
<br />
This first video is a basic demonstration of the Q119's different cycle modes. The 1x24 and 3x8 modes are demonstrated at different speeds, with up, up/down and random sequencing, and the single and continuous cycle options. The use of the SET END button is also demonstrated.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
<div class="separator" style="clear: both; text-align: center;">
<iframe width="320" height="266" class="YOUTUBE-iframe-video" data-thumbnail-src="https://i.ytimg.com/vi/02q6l3kSXJY/0.jpg" src="https://www.youtube.com/embed/02q6l3kSXJY?feature=player_embedded" frameborder="0" allowfullscreen></iframe></div>
<br /></div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
This second video illustrates using
the Q119 in the 1x24 mode, with a sequence length of 14 steps, to
generate an approximation of a familiar Synergy sequence (the one from which
this blog takes its name). Driven by a pulse wave from a Q106 VCO in
LFO mode, it is modulating another Q106, whose triangle output is
going into an MOTM-440 OTA filter. Envelope is from a Q170
Envelope++, and it is controlling a Q109? VCA. Note that this actual patch is only an approximation of the original, for demonstration purposes. Please excuse the
rough tuning; I don't have a quantizer and I didn't spend a lot of
time on tuning the notes. Nonetheless, if you listen to much
Synergy, you should recognize it. I use an external clock and gradually speed up the sequence, in the same manner as the original. Just before the end, I take it up to a faster speed than Larry Fast's old Moog 960 was capable of, just to show off the Q119 a bit.<br />
<br />
<div class="separator" style="clear: both; text-align: center;">
<iframe width="320" height="266" class="YOUTUBE-iframe-video" data-thumbnail-src="https://i.ytimg.com/vi/SUWSjUUMR1M/0.jpg" src="https://www.youtube.com/embed/SUWSjUUMR1M?feature=player_embedded" frameborder="0" allowfullscreen></iframe></div>
<br />
You will notice something at the start of the video: there seems to be a "skip" at the very start of sequence, between the first and second notes. This is due to the fact that I'm using an external clock in this video. (When I reach to something above the top of the picture, I'm reaching for the requency control of the Q106 that is serving as the clock source.) The Q119 syncs its own clock when it is instructed to start, but it has no way of making an external clock sync to it. So when I start the sequencer, it starts at some random point in the external clock's cycle. If this is part way through the cycle, then the first step will be short, time-wise, and that is what you hear here: the first step occurs on the START button press, and then the next step occurs on the next clock transition, but I hit START at some point in the middle of the clock cycle, so the interval between the first step and the second step was short. If I had wanted that interval to be precise, I could have watched the LED on the Q106 and pressed START at the start of the cycle. Or I could have fed an external trigger source to the Q119's START jack, and to the Q106's hard sync input. </div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
This third video illustrates using the
sequencer in 3x8 mode. What is happening here is that the top row is
being used to modulate a Q106, whose sawtooth wave is going into an
SSL 1310 digital delay that is being modulated by an LFO. (There is
no filter in the patch.) The bottom row is being used to generate a
gate signal – turning the knob up causes the gate to be “on” on
that step, and turning the knob down causes it to be “off”, so
that that step does not sound. As the sequence plays, I play with
the bottom row to make different notes in the sequence sound.</div>
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Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com1tag:blogger.com,1999:blog-279251740880432906.post-25610423304027758092015-11-01T23:14:00.002-06:002015-11-01T23:14:39.837-06:00Review: SSL 1250 Quad LFO<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhOyccMKZ9Nfgt4A6bdsuuGdo-Na8LT0slGFr0ocG35sHtrcHNXOQe9_gKt0-hQtnfhrFlNMawoq-y8k6YyKP9Jg6krVHPkanPxmO7Zxrdgc5A1H4weKl3JxnbKhjoR2oQdnXeClea4gYU/s1600/DSC00479.jpg" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhOyccMKZ9Nfgt4A6bdsuuGdo-Na8LT0slGFr0ocG35sHtrcHNXOQe9_gKt0-hQtnfhrFlNMawoq-y8k6YyKP9Jg6krVHPkanPxmO7Zxrdgc5A1H4weKl3JxnbKhjoR2oQdnXeClea4gYU/s320/DSC00479.jpg" width="74" /></a></div>
The Synthetic Sound Labs Model 1250 Quad <a href="http://electronicmusic.wikia.com/wiki/Low_frequency_oscillator" target="_blank">Low Frequency Oscillator</a> is what it says it is: four LFOs in one panel, formatted in the <a href="http://electronicmusic.wikia.com/wiki/Dotcom" target="_blank">MU (Dotcom)</a> <a href="http://electronicmusic.wikia.com/wiki/Modular_synthesizer" target="_blank">modular synth</a> <a href="http://electronicmusic.wikia.com/wiki/Format" target="_blank">format</a>. It's a pretty simple module. The panel is divided into five sections: four sections are each for one LFO, and a bottom section contains the output jacks. Each LFO has three controls: a rate knob, a <a href="http://electronicmusic.wikia.com/wiki/Waveform" target="_blank">waveform </a>select switch (<a href="http://electronicmusic.wikia.com/wiki/Sine_wave" target="_blank">sine </a>and <a href="http://electronicmusic.wikia.com/wiki/Square_wave" target="_blank">square </a>waves are available), and a peak indicator lamp which is also a pushbutton. Pressing it switches the LFO between high range and low range. The lamps are red LEDs and actually look much redder than in the picture to the right; I think the infrared filter on my camera prevented the deep red from registering.<br />
<br />
This is an LFO meant to drive slow, evolving <a href="http://electronicmusic.wikia.com/wiki/Patch" target="_blank">patches</a>. On the high range, with the knob full clockwise, the period is about 22 milliseconds, which works out to 45 Hz. With the knob at 5 (straight up), the period is 50 ms, or 20 Hz. As you turn the knob further left, the period increases linearly, which per the law of reciprocals means the frequency decreases exponentially. With the knob at 2 (the 9 o'clock position), the period is 150 ms, a frequency of 6.6 Hz. At the low end of the knob's travel, between 0 and 1, the change is much more than linear -- with the knob full CCW, I measured a period of 80 seconds.<br />
<br />
If you want really slooooooooow, switch to low range. With the knob full clockwise, the period is about 1200 ms, or around 0.8 Hz. At the 5 setting, it's 3 seconds. At the 2 setting, it's 8.5 seconds. At the 1 setting, it's 36 seconds. With the knob full counterclockwise… I was not patient enough. After three minutes, it had climbed from zero volts to +0.45V. If I've done my math right, that's a cycle time of about 45 minutes! The cycle indicator light starts to light up when the sine wave rises +1.5V, and reaches full brightness by +3.5V; it goes out when the sine wave drops below 1.5V. (This is true whether the sine or square wave is selected.) At moderately slow rates, it's rather hypnotizing to watch. I did a quick check of all four oscillators to make sure they were all calibrated the same, and didn't see any noticeable differences. <br />
<br />
Looking at the waveforms on the scope: The square wave looks good. The sine wave is a bit distorted; it looks a bit <a href="http://electronicmusic.wikia.com/wiki/Triangle_wave" target="_blank">triangle</a>-ish. There's a distinct corner at the turn point, and the rise and fall portions look a bit straight-lined on either side of the horizontal axis. (A perfect sine wave is straight only right on the axis; it has at least a little bit of curvature everywhere else.) It's not as bad as that makes it sound; most of the waveform looks like a good sine wave, and using it to modulate a VCO, I didn't hear any abrupt reversals in pitch rise and fall, as one would if the VCO were modulated with a triangle wave. Also, the sine wave doesn't quite make it to the 5V rails; it turns at about +/- 4.5V. The square wave looks good. There are no visible changes or variations in the waveform with frequency. <br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEidNLVS05mOADQXVR-mj4sVCTobco9b7P6AFyo9j5n7lAINu5b86DYU4daw9GE2swKw_6QRfpsB96HLbE5Uyl-RVHTP4EWop3Fxv0Y66oLTeLKIqp28rA0HWx82L-fMLGPFHuMLrWPC7FM/s1600/DSC00460.jpg" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEidNLVS05mOADQXVR-mj4sVCTobco9b7P6AFyo9j5n7lAINu5b86DYU4daw9GE2swKw_6QRfpsB96HLbE5Uyl-RVHTP4EWop3Fxv0Y66oLTeLKIqp28rA0HWx82L-fMLGPFHuMLrWPC7FM/s320/DSC00460.jpg" width="87" /></a></div>
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The build quality looks good, up to SSL's usual high standards. There is one main board and a smaller jack board, as you can see in the photo to the right. (That blurry white cable with the colored wires coming out is my tacky homemade power cable.) Most of the components are surface mount. The main board is flush to the back of the panel, and the jack board only stands off about one inch (2.5 cm), so there should be no problem installing the 1250 in the most shallow cabinet or <a href="http://electronicmusic.wikia.com/wiki/Skiff" target="_blank">skiff </a>imaginable. The panel is standard MU construction and all of the dimensions are correct.<br />
<br />
The SSL 1250 serves a basic but essential function in a modular synth: to avoid highly repetitive modulations that can become fatiguing to listen to, you need to be able to mix several LFOs to create modulation shapes that are more complex but not totally chaotic. The 1250 does this job admirably. And the blinkylights factor is high too. The one improvement I might suggest is some onboard way to output a combined waveform without having to use a separate mixer. If the output jacks were chained -- that is, the output of a given jack combines with the next higher numbered jack when no <a href="http://electronicmusic.wikia.com/wiki/Patch_cord" target="_blank">cord </a>is plugged in -- that would be useful.<br />
<br />
SSL is at <a href="http://www.steamsynth.com/">www.steamsynth.com</a>. They sell both direct and through dealers. Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-88538161598454546532015-08-27T22:44:00.001-05:002015-09-19T19:26:08.877-05:00Analog computers and synthsA few weeks ago, a poster at <a href="http://www.vintagesynth.com/" target="_blank">VSE </a>asked a good question: To what extent, if any, did the design and use of analog computers in the mid-20th century influence the development of music synthesizers? My first thought was, "probably not much". Then I did some research...<br />
<br />
First, let's go over what an analog computer is. An analog computer, put simply, is a device that accepts input parameters which are represented by something inside the computer. It performs computing functions through mechanisms and/or electronic circuits, and the outputs are expressed by quantities of something the mechanism can produce.<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiZwJhyzZmvvHn1x6jCF6DfnDWhadhAFfcGlSmCixp5N72vvqPMew2nu-KxM4DLzUVpYfe06xWGPE9qd42xeg6UvPYkIx_lwnoloIqYVeZ0v-fU9WyOUgzXJq1IvrmbeWchanuiAGRYp3A/s1600/Philbrick+Analog+Computer.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiZwJhyzZmvvHn1x6jCF6DfnDWhadhAFfcGlSmCixp5N72vvqPMew2nu-KxM4DLzUVpYfe06xWGPE9qd42xeg6UvPYkIx_lwnoloIqYVeZ0v-fU9WyOUgzXJq1IvrmbeWchanuiAGRYp3A/s320/Philbrick+Analog+Computer.jpg" width="292" /></a></div>
<div style="text-align: center;">
A Philbrick K-3 analog computer, circa 1950. From the Philbrick Archives.</div>
<br />
<br />
Analog computers preceded the development of electricity. The first, simple analog computing devices go back to the Middle Ages, but significant ones started appearing during the pre-industrial scientific discovery period from 1600 to 1800. Generally they relied on sliding or rotating parts to represent measurements which were input or output. A simple but important example is the slide rule, invented in the 17th century. A basic slide rule multiplies two numbers by positioning one operand on a sliding scale against a fixed scale; the amount by which the sliding scale is moved represents (by reading off of a scale) the product.<br />
<br />
In the early 20th century, a number of powered analog computers were invented to do specific calculations. An early driver behind the development of this technology was the need for a device called a "gun director". This was a computer that computed the elevation and azimuth angles at which an artillery piece needed to be pointed in order to hit a target, given the range to the target, the wind, the weight of the shell being fired, and possibly other factors. The Norden bombsight was a famous electro-mechanical analog computer deployed by the Allies during World War II. To use it, a bombardier looked through a sight glass to find the target to be bombed. From the pointing angles of the sight, and the rate at which the bombardier had to move the sight in order to keep it on target, the bombsight computed the heading that the aircraft needed to fly, and the time at which the bombs should be dropped. In this computer, the quantities being computed were represented by the movements of levers or gears. (The bombsight was usually coupled to the airplane's autopilot so it could actually fly the aircraft during the bombing run, and to the bomb racks so it could release the bombs at the right time automatically.)<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjPqrQ69CB-SBfqDCehnautrPgAv2brgg_xSOoB00IE80_aHng7BEyP2EWePtABeHBtfBjp-1YXSj-PACUWeyQXeVFtF4Qo5KElJokL8oDNK9GiAJRXDrFZYIuanWOrsSdW4C7nao0sMKI/s1600/Norden+Bombsight.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="225" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjPqrQ69CB-SBfqDCehnautrPgAv2brgg_xSOoB00IE80_aHng7BEyP2EWePtABeHBtfBjp-1YXSj-PACUWeyQXeVFtF4Qo5KElJokL8oDNK9GiAJRXDrFZYIuanWOrsSdW4C7nao0sMKI/s320/Norden+Bombsight.png" width="320" /></a></div>
<div style="text-align: center;">
Norden bombsight (top left) and servos controlled by the bombsight.</div>
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<br />
Electronic analog computers started to appear around 1930. As was generally the case of the mechanical analog computers, most of the early electronic devices were hard-wired to perform a specific computation; because of this, early uses were limited to problems that were both important and difficult, enough that it was worth the cost to build a computer. An early example was a device known as the "AC network analyzer", which was built to solve problems that electrical power utilities were encountering as individual power stations were being combined into large grids.<br />
<br />
In 1938, electrical engineer George A. Philbrick, then employed by the Foxboro Company of Massachusetts, wrote a proposal for an electronic analog computer that would model various types of closed-loop manufacturing processes. One of the problems that Philbrick had to solve was how to design circuits that would perform the needed math operations in a general sense, that is, not specific to a particular problem. In 1943, Philbrick was working on a contract with the U.S. Army to devise improvements to the M9 gun director, which had been built by Bell Labs. It worked, but it was too slow to compute in real time. Philbrick came in contact with Loeb Julie of Columbia University, who had devised the first experimental operational amplifiers. (Yes, there were op amps decades before the first integrated circuits.) Philbrick realized that Julie's op amps could be used to perform a variety of analog computing math functions, and he began working on his own improvements.<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg5hK3P60V7ClH6DhfxZ96mldA9f0p7fl5G6I851J_KOHqCOz_-beTIKBUXVqtrXBqgO2hHKwz3CpZLWeN865imIpv5jn3vHY3LmCi1wy-G5YhoXUxTclrvRijmU_z9Z4_HYhVrdzjYE_U/s1600/philbrick+k2-p.JPG" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg5hK3P60V7ClH6DhfxZ96mldA9f0p7fl5G6I851J_KOHqCOz_-beTIKBUXVqtrXBqgO2hHKwz3CpZLWeN865imIpv5jn3vHY3LmCi1wy-G5YhoXUxTclrvRijmU_z9Z4_HYhVrdzjYE_U/s1600/philbrick+k2-p.JPG" /></a></div>
<div style="text-align: center;">
Philbrick K2-P op amp </div>
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After WWII, Philbrick started his own company, George A. Philbrick Researches. The company was heavily involved in both analog computing and commercial op amp design and manufacturing. The company published a widely regarded collection of papers and notes concerning analog computing -- system design, circuit design, programming, and operations. Analog computers were becoming more compact, and general-purpose units were appearing that offered a number of function modules which could be interconnected by the user in any desired configuration using patch cords. In fact, Philbrick's company developed the idea of a "modular computer", in which individual function blocks could be purchased and combined as needed to apply to a problem -- a concept very similar to the modular computers that would come later. At some point Philbrick hired a certain young electrical engineer, one <a href="http://electronicmusic.wikia.com/wiki/Pearlman%2C_Alan" target="_blank">Alan R. Pearlman</a>, who took an interest in the op amp business. So much so that, in the early 1960s, Pearlman and another Philbrick employee broke away and established their own company, Nexus Research Labs, which continued their work in the op-amp and analog computing business.<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi26Mopq0unR2rrLqvbNSHVinOKs80U6W4mstkYrPOtUh3KSE4bXRRY_7lVEO_djQJqeS9LSzongBxFWg6LXPCANu6AVhlg6O5iJn1owGlbRN0haDMR4UcIgukXhXptlWD0mm7CuDBjkmI/s1600/Philbrick+K3+modules.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="197" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEi26Mopq0unR2rrLqvbNSHVinOKs80U6W4mstkYrPOtUh3KSE4bXRRY_7lVEO_djQJqeS9LSzongBxFWg6LXPCANu6AVhlg6O5iJn1owGlbRN0haDMR4UcIgukXhXptlWD0mm7CuDBjkmI/s320/Philbrick+K3+modules.png" width="320" /></a></div>
<div style="text-align: center;">
Philbrick K3 analog computer modules. From the Philbrick Archive.</div>
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If Pearlman's name doesn't sound familar, look at his initials -- A.R.P. In 1966, Pearlman's group sold Nexus Research Labs to Teledyne. The sale made Pearlman wealthy, and he used some of that wealth to found <a href="http://electronicmusic.wikia.com/wiki/ARP_Instruments" target="_blank">ARP Instruments</a>. Look at the photo above. Looks vaguely familiar? The Philbrick analog computer systems were <i>modular</i>. There were about 10 function modular that the user could purchase and configure in a case as needed. Compare to this:<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiYmYsHfEeCELv-xCVXfCHlOBHBn3pqWJUhUEstr5PHuGfSkchr-kTSIVQr5lPVghhXGtopldKGE763xm0r8MbHm7KGFddsc1tfjTs5lggk87ZF5g_NaYIwcAB4L1T6yq3qbNGw1_4ALqY/s1600/arp_2500_m1047.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiYmYsHfEeCELv-xCVXfCHlOBHBn3pqWJUhUEstr5PHuGfSkchr-kTSIVQr5lPVghhXGtopldKGE763xm0r8MbHm7KGFddsc1tfjTs5lggk87ZF5g_NaYIwcAB4L1T6yq3qbNGw1_4ALqY/s320/arp_2500_m1047.jpg" width="118" /></a></div>
<div style="text-align: center;">
ARP 2500 model 1947 voltage controlled filter</div>
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However... The first of what we consider synthesizers today didn't come from Pearlman. The two men who are generally credited with developing the basic building blocks of the analog synthesizer -- the voltage controlled oscillator, filter, and amplifier -- are <a href="http://electronicmusic.wikia.com/wiki/Moog%2C_Robert" target="_blank">Robert Moog</a> and <a href="http://electronicmusic.wikia.com/wiki/Buchla,_Donald" target="_blank">Don Buchla</a>. Moog has an obvious, if indirect, connect to Philbrick via Columbia University, where Philbrick and Loeb Julie worked on the first op-amp designs in the 1940s, and where John Ragazzini and Rudolf Kalman had continued to work on analog computing concepts through the 1950s. The Columbia-Princeton Electronic Music Center opened at Columbia in the mid-1950s, but it is not clear how much cross-fertilization there was between it and the analog computing labs. Moog just missed experiencing the <a href="http://electronicmusic.wikia.com/wiki/RCA_Synthesizer" target="_blank">RCA Synthesizer</a>, which was installed at the center in 1958; he had graduated in '57.<br />
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Little is written about what Moog actually studied or did at Columbia (far more is written about his theremin side business by which he paid his way through school), so further investigation is difficult. He did get his degree there in electrical engineering, and in a mid-1950s electrical engineering curriculum, he most certainly would have had instruction on computer circuits, both analog and digital. There were probably analog computers to use, and possibly they were Philbrick units like the one pictured above, thanks to the connection to the university via Loeb Julie. Where did Moog come up with the idea to make his first synths modular? Did he spend some time with a Philbrick analog computer at Columbia? Did he, perhaps, try to coax sound synthesis out of it? <br />
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Buchla is even more of a puzzle. There is almost no information available on the Internet about what he did prior to founding <a href="http://electronicmusic.wikia.com/wiki/Buchla_and_Associates" target="_blank">Buchla and Associates</a> in 1962. It is known that he was involved with <a href="http://electronicmusic.wikia.com/wiki/Subotnick%2C_Morton" target="_blank">Morton Subnotick</a> and the San Francisco Tape Music Center, which was a tape studio and had little if anything to do with analog computers. He was involved in some way with the University of California, Berkeley (it's not clear if he was actually a student or faculty there or not), which at the time was the world's foremost center of nuclear physics research, a field in which a considerable number of analog computers were used to model nuclear reactions. Buchla studied physics (along with several other fields) and probably would have come into contact with the nuclear physics program's analog computers. To what extent this influenced his later thinking about synthesizers is difficult to say.<br />
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So to answer our question: did analog computers influence the development of analog synths? The answer, at this point, is "maybe". We know that Pearlman was heavily involved in analog computers, but he came in a little after Moog and Buchla. We know that Moog was at Columbia at a time when the school was involved in both analog computing and electronic music, and we can see similarities between his modular synth designs and some of the modular computer designs that he might have worked with. Buchla is less certain, but he probably would have at least seen analog computers at Berkeley. <br />
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For more information about George Philbrick and his pioneering company (it's a worthwhile read for anyone interested in electrical engineering history), see the <a href="http://www.philbrickarchive.org/" target="_blank">Philbrick Archive</a> at www.philbrickarchive.org. Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-85728056180893235172014-10-19T21:19:00.000-05:002014-10-19T21:19:11.276-05:00Fetishizing the Moog modulars
<div style="margin-bottom: 0in;">
Lately I'm noticing a big surge of
interest in the vintage modular Moogs. Now, this in itself is not a
bad thing. It's a good thing, and not only from the historical preservation sense. It's always good to have a perspective of history, and to see
how Bob Moog and his compatriots made their decisions and went about
doing things without access to all of the technology we have today.
Remember, in 1963 when Moog and Buchla built their first modules, the
integrated circuit was still largely confined to Fairchild Semiconductor's labs.
The commercially package operational amplifier was a big ugly box
that plugged into a tube socket and contained a pair of 12AX7 tubes
inside it. There were no OTAs, no 4000 CMOS logic; Doug Curtis was
in elementary school, and Ron Dow had not yet gone to Dave Rossum
and Scott Wedge to beg for money (which was a good thing, since
Rossum and Wedge were themselves high school students and didn't have any
money).
</div>
<div style="margin-bottom: 0in;">
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</div>
<div style="margin-bottom: 0in;">
Yes, things were different back then.
Moog (and, independently, Buchla) had just thought of the idea of
“voltage control”, in which he imagined that a generated signal
might be able to remotely control the functioning of another circuit,
thereby increasing the possibilities for more animation in electronic
music, e.g., that the output of one oscillator could control the
frequency of another in order to introduce vibrato, without a person
having to constantly turn the frequency knob up and down. This was
new territory; at first Moog had no idea how to do it with components
that were available to him. As he attacked the problem, he made it
work, but there were a lot of compromises: many components being made
to do things that they weren't designed to; use of some expensive
components which forced cost cutting in some other areas, and the
necessity to keep the circuits confined to a reasonable sized
package. There were also things to consider like what we now call
the “user interface” was to function. (We all know the story of
how the synthesizer came to be primarily a keyboard instrument: the
switches of an organ keyboard, wired to a resistor grid, worked a lot
better than primitive pitch-to-voltage converters and provided an
interface that looked familiar to musicians.)</div>
<div style="margin-bottom: 0in;">
<br />
</div>
<div style="margin-bottom: 0in;">
Consider Moog's first voltage
controlled oscillator, the model 901A/B duo. At a list price of several
hundred dollars in 1964, what you got for a 901B VCO was a basic oscillator with
four waveform outputs. If you wanted volts/octave response (which
was essential for any kind of tonal music), you had to also buy the
separate 901A driver module which contained the exponential
converter. And oh by the way, the VCO contained absolutely no
temperature compensation, which meant you had to constantly re-tune
as the circuits warmed up and/or the room temperature varied.</div>
<div style="margin-bottom: 0in;">
<br /></div>
<div style="margin-bottom: 0in;">
As another example, consider the Moog 904B VCF. Here's a photo of one:</div>
<div style="margin-bottom: 0in;">
<br /></div>
<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgg143iTixZyzUUfesJtkt2d3tKSdwNXeJ1GRP0MpsufSo71BVZ3OPwytM7Tp79nmz-k9prrauZEc5TfLZiGQiVB0H4JSmN8IkU8wApZNpq2ls6-xn5-MVTtL-xJP3hTYjuydPbTng5gbI/s1600/moog+904B_hpf.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgg143iTixZyzUUfesJtkt2d3tKSdwNXeJ1GRP0MpsufSo71BVZ3OPwytM7Tp79nmz-k9prrauZEc5TfLZiGQiVB0H4JSmN8IkU8wApZNpq2ls6-xn5-MVTtL-xJP3hTYjuydPbTng5gbI/s1600/moog+904B_hpf.jpg" height="320" width="155" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Moog 904B. Photo courtesy of David Brown at <a href="http://modularsynthesis.com/">modularsynthesis.com</a></td></tr>
</tbody></table>
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</div>
<div style="margin-bottom: 0in;">
Note that it's a 2U wide module and how big it is and how much empty
space there is on the panel. Why is it so big? Because the circuit
board behind the panel needed to be that big in order to cram all of
the circuitry in. Here's another, more drastic example of that sort of thing:</div>
<div style="margin-bottom: 0in;">
<br /></div>
<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiWKx16EQV715a07AdjmM_ZpjPx30pfsN0oKX1rK1QvjpG3OvBtSgxOy-kEGyxOzjiaqjCKfJNw8baVImpyYTp7sLa0SbMhb-SyjOL6ZaaN1T_egM8_oXkb6ZnWCPa22hBHZrOivcnhj7c/s1600/moog+905.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiWKx16EQV715a07AdjmM_ZpjPx30pfsN0oKX1rK1QvjpG3OvBtSgxOy-kEGyxOzjiaqjCKfJNw8baVImpyYTp7sLa0SbMhb-SyjOL6ZaaN1T_egM8_oXkb6ZnWCPa22hBHZrOivcnhj7c/s1600/moog+905.jpg" height="320" width="155" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Moog 905. Photo courtesy of David Brown at <a href="http://modularsynthesis.com/">modularsynthesis.com</a></td></tr>
</tbody></table>
<div style="margin-bottom: 0in;">
This is the 905 reverb. Lots of wasted panel real estate? You bet. It's that large because it uses a spring reverb tank, which is mounted in the module itself, right behind the panel. Modern modulars that offer spring reverb modules mount the tank remotely, somewhere in the rear of the case. Although, oddly, the <a href="http://cluboftheknobs.com/modules.html" target="_blank">Club of the Knobs</a> reproduction of the 905 retains the same 2U wide panel design, even though it uses remote mounted tanks:</div>
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<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjMrY0bXZ2C_KRaC0zR-p_pLs8lNdHHazzqTrsShdw3WVNkrwx_VYvXAkr3IMzP5CdQdApcPYxCLtZtORwud6uXkhmNyQnlJkAIKf9X63yFiCcIoWwrKQXN1EuADo6axifBSmRGrs7iPM4/s1600/cotk+905.gif" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjMrY0bXZ2C_KRaC0zR-p_pLs8lNdHHazzqTrsShdw3WVNkrwx_VYvXAkr3IMzP5CdQdApcPYxCLtZtORwud6uXkhmNyQnlJkAIKf9X63yFiCcIoWwrKQXN1EuADo6axifBSmRGrs7iPM4/s1600/cotk+905.gif" height="220" width="320" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Club of the Knobs C905 reverb. Photo from COTK's Web site.</td></tr>
</tbody></table>
<div style="margin-bottom: 0in;">
This is taking authenticity to a bit of an extreme. Clearly, a 1U panel would have been sufficient. The Eurorack users always say that all of the large format modulars take up too much space, and this sort of thing doesn't help.</div>
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<br /></div>
<div style="margin-bottom: 0in;">
There were a lot of things about the
Moog modulars that were different from today's modulars and made them
not so easy to interface to or work with. Most Moog VCOs and other
signal generators output a signal that is only 1.5V peak to peak.
This I assume was a choice made based on typical use of signals as
modulation sources, but, for example, it means that the output of a
VCO or an envelope generator can't be made to drive a VCF though its
full frequency range without being amplified. For reasons totally unclear to me, <a href="http://www.mos-lab.com/" target="_blank">MOS-LAB</a> recently decided to go back to Moog's 1.5V standard for its reproduction of the 901B and 921B VCOs. </div>
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<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiMd2yybpNFVn7aawLW3HR-_KykDcVUl-RZVphaQP2M0no3a8F9NPy4hyphenhyphenVRoJ4l1rWduH9ruO2IfZrwBq1bp6mteSJbf3uQaApFXkySc3Hq7wchxgUZX0j96pBERiMOovLxrVC0jPauuYM/s1600/moslab+921b.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiMd2yybpNFVn7aawLW3HR-_KykDcVUl-RZVphaQP2M0no3a8F9NPy4hyphenhyphenVRoJ4l1rWduH9ruO2IfZrwBq1bp6mteSJbf3uQaApFXkySc3Hq7wchxgUZX0j96pBERiMOovLxrVC0jPauuYM/s1600/moslab+921b.jpg" height="320" width="72" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">MOS-LAB 921B VCO. Photo courtesy of <a href="http://mos-lab.com/">MOS-LAB.com</a></td></tr>
</tbody></table>
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</div>
<div style="margin-bottom: 0in;">
And there's the infamous S-trigger
signals. On a Moog modular (and other vintage Moog synths such as
the early Minimoogs), something that generates a trigger or gate
signal does not output a voltage pulse. Rather, the output is a
simple transistor that is saturated in the “low” or “off”
state, shorting the output to ground, or cut off in the “on” or
“high” state, which leaves the output “floating”
electrically. The output expects that whatever trigger/gate input it
is patched to will “pull up” the output by applying a voltage
through a resistor. When the output is in the high state, its
voltage rises to the pull-up voltage; when it is in the low state, it
shorts the output to ground, and the pull-up resistor limits the
current that flows to ground. We've all seen that the modular Moogs
use the infamous “Cinch-Jones” two-bladed connector for trigger
outputs and inputs, requiring a separate type of patch cord to
connect them (and thus the modular Moogs do not have fully unified
patching). This is why; if a trigger/gate input, with its pull-up,
were inadvertently connected to a signal output, it could potentially
damage the output circuit. But it's a pain because of the special cable needed, and because you need an adapter to interface any external trigger or gate source. Mercifully, neither MOS-LAB nor COTK has chosen to use the S-trigger on their Moog reproductions, even though the connector itself <a href="http://www.mouser.com/catalog/catalogusd/647/2825.pdf" target="_blank">is still available</a>. </div>
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<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhnP8NSivqrUa5Qff4CW8T1SQcP-nfZDf78-ULUlRe5EQhc4Ca6v2ZhISHTRxxpZ5ld02kHj5vf2sE_pGuDVwpQjkXramVM_rEkqDj7sw73NHiCAE5h90eTsuOrX8nTN38X0S5tWJCUoKc/s1600/moog+911.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhnP8NSivqrUa5Qff4CW8T1SQcP-nfZDf78-ULUlRe5EQhc4Ca6v2ZhISHTRxxpZ5ld02kHj5vf2sE_pGuDVwpQjkXramVM_rEkqDj7sw73NHiCAE5h90eTsuOrX8nTN38X0S5tWJCUoKc/s1600/moog+911.jpg" height="320" width="79" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Moog 911 envelope generator; note Cinch-Jones gate input connector at bottom left. Photo courtesy of David Brown at modularsynthesis.com</td></tr>
</tbody></table>
<div style="margin-bottom: 0in;">
And last but not least, there's the cost of construction using those "authentic Moog" methods and circuits. As I wrote above, there were a lot of places where the Moog designs had to use methods and techniques that were a lot more expensive (such as building op-amps out of discrete circuitry) because more capable components weren't available at the time. Consider: <a href="http://www.synthesizers.com/" target="_blank">Synthesizers.com </a>offers two step sequencers -- the Q119 and the Q960. The Q960 is a fairly faithful recreation of the Moog 960 sequencer design, up to and including the incandescent lamps which indicate the active stage (which most users replace with LEDs because the lamps burn out frequently). The Q119, on the other hand, has most of the same capabilities and controls but is microprocessor controlled, and all of the indicator lamp are LEDs. The two share many capabilities -- but the Q119 is about $300 less expensive, plus in order to duplicate the Q119's 24-step mode with the Q960, you need to add a Q962 sequential switch, at an additional $160. </div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiWzf69-UxHSj-EWOWVjHD3azgfgWAzduG98mU4NoJ7igMbxy-7umac00uOckXELrvhw8wALylgEQBB8yS6V6ZrEy_kxmN4Y59WD2Fc1X0V2zuZ4Xp2030vD1_51JqzxwfCHDiXokuodwQ/s1600/moog+960.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiWzf69-UxHSj-EWOWVjHD3azgfgWAzduG98mU4NoJ7igMbxy-7umac00uOckXELrvhw8wALylgEQBB8yS6V6ZrEy_kxmN4Y59WD2Fc1X0V2zuZ4Xp2030vD1_51JqzxwfCHDiXokuodwQ/s1600/moog+960.jpg" height="162" width="320" /></a></div>
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<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhIifTG4VaPm0VWWl27VRvDU3uPEtsObsWyqF2U-Sz5sU-0VGAWTnidiDAg1c1XNy3dsc5u3uSM0iQ2SiSHsBE77ur3Vclc3K8LKlB3KQ91HVbVuLTfvloCqmCM2xEM-rNVmXj-xVXll00/s1600/q119_l.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhIifTG4VaPm0VWWl27VRvDU3uPEtsObsWyqF2U-Sz5sU-0VGAWTnidiDAg1c1XNy3dsc5u3uSM0iQ2SiSHsBE77ur3Vclc3K8LKlB3KQ91HVbVuLTfvloCqmCM2xEM-rNVmXj-xVXll00/s1600/q119_l.jpg" height="174" width="320" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Moog 960 (top) and Synthesizers.com Q119 (bottom). Top photo courtesy of David Brown at modularsynthesis.com; bottom photo courtesy of Synthesizers.com</td></tr>
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<div style="margin-bottom: 0in;">
So given all of the above, I'm starting
to wonder if the current market isn't fetishizing the modular Moogs a
bit much. Of course, the Dotcom/MU format was based on the physical
dimensions of the original Moog modulars, and Roger Arrick's designs
continue to take certain design cues from the Moogs, such as the
black panel background and the knob style. But Arrick started out
with fresh circuit designs using contemporary electronics technology.
And he's no slave to the Moog look and feel; he has never hesitated
to make a module smaller than the functionally equivalent Moog module
when the circuit design allowed for it. The other notable thing was
that Arrick avoided both the weird mix of power supply voltages and
the edge connectors that Moog modules used; Synthesizers.com set the
standard power for the MU format at +/- 15V and +5 volts, and the
flexible power supply harness doesn't limit the modules' board
mounting geometry the way the Moog edge connectors did. (The Dotcom
MTA-100 power connectors are also a lot less prone to corrosion
problems than the Moog edge connectors are.) As for Club of the Knobs, they started out copying the Moog modules, but soon realized that simply duplicating the Moog lineup would be too limiting. And although they continue to stick to the general Moog format, they have long since blown past the limitations of the original Moogs with module designs that Moog could never have thought of or implemented with the technology available at the time, such as the C950A MIDI interface / arpeggiator. </div>
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<div style="margin-bottom: 0in;">
That's why, to be honest, I really
don't want to see a big comeback of slavish Moog-modular clones. Even
putting aside the difficulties of obtaining exact replacements for the
vintage Moog parts, the 1960s Moog modulars were just all-around
limited compared to what is available today. Yes, it's great that
Moog has been able to sell several of the $1.5M copies of the Keith
Emerson modular; the units will instantly be valuable collector's
items as well as being highly educational, and more power to Moog for
being able to build them and sell them at that price. What bothers
me is the people might get the idea that the 5U formats are all about duplicating what has been done in the past, with the implication being that you have to turn to the Euro format to find any modern or fresh ideas. That would be a self-limiting move for 5U. And as someone who wants modern capabilities but prefers to work with 5U formats, I don't want to see that happen. </div>
Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-43516051705530495622014-10-09T21:53:00.000-05:002014-10-09T21:53:19.039-05:00Review: Izotope IrisIris is a plug-in from <a href="https://www.izotope.com/" target="_blank">Izotope</a>, a company that's better known for products like Trash, Stutter Edit, and "make my track the loudest thing on the dial" mastering tools. Despite what that might suggest, Izotope has a lot of background in <a href="http://electronicmusic.wikia.com/wiki/Fourier_transform" target="_blank">Fourier analysis</a> and spectral processing tools, and they put most of it into Iris. The difference is, with Iris, you get to control it yourself. <br />
<br />
Iris employs a method of synthesis whose availability to the masses is relatively recent, and that I had no previous experience with -- spectral editing. So when I first installed it and fired it up, I wasn't at all sure what to expect. But I have been pleased, if at times a bit baffled, at the results I've gotten so far. With that said, let's dig in.<br />
<br />
<h3>
Spectral Editing</h3>
<div>
What exactly is spectral editing? Well, it involves taking a sample of a sound, and picking out bits of its spectrum over time that you want to reproduce, excluding the rest. Iris presents a sample that you choose in a two-dimensional window, with time on the horizontal axis and frequency on the vertical axis. Energy at a given time/frequency coordinate is represented by a pixel on the screen, with the brightness of the pixel indicating how much energy is present. You use various methods of selecting regions of the time-frequency space that you want to reproduce. Then, when you play the note, it reproduces the regions you select.</div>
<div>
<br />
<h3>
The User Interface</h3>
Here is a shot of the screen you are presented with when you first start the plug-in:<br />
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<table align="center" cellpadding="0" cellspacing="0" class="tr-caption-container" style="margin-left: auto; margin-right: auto; text-align: center;"><tbody>
<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhs8yfB30rscFXc77KVEmrA72fnWXN5uYwvrPsBfxaJIH5QAVzutAOEcmu88SSS_DA3tCGU6Y8rsrGgV9uR4FgnLAxobA_mUNeVZh-2T-Uhknj84xvNwJd5djE3Dtz_ZbO1r3YsNVzYSzY/s1600/iris0.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhs8yfB30rscFXc77KVEmrA72fnWXN5uYwvrPsBfxaJIH5QAVzutAOEcmu88SSS_DA3tCGU6Y8rsrGgV9uR4FgnLAxobA_mUNeVZh-2T-Uhknj84xvNwJd5djE3Dtz_ZbO1r3YsNVzYSzY/s1600/iris0.jpg" height="240" width="400" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">Iris initial view</td></tr>
</tbody></table>
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<div>
</div>
The big window with the Izotope logo is where you will load a sample to edit. The toolbar over to the left consists of drawing tools that you use to select regions to be reproduced. The controls on the right allow you to select a layer, tune an edited sample, set up an amplitude envelope for the layer, and route modulations. For each patch, Iris allows up to three layers, plus a "sublayer" which uses fixed waveforms but still allows for spectral editing.<br />
<br />
The buttons at the top right switch between different views, or presentations of the user interface. Any of the three layers or the sublayer may be selected for editing by clicking on the "1", "2", "3", or "Sub" buttons. There are two other views available
by clicking their buttons at the upper right of the window: the All view
and the Mix view. We'll discuss these views later. For now, we'll concentrate on the single-layer view, which is where you will probably do most of your work.<br />
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<h4>
Sample Loading and Manipulation</h4>
When you load a sample, either by dragging and dropping one into the window, or by selecting one from Iris' library via the browser, you are presented with a window that looks like this (click on the image to see a full size version):<br />
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<a href="https://www.blogger.com/blogger.g?blogID=279251740880432906" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"></a><a href="https://www.blogger.com/blogger.g?blogID=279251740880432906" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"></a><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEht5yzcqBgsv9vmXutyLNgJqhnUrNL32jxiRQbg6CN-ZZzxqWl_oQE0aJ4ZsaiXcusCPIq79UfCIuAPlowYW9nct_Q3loGfw3aLz7wbbqss5Vvrshhk2BAV3BAlAndA8UyGlQwX3IsYf24/s1600/iris1.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEht5yzcqBgsv9vmXutyLNgJqhnUrNL32jxiRQbg6CN-ZZzxqWl_oQE0aJ4ZsaiXcusCPIq79UfCIuAPlowYW9nct_Q3loGfw3aLz7wbbqss5Vvrshhk2BAV3BAlAndA8UyGlQwX3IsYf24/s1600/iris1.jpg" height="241" width="400" /></a></div>
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(If you have a copy of Iris, the above sample is "SK8 Circuit Singer".) What you're looking at here is a spectral energy (or power, if you want to think of it that way) display. <br />
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Note the frequency scale along the left edge; it indicates that this sample has its highest-energy bands in the 1.5/2.0 KHz range. If you look at the right side of the on-screen keyboard, you'll see a little tag at F#6; this indicates what the sample's "natural" note is, i.e., at what note the sample will play back with no pitch shifting. You can click and drag this tag to change the setting.<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg1hlJjlOiBuAUNc_t4_KMhh8xxHdiWM-_yM_RMSpt9Pa4y2lU5di0c4frTTp5d_v6fmsWPp0KB9khSv8ZMV0OgOY5FWuO5pIae6OLWndfsyr239zNRWmFXY6ynWKRNCoE-wl1BlafI8C0/s1600/irisf_ruler.png" imageanchor="1" style="clear: left; float: left; margin-bottom: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg1hlJjlOiBuAUNc_t4_KMhh8xxHdiWM-_yM_RMSpt9Pa4y2lU5di0c4frTTp5d_v6fmsWPp0KB9khSv8ZMV0OgOY5FWuO5pIae6OLWndfsyr239zNRWmFXY6ynWKRNCoE-wl1BlafI8C0/s1600/irisf_ruler.png" height="400" width="22" /></a></div>
Immediately above the sample window you can see a gray bar (shown above this text) with an arrow at each end. You use this to set start, stop and loop points (although Iris will do it automatically, as will be explained later). This is all fairly conventional and similar to other types of sample editors; it allows for one-shot, forward looping, and alternate forward-and-back looping. The bar above this graphically portrays the sample's amplitude envelope, and the gray region indicates the bounds of the currently selected loop, which may be different from where the arrows are set if Iris has chosen the loop points automatically. Below and to the left of the screen are rulers; the one on the left (see to the left of this text) displays frequency, and the lower one (below the next paragraph) displays sample word counts. You can scroll the view horizontally and vertically by grabbing one of these rulers and dragging it.<br />
<br />
(If the rulers don't appear, click on the wrench icon at the upper left
to bring up the preferences dialog, and click on "Show Time Ruler"
and/or "Show Freqeuncy Ruler", as you prefer. You can also choose time
or number-of-samples scales for the time ruler, and there are several
choices of units for the frequency ruler.) <br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgG2vl7OYn9i8Ljm6RBCmuIu5O_Tz0NPUIVrZ_nG-D1k1hz7z9CfFL-laOJVVxTUimKdt8hqyRU9KvKTq8LZ2IJiATf_XyMm2w378sDw3xMMWyDtV4ftztJrtQG8cUmYB-jhXm8xHK0wNc/s1600/iriszoom.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgG2vl7OYn9i8Ljm6RBCmuIu5O_Tz0NPUIVrZ_nG-D1k1hz7z9CfFL-laOJVVxTUimKdt8hqyRU9KvKTq8LZ2IJiATf_XyMm2w378sDw3xMMWyDtV4ftztJrtQG8cUmYB-jhXm8xHK0wNc/s1600/iriszoom.png" /></a>Other tools for view manipulation are at the top left corner. When you click on the top one (the one that looks like a screen with a magnifying glass over it), it zooms in or out enough to show you all of the drawing that you have done in the layer. The second one (the magnifying glass with an X) does a full zoom out and shows you the entire sample. The third one is an arbitrary zoom tool; when you click on it, you can draw the diagonal of a rectangle that you want to zoom in on. The hand tool at the bottom is a scroll tool; click on it, and then you can grab and drag the content of the window.<br />
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<h3>
Editing</h3>
(A note here: Izotope refers to edited/painted regions of spectral content as "selections". To avoid confusion with the conventional use of the word "selection" in computer UI phraseology, I'm avoiding the term here; I'll refer to such areas as drawn/painted regions.)<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjIzIdWHVF4jbLj-lEAfY0S7cWI52CejKFwqBLcnlLshS5ccYRSOn7vRV_Gq-JCfDKI28QZpBHNT-puQiiCudrBb9FOlwUyYzMWBj-N5cjo7UFv3hmaWwaIDhzQ8c0aKY5iS3jDnySNw8Q/s1600/iris2.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjIzIdWHVF4jbLj-lEAfY0S7cWI52CejKFwqBLcnlLshS5ccYRSOn7vRV_Gq-JCfDKI28QZpBHNT-puQiiCudrBb9FOlwUyYzMWBj-N5cjo7UFv3hmaWwaIDhzQ8c0aKY5iS3jDnySNw8Q/s1600/iris2.png" /></a>You actually choose spectral content to be played by drawing in the window using the tools from the center left margin (shown to the right of this text). A better analogy for most of the tools is actually painting; you move the tool and as you do so, it paints a certain area which is to be played. The three tools at the top select rectangular areas. The top tool paints a region of time; you choose the tool and then drag horizontally across the time band you want, and it paints all frequencies within the time band. The third tool does the same for frequency bands; you drag up or down and it paints all content across time within that frequency band. The middle tool selects as rectangular area.<br />
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(When you first load a sample, before you paint anything, if you play a note you will hear the unedited sample. This is turned off when you do your first painting operation on the sample. If you paint an area and then erase it, so that nothing is painted, you will hear nothing.) <br />
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The fourth tool, the brush tool, allows areas to be painted freehand. Click and hold and move the mouse around to paint. Painting is additive, so if you paint over any areas that are already painted, nothing happens. You will notice when you select the brush tool that a slider control appears above the on-screen keyboard, just to the left of the word "KEYBOARD". This allows you to set the brush diameter. <br />
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The bottom tool in the tool selection is the eraser. It works similarly to the paintbrush, except that instead of painting, it un-paints already painted areas. The size slider also works with the eraser. <br />
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The lasso tool allows you to cause an arbitrary-shaped area to be painted by drawing a boundary around it. Click and hold and freehand the boundary. If you don't complete the loop, it will be completed for you with a straight line connecting the start and end of your draw when you let the mouse button up. <br />
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The remaining tool is the "magic wand" tool. This is a bit hard to explain and I don't quite have a handle on exactly what it does. When you use it to click on a painted region, it will select content that is harmonically related and coincides in time with the region you selected. So far, for me it's a "try it and see what it does" thing.<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjJaPIJ8Ooh9qsU9wadOroE-zI4ckZQqdbVQqxYzIXBGiBA_lDwF1mV4k8RCylGYW9-z2D3OF7BEr9S8IcbAxrf0FK3bDoMgs4RNIE1z4cIHb6NwOdU1OSmSS4ceYXHOv74qr0AH3PNqug/s1600/irisinv.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjJaPIJ8Ooh9qsU9wadOroE-zI4ckZQqdbVQqxYzIXBGiBA_lDwF1mV4k8RCylGYW9-z2D3OF7BEr9S8IcbAxrf0FK3bDoMgs4RNIE1z4cIHb6NwOdU1OSmSS4ceYXHOv74qr0AH3PNqug/s1600/irisinv.png" /></a></div>
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The next set of tools allow you to make bulk changes to the areas you have painted. Iris maintains an idea of the bounds, with respect to frequency and time, of the areas you have painted. When you click on the first one (with the horizontal areas) it inverts all of the painted and un-painted areas within the frequency bounds of where you have drawn. Similarly, the tool with the vertical arrows inverts within the time bounds. The one with the diagonal arrows inverts across the entire sample. The last two tools, with the four arrows and the box with an X, paint or un-paint the entire sample, respectively. <br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjXPYGx_G4oYYli1LHnMaBmNsqbySykvoBbacFqmZ8DEA6cPawLZHkQlyz3PJPrnZ-HS4eJWH4IbA6JihwyKVb5xeP1bsaWw3SLLTZs-BLtEA800_4iOVJlGxNcxL7o433eTkUGNaM_MF4/s1600/irisundo.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjXPYGx_G4oYYli1LHnMaBmNsqbySykvoBbacFqmZ8DEA6cPawLZHkQlyz3PJPrnZ-HS4eJWH4IbA6JihwyKVb5xeP1bsaWw3SLLTZs-BLtEA800_4iOVJlGxNcxL7o433eTkUGNaM_MF4/s1600/irisundo.png" /></a></div>
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The two tools below that are the undo and redo buttons for painting and drawing. The bottom tool is a preview button; it plays the spectral-edited sample without any pitch shifting. <br />
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An example of an edited sample:<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjQb0w3jwBzR0g6XiW1WKQwdbHAfdTFC7GQQOthQ73EkugMj8T8-ygazxD6O2TP3VGBpyjYgtIzbJnvmAoctn0bXG4Ro2apafRkO9FZpPyhkjzqbkeU1Aaou5YII9adqo2WouBdw4vBa2Q/s1600/iris4.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjQb0w3jwBzR0g6XiW1WKQwdbHAfdTFC7GQQOthQ73EkugMj8T8-ygazxD6O2TP3VGBpyjYgtIzbJnvmAoctn0bXG4Ro2apafRkO9FZpPyhkjzqbkeU1Aaou5YII9adqo2WouBdw4vBa2Q/s1600/iris4.jpg" height="241" width="400" /></a></div>
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Note the four painted regions, including the narrow one at the bottom, and the two vertical dashed lines that indicate the loop start and end points. These illustrate the results with various drawing tools: The area at the left was freehanded; the area at the lower right was drawn with the lasso tool, and the area at the upper right was drawn as a rectangle and then attacked by the eraser. <br />
<h4>
Brushes and Erasers Don't Scale with Zoom</h4>
Note that when you zoom in or out in the view, the size selection for the brush and eraser tools doesn't scale; it remains the same on-screen width at all zoom levels. This means that if you paint a line, zoom in, and then paint another line, the second one will be narrower than the first one. <br />
<h4>
Auto Loop Points Adjustment</h4>
<div>
Whenever you do any drawing or add or alter painted regions, the sample loop points automatically move so as to move the leftmost and rightmost edges of all the painted regions. This is done to eliminate periods of silence that would result if the entire sample were played. Refer back to the above figure and notice how Iris has automatically set the loop points in this example, as indicated by the vertical dashed lines, and gray area in the waveform display above the editing window. You can override this and set the loop points manually if you wish, and you can also set the sample start point independently of the left loop point.</div>
<h4>
Dragging Painted Regions</h4>
<div>
When any of the above drawing tools is selected, except for the eraser or magic wand, moving the cursor over an existing painted region causes the cursor to change to a hand icon. When this occurs, you can grab and drag the region that the cursor is positioned over. Note that Iris keeps track of contiguous painted regions, as indicated by the crawling dashed lines surrounding them, and it updates these each time you make a change; touching or overlapping painted regions will be combined for dragging purposes. Also note that dragging moves the region, not the sample content underneath.</div>
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<h4>
The Undo Trap</h4>
<div>
You might notice that there are two sets of undo buttons on the user interface: one down near the bottom of the left margin, above the preview button, and another set at the top margin to the right of the patch selection controls. The ones on the left, you can use to undo and redo drawing operations. Beware of those ones at the top. They undo operations like sample selection and parameter changes. If you select a sample, do a bunch of spectral editing on it, and then hit the undo button at the top of the window… it will de-select the sample and you'll lose all the drawing you've done.</div>
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<h3>
Layer Controls </h3>
<div>
<div class="separator" style="clear: both; text-align: center;">
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEje6UMfUJq_THMuv8Z02F_PJ4a88SiYAwUZhC8Uc8cOUdRfdSgj-zmON1T6yvyXRvN7eTwjTuC81ZwsWUrMIoYTZTfaI8ZDK_eTAlO6nlkNlS1DckwakrLprsR-CYfslxCFdtNyO7bVXRk/s1600/irisamp.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEje6UMfUJq_THMuv8Z02F_PJ4a88SiYAwUZhC8Uc8cOUdRfdSgj-zmON1T6yvyXRvN7eTwjTuC81ZwsWUrMIoYTZTfaI8ZDK_eTAlO6nlkNlS1DckwakrLprsR-CYfslxCFdtNyO7bVXRk/s1600/irisamp.png" height="194" width="200" /></a> There's a small set of layer controls for each layer, corresponding to other layer-specific features. The four knobs above the amp envelope allow you to set tuning for the layer, control its overall volume, and pan it in the stereo image (the samples are all mono). Each layer has its own amplitude shaper; the amplitude envelope is an entirely conventional ADSR envelope, which you set up graphically by dragging on the little squares. Each layer has an LFO which can be routed to pitch or amplitude (you make it active by clicking on the little power-button symbol next to the word "LFO"). You can set the effects sends for this layer or for the whole mix, depending on the effects mode; I'll say more about this further down. </div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEilLtogfi5M91hIG9FB6-gAZPshZ0F-HE4vrjBC_5BRLvyNWXlBxEunU0i_TI8PLw08fo2tCpN-oDxBJ1z8VyvTkNFLz-yoT9qsq6zOPacuPsvqsYY03fvoFXzXyt4LQ4kGfYIZtU1SPLk/s1600/iriscontrol.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEilLtogfi5M91hIG9FB6-gAZPshZ0F-HE4vrjBC_5BRLvyNWXlBxEunU0i_TI8PLw08fo2tCpN-oDxBJ1z8VyvTkNFLz-yoT9qsq6zOPacuPsvqsYY03fvoFXzXyt4LQ4kGfYIZtU1SPLk/s1600/iriscontrol.png" /></a></div>
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Several significant controls are in the text box under the word "CONTROL". Clicking next to the word "Loop" allows you to choose the loop mode for this layer. Forward, reverse, alternating, and one-shot in either direction are available. </div>
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The "Pitch" item is a pop-up menu with three choices. Iris' normal mode for sample playback is to pitch the sample by speeding up or slowing down playback, the way that early hardware samplers did (this is misleadingly labeled "Resample" in the menu). This of course means that the sounding of a higher-pitched note will be compressed in time compared to a lower note. Choosing "Radius RT" engages a pitch-shifting algorithm that makes all notes play back in the same amount of time. On my i7-based Mac Mini, I found the Radius RT default settings, and the user manual, to be excessively cautious about CPU usage; the manual warns that the pitch shifting algorithm is CPU intensive and accordingly the factory default is to only allow 4 voices at a time to use the algorithm. However, with four notes playing on a layer set to Radius RT, I didn't observe CPU usage to increase much, and I was able to boost the max voices up to 10, and the octave range to +/- 3 octaves, without any problems. I would not hesitate to use Radius RT for any patch where I wanted the playback or loop time to be constant. (On the other hand, allowing loop times to vary with pitch can add thickness to sustained chords and help cover up naff loop points.)</div>
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Retrigger mode prevents new notes from starting at the beginning of the sample when playing <a href="http://electronicmusic.wikia.com/wiki/Legato" target="_blank">legato</a>. In mono mode, you can use it to play legato and the new or retriggered note will pick up the playback at the point where the first one was. In poly mode, it's a bit strange: If you set up the amp envelope with a long release time and turn retrigger mode off, and then play and hold one note while playing a second note, the second note will began playing at whatever point the first note is at in its playback when you struck the second note. Of course, if you are in resample pitch mode, they won't stay together because one will play back faster than the other. </div>
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To the right of the word "CONTROL" are mute and solo buttons. These work similarly to on a mixer. If you hit "Mute", the current layer is muted. If you hit "Solo", every layer other than the current layer is muted. A muted layer is indicated by its layer button turning red. </div>
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The modulation routing controls at the bottom right are not specific to the layer; they are global. We'll talk about modulation later.</div>
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<h3>
Views</h3>
</div>
<h3>
</h3>
<div>
There are three basic views available in Iris. The layer view we've already reviewed above; it allows you to see a single layer at a time and perform spectral editing. There are two other views available by clicking their buttons at the upper right of the window: the All view and the Mix view.</div>
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<tr><td style="text-align: center;"><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEikUuaLiTtNjOjRWaFYm8FWzi8lGDqJTyUqhR0ZGqlBqT3g83Gm_zj8NfA5euIL3-hKGXc1-FRULLu3YSq6-Q2BClmV9pS4Wfg3zbFYs1ce59wcTA-YzpeDUzz-LYSXgfLf6gjQERI4-DM/s1600/irisallwin.jpg" imageanchor="1" style="margin-left: auto; margin-right: auto;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEikUuaLiTtNjOjRWaFYm8FWzi8lGDqJTyUqhR0ZGqlBqT3g83Gm_zj8NfA5euIL3-hKGXc1-FRULLu3YSq6-Q2BClmV9pS4Wfg3zbFYs1ce59wcTA-YzpeDUzz-LYSXgfLf6gjQERI4-DM/s1600/irisallwin.jpg" height="240" width="400" /></a></td></tr>
<tr><td class="tr-caption" style="text-align: center;">The "All" view </td></tr>
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The All view is similar to the individual layer view, but it displays small versions of all four layer samples in four panes in the sample editing area. You can scroll these up and down using the small buttons to the right of the panes, or grab and drag the frequency rulers. You can actually do spectral editing in these panes, although it's awkward due to the small size. (Recall what we said a while ago about the painting tools not scaling with the zoom level.) This view contains the same patch parameter settings on the right side as the individual layer views. You choose which layer they effect by clicking in its pane in the sample viewing area. <br />
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<tr><td class="tr-caption" style="text-align: center;">The Mix view</td></tr>
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The Mix view shows all of the patch parameters for all four layers, plus the global patch parameters and effects settings. The sample view area shows only an overview of the sample with none of the spectral editing, and you cannot edit here. However, you can change all of the patch parameters. The area on the right contains the filter controls and the global LFO settings, plus the effects mode setting and the MIDI Learn button, which are only accessible from this view.<br />
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<div>
The filter effects the overall mix, including the outputs of the effects. There are 11 algorithms; three each of low pass, bandpass, and high pass, plus a peak booster (a sort of very narrow bandpass). Most of them sound like they are intended to emulate familiar analog filter circuits. Cutoff and resonance are adjustable on all, plus there is a dedicated ADSR envelope that can modulate cutoff.<br />
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<h3>
Modulations</h3>
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<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg6vZ48d_Li10ys773lshnpv2WCmepeIR-89DUz7ZGxiEHX2_By2ER9Ce1w_04ReYRFSyV_WrvJr5KgVfqWMmmLuMME8J78o-XnzSrxeB1vEv3G6esjn0Gu4wJ3OeMCjbaxtp5mPh_unqw/s1600/irislfo.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg6vZ48d_Li10ys773lshnpv2WCmepeIR-89DUz7ZGxiEHX2_By2ER9Ce1w_04ReYRFSyV_WrvJr5KgVfqWMmmLuMME8J78o-XnzSrxeB1vEv3G6esjn0Gu4wJ3OeMCjbaxtp5mPh_unqw/s1600/irislfo.png" /></a></div>
Each layer has an LFO that can be routed to pitch, amplitude, or pan for that layer. These are the only parameters that are modulatable on a per-layer basis. Seven waveforms are available, and the onset of the LFO can be delayed using the "attack" parameter. The rate can be synchronized to tempo. When "Restart" is on, the waveform starts at zero for each note played; when it is off, the LFO is free running and all voices will have their LFOs synchronized in phase. There is also a global LFO which can be assigned to global amplitude, global pan, or the filter cutoff frequency. The global LFO's controls are in the Mix view. </div>
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<div>
The "Mod Routing" controls at the bottom right control global routing of velocity and aftertouch (they cannot be routed on an individual layer basis). Clicking on the arrow symbol next to "Mod Routing" brings up a pop-up window that allows velocity and aftertouch to be routed in varying positive or negative amounts to amplitude, depth of the global LFO, filter cutoff, and filter resonance. Note that all of these are global parameters that apply to the whole patch.</div>
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<div>
If this all seems a bit limited, it is. The saving grace is that almost every parameter that is visible in the Mix window can be assigned to a MIDI continuous controller. In the Mix window, click on the "MIDI Learn" button at the upper right, and all of the assignable parameters turn blue. Click on the parameter that you want to assign. Then, send a controller message by moving the controller of your choice (or programming it into your DAW), and Iris will associate that controller # with that parameter. </div>
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<tr><td class="tr-caption" style="text-align: center;">Mix view in MIDI Learn mode</td></tr>
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<h3>
Effects</h3>
<div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhhNBVx2pvAVtw6wIhOJAt1gcPUgDHQesogacJXMMZMok8uItgi2AIuPBsjO3TlTFkUZwywh0mKardK-ZRR_W1SPN6JRMbF4y2aI_lj85teQQSLUkwEhE_xpkBgH4cXgdLmcdezKuPMZYo/s1600/iriseff.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhhNBVx2pvAVtw6wIhOJAt1gcPUgDHQesogacJXMMZMok8uItgi2AIuPBsjO3TlTFkUZwywh0mKardK-ZRR_W1SPN6JRMbF4y2aI_lj85teQQSLUkwEhE_xpkBgH4cXgdLmcdezKuPMZYo/s1600/iriseff.png" /></a></div>
There are four effects: distortion, chorus, delay, and reverb. Only one instance of each is available. The effects can be configured in either of two modes: send effects, in which each layer can be routed to one or more effects, and master effects, in which the combined mix is routed through the four effects in series. In order to change the mode, you must select the Mix view. Then at the top right, to the right of the word "MASTER", there is a box labeled "Effects Mode" with icons for the two modes. Click on the one you want.</div>
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<div>
In Send mode, the Send Effects knobs on each layer allow you to send some amount of that layer to the effects; they work like effects sends on a mixing console. A mix of one or more layers can be sent to each effect. The effects outputs are mixed with the dry outputs of the layers to form the sum output. A slight annoyance is that before you can use an effect, you must click on the power-button icon to activate the effect. A convenience is that, from the layer views, you can click on the arrow icon to the upper right of the knob to bring up a pop-up window that allows you to adjust the effects parameters without having to go to the Mix view. </div>
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<div>
In Master mode, the four effects are in series: the distortion effect receives the summed dry mix of the four layers, and then the signal flow proceeds from left to right as shown in the mix view. In this mode, the Master Effects knobs, as shown in both the layer views and the mix view, control the dry/wet balance for each effect; if the knob is at zero, the signal will effectively bypass the effect. There is no way to control the balance of individual layers in the effects send when using Master mode.</div>
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The quality of the effects is decent. I actually found the distortion effect to be the most versatile. It allows the choice of five different distortions, ranging from the fairly mellow "Tube" to the absolutely brutal "Asymmetrical", plus an aliasing effect that did particularly interesting things to higher frequencies. The chorus is basic but functional and pretty effective for adding some depth to some of the flatter-sounding patches. The delay is also basic but it does what it is supposed to do, and it has a max delay time of 1.5 seconds on each stereo channel. The reverb was the only effect I didn't care for; I found it cloying and limited, and it tended to obscure the more subtle aspects of patches. (As you are going through the factory patches, try listening to some of them with the reverb off.) </div>
<div>
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<h3>
Macro Knobs</h3>
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiqjQr1jpvwP3so8haxbakgZwTlcY7suwPzVve6IdZf7kNNE6PF5NvJy2qaWCeBJrXvpB-fFw5VxurHAiZrBDa0eQqXzQP2rWus8yidq_RNh6Rn22Y5iVeS4o7aX3ZBAzmKIae0_r61hb4/s1600/Macro+Dialog.png" imageanchor="1" style="clear: right; float: right; margin-bottom: 1em; margin-left: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiqjQr1jpvwP3so8haxbakgZwTlcY7suwPzVve6IdZf7kNNE6PF5NvJy2qaWCeBJrXvpB-fFw5VxurHAiZrBDa0eQqXzQP2rWus8yidq_RNh6Rn22Y5iVeS4o7aX3ZBAzmKIae0_r61hb4/s1600/Macro+Dialog.png" height="320" width="111" /></a>At the lower right of the screen, in every view except the Mix view, next to the modulation routing knobs is a mechanism called the macro knobs. This allows you to assign a number of parameters which can all be changed by turning a single knob. Minimum and maximum ranges can be assigned on a per-parameter basis, and the range can be inverted so that a given parameter can be made to decrease as the knob is turned up, or vice versa. To access the knobs, you click on the pop-up icon next to the word "Macro". <br />
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The macro facility comes with some templates, called "styles", with pre-assigned parameters, or you can create your own using the parameter's pop-up menu. The way you do this is: locate the parameter you want to assign, then bring up the parameter's pop-up menu (right-click on Windows, control-click on OSX), and using the "Assign" menu. There are eight macro knobs available (which you can name, on a per-patch basis). Right-clicking or control-clicking on a macro knob brings up a pop-up menu, in which you can change the knob's name, look at the parameters assigned to it, or clear the existing assignments. (If you want to remove just one assignment, or update the range on an existing assignment, you need to go back to that parameter and access its pop-up menu.<br />
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The X-Y pad doesn't seem to work quite the way that the instructions describe. It seems to describe the two dimensions of the pad cursor movement being two additional macro knobs, but what I'm seeing is that they are hard coded to the #1 and #2 knobs -- moving the pad cursor horizontally causes the macro knob #1 to increase and decrease, and the vertical does the same to knob #2. The only thing about the pad that I see you can change is the four legends; right-click or control-click on the pad to bring up a pop-up that will let you rename them. <br />
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Izotope describes the macro knobs as being a way to quickly achieve a certain style of sound, by choosing the appropriate template, once you have chosen a patch. I'm not actually sure that's the best use for them. Yes, it may be handy sometimes for that purpose, but I see a lot more potential in using them for on-the-fly patch morphing and automation / MIDI control weirdness. The key to this: You can assign a macro knob to a MIDI controller using the MIDI learn mode. To do that, go to the Mix view, and then find the cleverly concealed pop-up icon for the macro knobs -- it's in a different place in this view, to the lower right of the Master knob. After you've done that, you can click on the "MIDI Learn" button, and then you can assign a MIDI controller to a macro knob the same way as for the other parameters.<br />
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<h3>
Conclusions</h3>
Iris is a pretty cool piece of software. This is the sort of thing for which soft synths really stand out: a synthesis method that would be all but impractical to implement in dedicated hardware, because of the cost of designing it and the need for a screen to do the editing. I've done a lot of playing with it and I feel like I've only scratched the surface; it's a totally new method of synthesis to me, and I have a lot to learn yet. The edit rendering seems to work well, without any aliasing or Fourier analysis artifacts that I've noticed. The effects, as noted above, are good with the exception of the reverb which I'm not a big fan of. I do wish there was more flexibility in the effects routing.<br />
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I only have a couple of criticisms of my own. My main one is that I don't understand why, in this day and age and in this context, the envelopes are restricted to ADSR; the standard for advanced soft synths these days is multi-segment envelopes, and Iris is one synth that could particularly benefit from them. The envelopes would also benefit from both a start delay and a time limit on the sustain phase, which would allow samples to be layered in and out on sustained chords and drones.<br />
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I would like to see more drawing tools offered for the spectral editing. Circular and elliptical shapes could be useful, as well as the ability to draw arbitrary polygons by point-to-point clicking. Similarly, the ability to erase by some means other than just freehand is desirable; I'd like to see an "anti-paint" function which works like region painting, but has the effecting of erasing any painted areas that are anti-painted over, and is capable of performing any draw operation that painting can.<br />
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A lot of the commenters at <a href="http://www.kvraudio.com/forum/" target="_blank">KVR </a>would like to see a time stretch function that can be applied to samples, so that loops can be set up to exact lengths. I haven't found a pressing need for that, though; if I need a sample time stretched, I can export it to some other software to do that job. However, the ability to set loop start and end points to exact times would be helpful. I'd also like to see the ability to set up a separate release loop, like some samplers have.<br />
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Another common request at KVR is to be able to set a volume level on a painted region, so that some choice in between full off (not painted) and full on (painted) is available for a region. Not sure what I think about that... I think that if there were multi-segment envelopes, it would go a long way towards fulfilling the same purpose.<br />
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I will say that, running Iris as an AU plug-in under Metro [version] on OSX [version], I have found the software to be completely stable. I have encountered no crashes, no functions that ceased working after a while, and no audio or visual glitches. Iris integrates well with this uncommon DAW software, so the software engineers must have done their homework regarding the AU specification. I have also run the stand-alone version and encountered no problems. I have not tried the VST installation -- although Metro accepts both, when I have a choice, I go with the AU version unless it doesn't work for some reason. <br />
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<h3>
Sound Samples</h3>
All of the below are uncompressed WAV files, so they may take a few seconds to load. <br />
<ul>
<li>Factory patch: <a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/blackgalaxy.wav" target="_blank">Black Galaxy</a>. Demonstrates Iris' capability for creating slowly evolving sounds.</li>
<li>Factory patch: <a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/toddlersq.wav" target="_blank">Toddler Squarepusher</a>. Chosen mainly because I liked the name, and it does fit.</li>
<li>My patch: <a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/zombiealarm.wav" target="_blank">Zombie Alarm</a>. Shows the ability to do rhythmic repeating sounds.</li>
<li>My patch: <a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/coffeejack.wav" target="_blank">Coffee Jackhammer</a>. Just plain screwy. </li>
</ul>
</div>
</div>
Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-1797040744060089262014-10-04T15:42:00.000-05:002014-10-04T15:42:34.581-05:00The OTA, and why it's important<div>
Many (in fact, most) analog synthesizers rely heavily on an integrated circuit called the <i>operational transconductance amplifier</i>, or OTA for short. The OTA is a variation on the common operational amplifier, or op-amp. From an electrical engineer's point of view, the main difference between an op-amp and an OTA is that, while an op-amp outputs a voltage that is proportional to the difference in voltage between its two inputs, the OTA outputs a <i>current</i> that is proportional to the difference in voltage between its two inputs.</div>
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This, however, is not what makes an OTA valuable from a synth designer's perspective. Analog synths rely on two behaviors in which the OTA differs from the op-amp:</div>
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<div>
1. The OTA has a third input, called the amplifier bias current or Iabc for short. (The letter I is the standard electrical engineering symbol for current.) With the conventional op-amp, the gain is determined by an external feedback resistor. The OTA's gain is set by the amount of current that is allowed to flow into the Iabc input.</div>
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2. The OTA's input impedence is proportional to the current flowed into the Iabc input.</div>
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What does this mean, in practical terms? Consider the first: It's a simple matter to convert a voltage to a current, or vice versa. So if you take an external voltage input, convert it to a current, and feed that to the Iabc input to control the OTA's gain -- well, that's what a <a href="http://electronicmusic.wikia.com/wiki/Voltage_controlled_amplifier" target="_blank">VCA </a>is. Another way of looking at it is that the OTA is an analog multiplier; the difference between the two input signals is multiplied by the Iabc input. That opens up all kinds of possibilities, such as ring modulators and wave shaping circuits.</div>
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</div>
<div>
As for the second characteristic, if a voltage is converted to make the Iabc input, then the OTA becomes, in effect, a voltage-controlled resistor. As it happens, a voltage-controlled resistor is a very handy thing to have when designing <a href="http://electronicmusic.wikia.com/wiki/Voltage_controlled_filter" target="_blank">VCF </a>circuits. </div>
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</div>
<div>
Now the problem: Other than analog synth circuits, there are few uses for OTAs these days. There was a time when OTAs were widely used in radio and microwave systems, but digital signal processing has taken over in these applications. Consequently, OTA integrated circuits are disappearing from the market. A survey of the casualties:</div>
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</div>
<ul>
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<li>Intersil CA3080: Originally designed by RCA and introduced in 1969. A good, basic OTA cicruit widely used in both commericial and DIY designs over the years. Because it was produced in huge numbers, there's some stock still around and it's still possible to find them. But of course, that stock is a finite number and it will run out eventually. (<a href="https://www.rocelec.com/parts/results/all/?s=3080" target="_blank">Rochester Electronics</a> still has some, although they aren't cheap.) </li>
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<li>Intersil CA3280: Long regarded as the Cadillac of OTA designs (and often priced accordingly), this was also widely used particularly in top-end VCA designs. Blogger Don Tillman made a <a href="http://www.till.com/blog/archives/2005/06/last_of_the_ota.html">heroic effort </a>to rescue the 3280, but was unsuccessful. Because of the higher price, it was probably not produced in as large a quantity as the 3080, and existing stocks have nearly dried up. </li>
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<li>Intersil CA3060 and 3094: These were variants of the 3080. I believe the 3094 was a dual OTA with buffered outputs, and the 3060 was a triple OTA. They are both long out of production and there are no sources that I know of. </li>
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<li>National LM13600: I think this part was used mainly in DIY designs. It can still be found, but except for repair of existing circuits, it isn't as widely sought out because it's not the easiest part to incorporate in a circuit, and it's specs aren't as good. The still-in-production LM13700 is better in both respects. (Internet rumor has it that the original design was undertaken by an intern, as an exercise.) </li>
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<li>Rohm BA662: Heavily used in '70s and early '80s Roland gear. Long out of production. There is no source that I know of.</li>
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<li>Rohm BA6110: Widely promoted as an "almost pin compatible" substitute for the BA662 after the latter was discontinued. However, the 6110 is also now discontinued, and existing stocks have nearly dried up.</li>
</ul>
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There are only two OTA ICs in production at this point, the above-mentioned National LM13700 and the On Semiconductor NE5517. The LM13700 is regarded as a decent design, but not as well spec'ed as the 3280 and not quite as versatile as the 3080. But by default, it is the part of choice for new designs. The NE5517 is a bit of a story; it was formerly manufactured by Phillips, but the fab in France that manufactured it burned down in 2003. On Semiconductor then purchased rights to the design and picked up manufacturing. I've seen some Web references that say that the On Semi parts aren't spec'ed as tightly as the former Phillips parts.<br />
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So where does all this leave us? We have no indication as to what National's future plans are, regarding how long the LM13700 will be in production. Probably for a while, since there is little competition now, but one never knows. There are people in the synth design world who know how to design OTA circuits from discrete components, and to name one example, the Synth Tech <a href="http://www.synthtech.com/motm440.html" target="_blank">MOTM-440</a> uses just such a circuit. But it's not as DIY-friendly an approach; there are a lot of subtleties to designing a good OTA circuit, and it's more expensive and more trouble then using an IC. But as far as technological progress goes, it's definitely a step backwards. Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-77012707017255455252014-08-24T21:20:00.001-05:002014-08-24T21:20:55.252-05:00Miscellaneous thoughts for AugustSome things I've taken note of recently:<br />
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<h2>
Buchla Music Easel back in production</h2>
After about 40 years, the <a href="http://buchla.com/shop/music-easel/" target="_blank">Buchla Music Easel</a> is back in production. The original Music Easel was produced in the mid-1970s; it basically consisted of a <a href="http://electronicmusic.wikia.com/wiki/Semi-modular" target="_blank">semi-modular</a>, two-VCO analog monosynth coupled to a 2-1/2 octave, capacitive touch sense keyboard with variable tuning and scaling. <br />
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Both the new Easel, and most copies of the original, took nearly all of the internal circuits out to a big edge connector called the "program card" connector. Back in the day, Buchla sold cards that plugged into these connectors that the (soldering-iron-equipped) user could use to program patches. This was done by soldering resistors onto the cards to make certain connections. Unknown to most people, though, the program card connector contained additional interconnects that made it possible to actually play the Easel remotely. Buchla at the time was doing some experiments with computer control, e.g., the Model 700 that connected to a <a href="http://www.psych.usyd.edu.au/pdp-11/" target="_blank">DEC PDP-11 minicomputer</a>. However, no interface for the Easel ever reached production.<br />
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Now, 40 years later, it has. Buchla has introduced the <a href="http://buchla.com/shop/buchla-music-easel-iprogram-card/" target="_blank">iProgram </a>interface card, which connects via wireless Ethernet to an iPad. Using software supplied by Buchla, the iProgram can function as both a patch editor/librarian, and a MIDI interface. That's not a bad trick, and it says something about the Easel's original design that it can be interfaced to something that would not be developed for another 40 years. <br />
<h2>
The Return of Richard D. James</h2>
The first album of all-new Aphex Twin material since 2001 is nigh. Syro, announced by means of <a href="http://www.synthtopia.com/content/2014/08/21/aphex-twins-syro-from-influential-electronic-fartist-richard-d-james-now-official/" target="_blank">this bizarre press release</a> (via Synthtopia), will be available in September. Oddly enough, I didn't realize it had been that long.<br />
<h2>
Modulation Wheels</h2>
I've been doing a lot of patching with the <a href="http://johnbowen.com/solaris-overview.html" target="_blank">Solaris </a>recently. Now, I "grew up" as a synth performer using mostly Roland keyboards (that is, I've owned lots of synth, but the ones of other makes have been mostly rackmount units. Of the ones I've had with actual keyboards, most have been Rolands.) So I've long since grown accustomed to the Roland "T-handle" control for pitch bend and modulation, including the fact that it's spring loaded in both the pitch and modulation axes. <br />
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The Solaris, on the other hand, has traditional pitch and modulation wheels, and the mod wheel is not sprung. I've often found this vaguely disconcerting. However, last week, I was working with a three-oscillator bass patch I had built, and I wanted to assign a performance control to detune one of the oscillators. Assigning it to aftertouch did not do what I wanted. I was going to use a soft knob, and then it occurred to me -- assign it to the mod wheel. By doing this, I was able to put it in various positions to achieve the amount of detne that I wanted at different times. This really led me to appreciate the virtues of a mod wheel that stays where you put it.<br />
<h2>
Stuff I Want</h2>
I've got my eye on Madrona Labs' <a href="http://madronalabs.com/products/kaivo" target="_blank">Kaivo </a>plug-in. I've long wanted a <a href="http://electronicmusic.wikia.com/wiki/Physical_modelling" target="_blank">physical modeling</a> synth that provides the performer with access to the parameters inside the guts of the modeling algorithms. Kaivo appears to be all that and a lot more. I'm also looking for something that will give me a much larger variety of tuned and semi-tuned percussion sounds that one will get from the typical drum machine, and it looks like AAS' <a href="https://www.applied-acoustics.com/chromaphone/overview/" target="_blank">Chromaphone </a>is just the thing. <br />
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I've also gotten interested in some of the effects boxes from Strymon, particularly the <a href="http://www.strymon.net/timeline/" target="_blank">Timeline </a>delay, which has MIDI control. I can think of all kinds of possibilities for that. I've read that Strymon includes a lot of ex-Alesis employees, from the time when Alesis was doing good stuff (before Keith Barr lost control of it), so that's a good sign.<br />
<h2>
Next Week</h2>
I have a review of Izotope's <a href="https://www.izotope.com/en/products/effects-instruments/iris/" target="_blank">Iris </a>spectral editor, which I'll try to have up by this weekend.<br />
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-78971093121774238132014-07-11T23:13:00.002-05:002014-07-11T23:13:22.089-05:00Reviving M1000XI've finally found the time to get Xcode going on my Mac Mini and rebuild M1000X, the program I published some years ago for doing patch editing on the Oberheim Matrix-6/6R and Matrix-1000. It now runs on Intel Macs under OSX 10.8 (it should also work under 10.9, but I have not tried it). There were some challenges involved in getting it going again. (OSX coding geekdom ahead...) I originally wrote M1000X in 2005 in pure C, using the Carbon libraries. Apple in its infinite wisdom has since deprecated Carbon, and is pressing hard to get all remaining Carbon apps converted to use the Cocoa libraries. Unfortunately, making a Carbon app written in C work with Cocoa involves a significant re-write, object-izing the code and reworking it into Objective-C. Additionally, the current version of the interface builder built into Xcode refuses to open the Carbon nib files, although the runtime still works with them. Fortunately, I still have my old PowerPC Mac up and running, so I was able to do the interface resource editing that I needed to do there and copy the nib file over to the Intel-based machine. Additionally, due to a problem with the way that bit fields in structures are specified in C (a problem that the language has had since the original K&R specification in the early '80s), I had to re-write some large structures that handle patches in memory. While I was at it, I took the opportunity to fix a couple of bugs, and to put code in to work around two bugs in the Matrix-1000 OS that I didn't become aware of until after I had released version 1.0.<br />
<br />
Anyway, version 1.1 of M1000X is available <a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/M1000X_home_1_1.html" target="_blank">here</a>. Just open the disk image file and drag the application to your Applications folder. There is a help file in there and a set of release notes also.<br />
<br />
So while we're here, this is a good place to talk about the Oberheim Matrix series of synths. These consist of two groups -- the Matrix-12 series, and the Matrix-6 series. The Matrix-12 series consists of two synths, the Matrix-12 and the Xpander, which basically is a "tabletop" Matrix-12 with no keyboard and half as many voices. Both of these synths are now very expensive collectors' items, and they also don't have a lot in common with the Matrix-6 series, so I won't dwell on them further here.<br />
<br />
The Matrix-6 series consists of the Matrix-6 itself, a 6-voice keyboard; the Matrix-6R, a rack-mount (2U) packaging of the Matrix-6, and the Matrix-1000, with the same voice architecture but more patch memory and some improvements, and packaged in a smaller (1U) rack-mount box. All of these synths are based on the CEM 3396 "synth on a chip", which provides an all-analog signal path consisting of two DCOs, one low-pass VCF, one noise generator, and two VCAs. The VCF can be FM'ed by one of the DCOs. The synth adds to this a bevy of modulation and signal sources: two LFOs, three DADSR (delay-attack-decay-sustain-release) envelope generators, two ramp generators, and a tracking generator. The patch memory consists of 100 patches on the 6/6R. The 1000 has 800 patches in ROM and 200 in RAM (hence the name).<br />
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The "matrix modulation" capability is what gives the synths their name. A radically different idea at the time it was introduced, the modulation matrix is basically a virtual version of the old EMS pin matrix (which is displayed graphically by M1000X). There are ten "pegs" that you can insert at any intersection in the matrix, and you can set the level (inverted or non-inverted) of signal to be routed from the selected input to the selected output. All of the internal modulation generators are available as sources, as are external expression input jacks and MIDI continuous controllers. Almost any parameter of the DCOs and VCF can be a matrix destination, as well as most of the envelope segments, the LFO rates and amplitudes, and the portamento rate. It really did at the time help break out of some of the limitations of its contemporaries, which at the time usually had very limited signal routing -- for instance, on many synths there was one LFO that could only be routed to oscillator frequency, and maybe one other one that could only be routed to VCF cutoff or VCA level.<br />
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While the voice architecture is quite capable, the user interfaces for patch editing are limited. The Matrix-6 and 6R both use the same mechanism, a membrane panel with three mode select buttons, a numeric keypad, and some forward and back buttons. Visual output is by way of a one-line vacuum fluorescent display with 16-segment characters; it's bright and easy to read (except for some punctuation characters that come out looking funky in the 16-segment format), but it can only display 20 characters. The display basically switches from display of patch numbers and names to displaying names and values of patch parameters, and patch editing is quite tedious. As for the Matrix-1000... well, it has no patch editing controls at all. Clearly there was a cost and packaging decision at some point (apparently Oberheim really wanted to get the Matrix-1000 crammed down to a 1U package). The only thing you can do from the panel is select patches, set up MIDI, and change a few system parameters. Output is limited to a 3-digit numeric LED display and a few status lights. This is why you need an external editor. <br />
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Yes, these are analog synths. They do use DCOs, and some (most?) of the modulation signal generating is digital, but the audio signal path is all analog. Oberheim produced many of these synths, and because of the limited user interface they are regarded as less desirable than many other analog synths, so they can be had at reasonable prices. If you're looking to get into vintage analog inexpensively, this is a good way to do it -- get one of these synths, and then use M1000X to overcome the UI limitations.<br />
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com1tag:blogger.com,1999:blog-279251740880432906.post-63845324476460162972013-08-25T08:48:00.000-05:002013-08-25T08:48:16.058-05:00How the Korg Poly-800 DCO worksBack when I wrote an article here about how the <a href="http://electronicmusic.wikia.com/wiki/Digitally_controlled_oscillator" target="_blank">digitally controlled oscillator (<span id="goog_951579756"></span>DCO<span id="goog_951579757"></span>)</a> in the Roland Juno synths (the old analog ones, not the Juno-D series) works. At the time, I had intended to also write something about how the DCO in the Korg Poly-800 (and its rack-mount sibling, the EX-800) works. I had heard from several sources that it was quite a different design from the Juno DCO, but at the time I wasn't able to find any solid technical information. And I don't actually own a Poly-800, so I didn't have a guinea pig to experiment on. (Years and years ago, I tried one out in a music store, and to be honest I was not that impressed. But back to the topic.)<br />
<br />
Recently, I got interested in the topic again, and after a few days, I managed to finally uncover some documentation. And yes, as it turns out, the Poly-800 DCO is quite different from the Juno. It actually flirts with the line between "analog" and "digital" a lot more than the Juno DCO (which has a completely analog audio path) does. And in some ways, it's more capable than the Juno DCO, but in other ways it's quite limited and just plain screwy.<br />
<br />
<b>Schematic Archaeology</b> <br />
<br />
A few months ago I went to the excellent FDISKC web site and downloaded a copy of the Poly-800 schematics. I recall looking at this before and not being able to make much sense of it, and it doesn't help that it's a scanned copy of an original that was already in rather poor condition. But this time, having read up a bit more on the synth's features and its patch programming options, I had a better idea of what to look for. For those who have not encountered one: The Poly-800 is an eight-voice synth. Its DCOs generate outputs in four octaves for each voice. There are 16', 8', 4', and 2' octaves that can be turned on and off individually. There are two choices of waveform (or so the synth likes to pretend; we'll talk about this later): square and sawtooth. No pulse width modulation on the squares, and no triangle or sine. The normal operating mode is a single DCO per voice, but the 800 can be put in a "double" mode wherein two DCOs are allocated to each voice, the penalty being that the synth is reduced to four voices.<br />
<br />
When I looked over the schematics, I noticed an IC with the part number MSM5232. It had two groups of outputs marked as being the four footages mentioned above. Aha, I thought, that must be the IC that generates a voice, or possibly two voices. I got to looking for some notation on the schematic that would explain that that part of the circuit was replicated some number of times (4 or 8 was what I expected), but I couldn't find any such. Also, the IC looked like it was maybe some sort of processor; it had incoming address and data lines. And then there were eight lines marked as "C1" through "C8". I couldn't figure out what those were. A bit of Web searching quickly uncovered that this part was once upon a time made by Oki Electric. However, Oki Electric spun off its semiconductor business into a separate company some years ago; I think it may have been through several changes of hands since then, and in any event, Oki Semiconductor, if it still exists, doesn't seem to have a Web site. So no going to the manufacturer for a data sheet.<br />
<br />
I saw several mentions of the Poly-800 service manual having the data sheet, but I only turned up a couple of online sources for the manual, and they looked sketchy (they demanded that you disable your firewall and virus protection in order to download). So no luck there. After hours of searching, I finally found a several-years-old posting that had a pointer to an Italian site. I crossed my fingers and clicked. It was there! And it explains a lot. And now I know...<br />
<br />
<b>The Original Chiptunes Synth</b> <br />
<br />
The reason I couldn't find any block-replication notation on the schematics was that a single MSM5232 handles all eight DCOs. As it turns out, the MSM5232 wasn't intended to be used in music synthesizers -- it was a tune chip for arcade video games. It contains eight counters that divide down a pair of master clock inputs, and bit shifters that act like octave dividers and produce all of the different footages. It also has a sort-of VCA for each voice, and a pair of onboard attack-sustain-release envelope generators. What it does not have is filters, a problem that we'll get to later.<br />
<br />
So here's how it works: Each DCO is, as stated above, has a counter-divider that is loaded with a value and then counts down every time the clock signal at the external clock input cycles. When it reaches zero, it sends a reset pulse, and then its value gets reloaded again.. This much is similar to the Juno DCO. On the Juno, each time the counter reaches zero, the pulse resets a fairly conventional sawtooth VCO core. However, the 5232 has no VCO core. Instead, it has a flip-flop that toggles its state on every counter reset -- which means that it is generating a square wave. That's the only waveform it can produce. <br />
<br />
Each DCO has a register into which the CPU places a note number when the DCO is to play a note, and a gate flag that turns the voice on and off. The note number is used to look up a counter value from an internal ROM, which will be used to divide down the incoming clock frequency. The flip-flop controlled by the counter drives a chain of octave dividers which generate the four footage outputs. Basically, there is only one octave's worth of counter values, and it taps into the octave divider chain in different places for higher or lower octaves. <br />
<br />
So far so good. Now here's where it begins to get screwy. One would think that the logical way to output the voices from the chip would be to have each voice output on its own output pin. That's not what it does. The voices are divided into two groups, and for each group, all of the outputs of a given footage are mixed onto one output pin; for example, all of the 16' footages for voices 1 through 4 come out mixed on pin 28. This answers a big question that is often asked about this synth: why does it use a paraphonic VCF? Answer: because the 5232 doesn't make the individual voice outputs available. The IC provides amplitude control over each voice, but not over the individual footages -- they can only be turned on or off, and the choice applies for all voices in a group. There are two ASR envelope generators onboard, one for each group, but the Poly-800 does not use them. Rather, it applies envelopes generated externally by the synth's CPU. These are input to the chip through eight input pins, one for each voice. I don't think the chip really has VCAs -- I think that all it does is toggle back and forth between the current envelope level and ground, which produces the square wave of the desired amplitude. <br />
<br />
Each group of four voices is driven by its own external clock source. The chip itself has no mechanism for any kind of pitch modulation, so pitch bend and envelope/LFO control over pitch have to be implemented external to the chip, by modulating the master clock frequencies. Each group of four voices has its own master clock input. This is reflected in the Poly-800's architecture; if you put it in the "double" mode, it divides the two groups and drives them with clocks of different frequencies when detune is selected.<br />
<br />
The drawing below shows the basic signal flows. (To reduce drawing clutter, only 4 of the 8 voices are shown.) Each voice consists of a note number register, a counter/divider, and four octave dividers. To play a note, the synth chooses a voice, writes the desired note number into its note number register, and then sets a flag telling the voice to play. The note number is used to look up the divide-down count in the ROM, which then goes to the counter/divider. This divides down the master clock (not show) for the group that the voice is in (purple or green) and produces the top octave. The four octave dividers then produce the four footages. <br />
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<b>The Mysterious Sawtooth Wave and Alleged Walsh Functions</b><br />
<br />
This leaves a big question: we've established in the MSM5232 is only capable of generating square waves. But the Poly-800 provides a choice of square or sawtooth waveforms. How does it do that? You may have read something about the Poly-800 using a mathematical technique called "Walsh functions" to generate the sawtooth. What's a Walsh function? Well, you might know that the process called the "Fourier transform" breaks up a waveform into a set of sine waves that are mixed at different amplitudes. Walsh functions are like the sine waves used in Fourier analysis: by adding together a set of Walsh functions at different frequencies and amplitudes, you can re-create an arbitrary waveform, within a certain bandwidth. And that's what the Poly-800 does to approximate a sawtooth wave: it uses the four footages of square wave that the DCO produces to do the inverse Walsh transform equivalent. When you have the "square" waveform selected for the DCOs, the four square-wave footages are mixed together in equal amounts before the composite signal goes to the filter. However, when "sawtooth" is selected, the four footages get routed into an analog adder circuit that adds them in a proportion such that the output roughly resembles a sawtooth. We say "roughly" because trying to do Walsh transforms with only four functions is about like trying to do additive synthesis with only four harmonics. (Further, it's not true that all of the Walsh functions are square waves; only some of them are, and it takes a more complete set to do a good Walsh transform.) Nonetheless, it does sort of produce a sawtooth wave. <br />
<br />
I've still got a lot more digging to do into the schematic. For one thing, I'd like to be able to identify how the source oscillator that produces the two clock signals for the 5232 works. It's obviously not a crystal oscillator since it has to be variable in frequency to an extent. It appears to be based on an LC-type resonant circuit, but that part of the schematic is in particularly bad shape and it's hard to read. <br />
<br />
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com2tag:blogger.com,1999:blog-279251740880432906.post-90718244410815592342013-08-24T21:01:00.000-05:002013-08-24T21:01:29.391-05:00Statescape WisconsinSo as I wrote in my last post, I've been looking for a while for a way to build a delay line that would allow the recirculating sound to interact with the sound being input in a way other than just being mixed together. I have wanted to explore other ways in which the input sound could modify the sound looping through the line. One thing I thought of was to build a delay line in which the input <a href="http://sequence15.blogspot.com/2013/08/amplitude-modulation.html" target="">amplitude modulates</a> the recirculating signal. As you might know, for any pair of sine waves that are input to a form of amplitude modulation, the output will contain sine waves at two new frequencies which are the sum and difference of the frequencies of the two inputs. If you use more complex waveforms, then each pair of component sines contained within the two input signals will produce sum and differences frequencies, which can produce a whole lot of partials in the output . Here is a block diagram of what I had in mind:<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiwbS9x2sTlr0XT3LCfSHlGF9rqH_dd1YIVp0j6qI0GgxYMW9tUGixUDErly4rZi0N_oM_DJ4Sb8aj72N5YQI3VlXX8OQ7zrqySlG6PiHfTGBcT6PX9rDhoFbmH3o2B6d3M9XiiXV5Rzc8/s1600/AM+Drawing.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="276" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiwbS9x2sTlr0XT3LCfSHlGF9rqH_dd1YIVp0j6qI0GgxYMW9tUGixUDErly4rZi0N_oM_DJ4Sb8aj72N5YQI3VlXX8OQ7zrqySlG6PiHfTGBcT6PX9rDhoFbmH3o2B6d3M9XiiXV5Rzc8/s320/AM+Drawing.png" width="320" /></a></div>
<br />
<br />
The first question was how to actually build such a delay line, and for me the answer was obvious: my favorite softsynth-building environment, <a href="http://www.csounds.com/">Csound</a>. I've coded up a number of delay lines in Csound previously, and the only big change here was to incorporate the amplitude modulation function into the feedback loop. However, getting that to work the way I wanted proved to be more difficult than I though at first. To illustrate why, I'll repeat the basic amplitude modulation calculation from my last post:<br />
<br />
A = (IG + M) * C<br />
<br />
where: M is the modulation signal, C is the carrier signal, IG is the
initial gain for the carrier (or, to put it another way, the magnitude
of the output when no modulation is present), and A is the
amplitude-modulated output. The problem here is the fact that when you first start up a delay line, it contains no signal. As you can see in the equation, if the carrier C term is zero, there is no output. So obviously if the AM process is implemented with the delay line feedback as the C term, the sound building process can never get started because no AM output is ever generated. <br />
<br />
So I tried coding it the other way, treating the input signal as the carrier and the delay line feedback as the oscillation. That solves the problem with the delay initially not containing anything; it gets filled with unmodified input signal until something starts wrapping back out of the line and amplitude-modulating with the input. However, it creates another problem: there has to be an input signal present all the time. Whenever there isn't, the AM output, and the signal getting fed back into the delay line, gets "blanked". And that's bad because I've found that, when doing these long-period delay things, it pays to be sparse with the input; if you are playing notes into it all the time, it quickly gets too busy for the listener to make any sense of it.<br />
<br />
I thought about going back to the first way, with the delay line feedback as the carrier, but with a software switch that would route unmodified input signal into the line whenever there was no output from the AM processing. But what I wound up doing was simpler: I computed the modulation both ways and added the results. This doesn't effect which frequencies are present in the output, only the relative levels. For the purpose, I decided it was good enough. And this had the advantage of not going silent whenever one signal or the other wasn't present.<br />
<br />
Once that problem was solved, the next problem was to figure out what kind of input signals would produce interesting results. I tried some standard synth things like PWM leads and pad sounds, and I found out right away that with those harmonically complex sounds, the results degenerated into a particularly nasty-sounding form of noise very quickly. So I had to have something harmonically simpler. For this purpose I chose the <a href="http://www.vintagesynth.com/kawai/kawaik5.php">Kawai K5m</a> <a href="http://electronicmusic.wikia.com/wiki/Additive_synthesis">additive synth</a>. This was sort of overkill, but it worked for the purpose. I built one basic sound with only a few harmonics, and capable of having its harmonic content varied by use of the <a href="http://electronicmusic.wikia.com/wiki/Mod_wheel">mod wheel</a>. <br />
<br />
I ran into a few problems, including one that I never manged to solve. The big one was a puzzling popping noise that appears at random times. I still haven't figured this one out. Also, I had some problem with subsonics appearing in the output. To address both of these problems, I added a pair of two-pole Butterworth filters to the algorithm, a low pass and a high pass. These didn't totally solve the popping problem, which you can still hear in places in the completed track. <br />
<br />
As for the results: they were surprisingly musical. The AM often added notes that I didn't play, and I was pleasantly surprised at how often the added overtones actually worked well with the notes that were played. Keeping everything harmonically simple helps a lot. There is a distorted sound that builds up when things get busy; it seems to be characteristic. All in all, I was fairly pleased. Now I have to think of what to do for the next delay line.<br />
<br />
Listen to Wisconsin <a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/Wisconsin.mp3" target="_blank">here</a>.<br />
<br />
And here is the Csound source code for the delay line:<br />
<br />
<br />
<pre>; Basic stereophonic delay line
itimel = 3.1 ; left channel delay time
itimer = 4.2 ; right channel delay time
ifbl = 1.7 ; left channel feedback (keep < 1)
ifbr = 1.7 ; right channel feedback
kcutlo init 20.0 ; hi-pass for damping subsonics
kcuthi init 2500.0 ; low-pass for suppressing pops
imodindex init 10
kleftch init 3 ; channel # of left channel (right is assumed +1)
afbl init 0
afbr init 0
; Get input audio
ainl, ainr inch kleftch, kleftch+1
; Scale values to -1..+1 range needed by formula
ainlscaled = ainl / 0dbfs
ainrscaled = ainr / 0dbfs
afblscaled = afbl / 0dbfs
afbrscaled = afbr / 0dbfs
; Compute with feedback as carrier and input as modulation, and rescale
amodinl = (1 + imodindex * ainlscaled) * afblscaled * 0dbfs / 2
amodinl butterhp amodinl, kcutlo
amodinl butterlp amodinl, kcuthi
amodinr = (1 + imodindex * ainrscaled) * afbrscaled * 0dbfs / 2
amodinr butterhp amodinr, kcutlo
amodinr butterlp amodinr, kcuthi
; Compute with input as carrier and feedback as modulation, and rescale
amodfbl = (1 + imodindex * afblscaled) * ainlscaled * 0dbfs / 2
amodfbl butterhp amodfbl, kcutlo
amodfbl butterlp amodfbl, kcuthi
amodfbr = (1 + imodindex * afbrscaled) * ainlscaled * 0dbfs / 2
amodfbr butterhp amodfbr, kcutlo
amodfbr butterlp amodfbr, kcuthi
; Push samples through the left and right delay lines
aoutl delay amodinl+amodfbl, itimel
aoutr delay amodinr+amodfbr, itimer
; Output direct + delayed audio
outch kleftch, (ainl+aoutl)*2
outch kleftch+1, (ainr+aoutr)*2
; Compute feedback for next cycle
afbl = aoutl * ifbl
afbr = aoutr * ifbr
</pre>
<pre>
</pre>
<pre>endin</pre>
Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-39769005186904057622013-08-07T22:44:00.002-05:002013-08-07T22:44:55.574-05:00Amplitude ModulationI've got a new Statescape to post this weekend. It relies heavily on amplitude modulation, as I'll explain in the post when I post it. However, before I do that, I figured this would be a good time to dig into what amplitude modulation is, how it works, and what can be done with it.<br />
<br />
So what is amplitude modulation? Quite simply, it is what you are doing when you feed an LFO or other signal into the control input of a VCA: the amplitude of one signal (the control signal) is modulating the amplitude of another signal (the audio input to the VCA). We do this all the time without thinking about it as "AM" as such. However, most of the time, when we do this we are using very slow control signals -- well below audio frequencies. Because of this, we don't usually hear the spectral artifacts that AM creates. If we hear them at all, we hear them as a beating or phasing effect rather than as a separate tone.<br />
<br />
However, we can use an audio-frequency signal as the carrier. When we do, we find that we no longer hear the modulation as a variation of the output level of the carrier; what we hear instead is the carrier with the addition of "sideband" tones generated by the AM process. Consider the simple case where the carrier and modulation are both sine waves. What will be heard as the output of the AM process are three tones: a tone at the carrier frequency, and two sideband tones having frequencies which are, if the carrier frequency is CF and the modulation frequency is MF:<br />
<br />
CF + MF<br />
CF - MF<br />
<br />
So if the carrier frequency is, say, 500 Hz, and the modulation frequency is 220 Hz, the two added tones will come out at 280 Hz and 720 Hz. Obviously, these frequencies are not harmonically related to the carrier signal or to each other. Such will usually be the case with AM; the generated tones will be inharmonic more often than not. The audible effect is to produce sounds that are often described as bell-like, percussive, noisy, or just plain weird. If the carrier and/or modulation are more complex signals with many harmonic overtones, each harmonic of the carrier will play off of each harmonic of the modulation and generate a pair of sideband tones. The result becomes cluttered pretty quickly, which is why, when playing with AM, it is often better to start with harmonically simpler signals<br />
<br />
(What, you might ask, happens if the carrier frequency is 220 Hz and the modulation is 500 Hz? Well, the "negative frequency" values become aliased -- they come out as real tones, but with opposite phase. In this example, we'd get a "real" frequency of 720 Hz and a "negative" frequency of -280 Hz. The 280 Hz sideband will in fact be there, but it will have the opposite phase that it would have in the first example.)<br />
<br />
In a conventional AM setup (as would be used by a radio station broadcasting an AM signal), an initial gain is assigned to the carrier and the modulation varies this gain by being added to or subtracted from it. The sum or difference of the modulation and the initial gain is what modulates the carrier. The effect of this is to set the output level of the carrier when there is no modulation. The instantaneous value of the modulation increases or decreases the initial gain, depending on how the modulation wiring is set up. Ring modulation is actually just a special case of amplitude modulation, in which the initial gain of the carrier is zero. Those who have played with a proper ring modulator (one that has both carrier and modulation inputs) know that if you don't put anything into the modulation input, you get nothing out. This is why.<br />
<br />
The basic amplitude modulation equation is:<br />
<br />
A = (IG + M) * C<br />
<br />
where: M is the modulation signal, C is the carrier signal, IG is the initial gain for the carrier (or, to put it another way, the magnitude of the output when no modulation is present), and A is the amplitude-modulated output. The multiplying of the carrier and modulation signals is a characteristic of all amplitude modulation methods. Don't confuse this with the effect in the frequency domain (where the frequencies are added, as discussed above); in the time domain, the signals multiply. As you can see, if the initial gain is zero, the computation reduces to a straight multiplication of the two signals, which is what ring modulation is. You can also see another characteristic of ring modulation: the carrier and the modulation are interchangeable; switching the two inputs of a proper ring modulator will produce the same result. <br />
<br />
Amplitude modulation can be easily accomplished in both the analog and digital domains. In the digital world, if you have access to something that allows you to run formulas on samples, like Csound or Max/MSP, it's pretty easy as shown by the above equation. In the analog domain, you need a "four quadrant" VCA or a ring modulator. With the latter, you can set the initial gain (if desired) by using a voltage source and adding it to the modulation with a DC-enabled mixer. (Note: This may not work with a diode-ring-type ring modulation circuit. I don't have one to try it with, so I don't know. It should work with most any four-quadrant VCA.) Because of the creation of the inharmonic sideband tones, you want to keep the signals you use harmonically simple, because complex waveforms tend to deteriorate into undifferentiated noise pretty quickly. Also be prepared to do some low-pass filtering to get rid of any excessively high frequency tones that are generated.<br />
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-85384471080179925402013-03-06T22:06:00.001-06:002013-03-06T22:06:07.184-06:0048 ribbon controllers under your fingersI'm always on the lookout for unconventional keyboards and alternate controllers. Today, I came across the <a href="http://www.endeavour.de/evo/overview.html" target="_blank">Evo keyboard</a> from Endeavour of Germany. This is basically a keyboard with a mini ribbon controller built into each key -- the main body of the key, with the exception of the very end closest to the player, contains a short length of position-sensitive capacitive material which can generate a digital control signal based on where it is touched. Here's a photo, appropriated from Endeavour's Web site:<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEj6XudIAyOV0we86IXgVSxSwlNAhAU4IcsA886sBogWZLsTPmYRgZkRriZMLW7xw9dpRC_SQU0NOcUNyqGoMWHS6UohQ7ZjL28usBwztm9OnlzSRvddy9JhONlXldRcBBjwi3H0Xm9isF8/s1600/stacks_image_444.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="213" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEj6XudIAyOV0we86IXgVSxSwlNAhAU4IcsA886sBogWZLsTPmYRgZkRriZMLW7xw9dpRC_SQU0NOcUNyqGoMWHS6UohQ7ZjL28usBwztm9OnlzSRvddy9JhONlXldRcBBjwi3H0Xm9isF8/s320/stacks_image_444.png" width="320" /></a></div>
It may not be obvious from the photo, but the keys are longer and a bit wider than normal, in order to have sufficient room for the sensors. Each sensor is 4 cm, or a bit more than 1.5" long, and is set back from the near end of the key by 1 cm. Note the absence of conventional pitch or mod wheels. The software that comes with the Evo allows you to map the sensor on each key to a different controller; for example, the low C key could be pitch bend, the D key modulation, the E key master volume, and so on. The key doesn't actually have to be pressed in order to activate the sensor, which raises some interesting possibilities. One mentioned Endeavour's Web site would be using the keyboard as a control surface for a mixer, by mapping a group of key sensors to faders.<br />
<br />
The Evo keyboard itself does not generate MIDI data directly; it uses a protocol that is proprietary to Endeavour. It must be connected to a computer in order to function. The interconnect is not USB, as one might expect; it is -- surprise -- 10baseT Ethernet. This is actually a very cool idea, and I've been wondering if/when the day would come when a lot of electronic instruments and controllers would use Ethernet for interconnect. Ethernet hardware is pretty cheap these days, and it doesn't have a lot of the limitations that USB has, e.g., 100m length limit for Ethernet vs. 10 feet for USB. And Ethernet has far more than enough bandwidth for the application. <br />
<br />
What if you want to control a MIDI synth with it? There is a software package that you can download from Endeavour that maps control messages from the Evo to MIDI messages. An interesting feature of this software is that you can play a chord, of up to 16 notes, and it will transmit each note on a separate MIDI channel -- the so-called "Mode 3" or "guitar mode" method of sending MIDI information. The nice thing about this is that it allows control information to be note-specific if desired; for instance, if the sensors are mapped to pitch bend, then sliding one finger will send MIDI pitch wheel messages only over the one channel assigned to that note, and only that note will bend. Of course, to make this work, you have to be controlling a synth which is 16-part <a href="http://electronicmusic.wikia.com/wiki/Multitimbral" target="_blank">multitimbral</a>. <br />
<br />
There's also a virtual analog synth plug-in available, and a driver that allows the Evo to interface to <a href="http://electronicmusic.wikia.com/wiki/Max/MSP" target="_blank">Max/MSP</a>. The VA synth is pretty conventional, other than having specific controls for the Evo sensors. The software is available for OSX and Windows; not sure which versions.<br />
<br />
The question that has to be asked at this point is: how is this an improvement over <a href="http://electronicmusic.wikia.com/wiki/Polyphonic_aftertouch" target="_blank">polyphonic aftertouch</a>? That's particularly relevant since CME just introduced <a href="http://www.sonicstate.com/news/2013/01/26/wnamm13-cme-mini-usb-keyboard-with-poly-aftertouch-video/" target="_blank">a controller with poly aftertouch that lists for $99 USD</a>. Yes, it has mini keys with limited travel, and they probably feel like cheap plastic. But at that price, you can afford to buy one and only use it for parts where you need poly aftertouch. And some good MIDI remapping software will let you do a lot of the tricks that the EVO MIDI interface software does (although probably not the mode 3 trick). And many modern synths, both hardware and software, will accept and process poly aftertouch messages even if they can't generate them. So what does the Evo offer over that? Well, for one thing, many players who have worked with poly aftertouch have found it difficult to actually control in the heat of the moment. The extension of fingers over the Evo sensors should be easier (thumbs, not so much, but...). Second, the Evo sensors can be, as noted, used without actually playing the key in question, which provides a lot of additional flexibility. And third, the Evo allows the sensors to be used in the "first touch" mode a la the ribbon controller on the <a href="http://electronicmusic.wikia.com/wiki/CS80" target="_blank">CS80</a>, where the place where the finger originally touches the sensor sets the zero point for subsequent movement. Can't do that with poly aftertouch.<br />
<br />
Endeavour sells two versions of the Evo, a two-octave (24 keys, C to B) and a four-octave, 48-key version. The lack of a high C is probably because a conventional high C requires a unique key shape, which would be a significant additional cost on this keyboard. There are a few questions not answered on the Web site, such as how the unit is powered: it doesn't say if it comes with its own power supply, or if it expects power over Ethernet. It does say that the Ethernet should be self-configuring. I think that it relies on the unit being plugged in to a network with a router or something else that will act as a DHCP server, so that the Evo can obtain an IP address automatically. (There is a way to set it manually if needed.) The only manual I could find was in German, which didn't help me much; the Web site states that the manuals are being translated, but the availability date it gives was last November. It doesn't do wireless (which would have been an interesting addition, especially for live performance), so you will need an Ethernet router or hub with an available RJ45 jack. <br />
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So this is a new take on per-note parameter control, and a pretty interesting one. There is a video demo on Evolution's Web site, but in my opinion it doesn't do a very good job of showing the Evo's unique capabilities. The Evo isn't cheap but the cost is actually not bad compared to a lot of other alternative controllers that I've come across. Is it worth the price? You decide. The two-octave version of the Evo markets for E499 and the four-octave version for E999; that's about $650 and $1300 USD, respectively. Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-41626871547351353152013-02-05T23:11:00.000-06:002013-02-05T23:12:17.164-06:00SSL Double Deka -- full review<br />
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Geez, only eight posts last year? I've got to do better than that. I'll kick off 2013 with a full review of the <a href="http://www.steamsynth.com/m_DoubleDeka.aspx" target="_blank">SSL Model 1130 Double Deka VCO</a> that I previewed last year. Now that I've finally had some time to experiment with it, I've got some audio clips to share. And a photo of the thing, finally powered up:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEio6OMCzR_Ic-Xw10231PFqq7TsJnpdhpSp_9fIQbwTCVqX9f3k-Uz8qG8tl0PwE3IaVt2erD6WvEseOrPrcJsvFgH78doCDe5wukh4e4hk5wk6C_1x7YPVssiXTDcuZfifAJWaszkczgk/s1600/IMG_6948.jpg" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEio6OMCzR_Ic-Xw10231PFqq7TsJnpdhpSp_9fIQbwTCVqX9f3k-Uz8qG8tl0PwE3IaVt2erD6WvEseOrPrcJsvFgH78doCDe5wukh4e4hk5wk6C_1x7YPVssiXTDcuZfifAJWaszkczgk/s320/IMG_6948.jpg" width="240" /></a></div>
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First of all: I mis-stated and grossly over-complicated the theory of operation in my preview article. Here's what the Double Deka basically does: Use a bank of sliders to draw a waveform, and the Double Deka reproduces that waveform, more or less. That's it. How it actually does it is rather complex -- there's a VCO that runs at ultrasonic frequencies, and it steps from one slider to the next on each cycle. By doing so, it re-creates a waveform that looks on the scope like the shape that you created with the sliders. </div>
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Now, some background: The Double Deka is a design created by renown module designer <a href="http://home.comcast.net/~ijfritz/index.htm" target="_blank">Ian Fritz</a>. The 1130 is a further development of the basic design, which adds some new features and also re-formats the module into the <a href="http://electronicmusic.wikia.com/wiki/Dotcom" target="_blank">Dotcom </a>module format. (A photo of the older version can be seen on the Bridechamber web site <a href="http://www.bridechamber.com/F_DoubleDeka.html" target="_blank">here</a>.) My 1130 is from SSL's second production run, from late 2012; this run fixes a circuit bug that the original run had which prevented the sync input from working properly. The 1130 mounts in a standard Dotcom-format case and has the standard Dotcom 6-pin MTA-100 power connector. It's a large module, 4U wide. </div>
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<b>Creating Waveforms with the Double Deka </b></div>
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The first thing everyone notices about the 1130 is the cool lighted sliders. The two banks of green sliders generate two waveforms, each driven by the master VCO. The orange sliders select which octave each bank operates in, using a deviously clever method that will be explained later. You create a waveform by "drawing" it with the sliders. As a simple example, here is a sawtooth wave set up in the A bank of sliders (the B bank is not used in this example):</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjGswQvQ1d1_FZMA7M-ryVcGzPz0D5mRZ7uGpnT79Hne99-V5vuprCV_ta5sw7zF61Me6kfu7Or3kWYEUi5k2BRg9sH76OYua2hioDVSfDAzor7ZSXa_6zj5Zb61cGqr6UCvUtMd41u3zk/s1600/IMG_6950.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em; text-align: center;"><img border="0" height="240" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjGswQvQ1d1_FZMA7M-ryVcGzPz0D5mRZ7uGpnT79Hne99-V5vuprCV_ta5sw7zF61Me6kfu7Or3kWYEUi5k2BRg9sH76OYua2hioDVSfDAzor7ZSXa_6zj5Zb61cGqr6UCvUtMd41u3zk/s320/IMG_6950.jpg" width="320" /></a></div>
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And here is the resulting waveform, as captured using the oscilloscope tool in <a href="http://www.motu.com/" target="_blank">MOTU </a>CueMix:</div>
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Similarly, here are the sliders set for a narrow pulse wave:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhunizh3gkaq9i7l9GNboyvHxVNtJgLd4Xq3lNM67F6z8N3nceY7xhTGR_rYpf3zl4X-vbKlMmHE_3OSpoRcXFeF1E6z2iQ1ER-8GhezNDIL5GgHt0txHQsT6HPJAHDvlM2sIpHdIdD6CQ/s1600/IMG_6952.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em; text-align: center;"><img border="0" height="240" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhunizh3gkaq9i7l9GNboyvHxVNtJgLd4Xq3lNM67F6z8N3nceY7xhTGR_rYpf3zl4X-vbKlMmHE_3OSpoRcXFeF1E6z2iQ1ER-8GhezNDIL5GgHt0txHQsT6HPJAHDvlM2sIpHdIdD6CQ/s320/IMG_6952.jpg" width="320" /></a></div>
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And here is the resulting waveform:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgAoOdm1iY0PzU5IIeMP5g65Qv8lVdpZ6ch98YbPYSzkp4NFqyEFaFmrAkTym4Jtz85fP9IRHwVR_1At9OC5X5ryLCC_bu2r-FrpLXOVTQnOZi1ohgD-DKcyzmFNgflI64QsIunVRF7iWw/s1600/Pulse.png" style="clear: left; float: left; margin-bottom: 1em; margin-right: 1em;"><img border="0" height="168" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgAoOdm1iY0PzU5IIeMP5g65Qv8lVdpZ6ch98YbPYSzkp4NFqyEFaFmrAkTym4Jtz85fP9IRHwVR_1At9OC5X5ryLCC_bu2r-FrpLXOVTQnOZi1ohgD-DKcyzmFNgflI64QsIunVRF7iWw/s320/Pulse.png" width="320" /></a></div>
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You'll notice that there is some sagging present in the waveform. The 1130 has a switch that can be used to select AC or DC output; when the switch is in the AC position, DC is filtered out. As it happens, the tops and bottoms of a square or pulse waveform look like DC to the filter; hence the tendency for the waveform to drift towards the axis when it should, in theory, be perfectly horizontal. I did try capturing some waveforms with the switch set to DC, and I still saw sagging, although not as much. However, I suspect that my recording chain (MOTM-890, Mackie mixer, MOTU 828 Mk III Hybrid) is not DC-continuous, so that could account for that. I have not tried putting an actual scope on the output -- I might do that next.</div>
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Here's a triangle wave, with the sliders:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjTAl3pA39T8y5yEVHpW2EkYpFHko3itwjG4zBF_5KAfF7qE7K_ZEBNtXgUXUeAe7Yx96ZItEtg06W1yKGjM1ftdOPQKfvteAZl591L1xVwkGAG7Accscm8n6rXFnJ9k1Hw39NYcqeyV2g/s1600/IMG_6951.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em; text-align: center;"><img border="0" height="240" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjTAl3pA39T8y5yEVHpW2EkYpFHko3itwjG4zBF_5KAfF7qE7K_ZEBNtXgUXUeAe7Yx96ZItEtg06W1yKGjM1ftdOPQKfvteAZl591L1xVwkGAG7Accscm8n6rXFnJ9k1Hw39NYcqeyV2g/s320/IMG_6951.jpg" width="320" /></a></div>
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And the resulting waveform:</div>
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Notice that in this one, and the sawtooth wave from above, the stair-step nature of the waveforms. Obviously, ten sliders is nowhere near enough to do a high-resolution waveform. The Double Deka isn't intended to be a wavetable oscillator; it's intended to be a device that allows experimenting with waveforms and creating a variety of waveforms, from a somewhat restricted set, on the fly. The stair-stepping does have some implications for the sound, which I'll talk about in a bit. </div>
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And finally, we have a waveform resulting from some random slider settings. I didn't photograph the panel for this one, but you get the idea from the resulting waveform:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhW-OOyX0XPFsi_VD5EcD63MyybXtqfnT_p_trHUbIJJYUgdoK0VISo2aYLzj5QP0fTUO7XVFjWcUk4ASqkBpYs_ySwLjU_xyXhC9wdTgtMjwmBQ492Zo9OMH6p0U_jZgjUJfV9RJUwmbY/s1600/Null+Waveform.png" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="168" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhW-OOyX0XPFsi_VD5EcD63MyybXtqfnT_p_trHUbIJJYUgdoK0VISo2aYLzj5QP0fTUO7XVFjWcUk4ASqkBpYs_ySwLjU_xyXhC9wdTgtMjwmBQ492Zo9OMH6p0U_jZgjUJfV9RJUwmbY/s320/Null+Waveform.png" width="320" /></a></div>
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So what do these waveforms sound like? Here is an audio clip, containing five waveforms in this order: square, sawtooth, triangle, pulse, and random slider settings. I didn't try to record a sine wave because I couldn't set the sliders precisely enough to come up with anything that sounded remotely like a sine.</div>
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<a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/DD%20Raw%20Waveforms.mp3" target="_blank">Waveforms</a></div>
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You'll notice that there's a certain tonal flavor that is common to all of the waveforms. That, I think, is due to the stair-stepping between slider values; it tends to impose an odd-harmonic structure on all of the waveforms that are output. I was able to get rid of that tone by running the output through a lag processor. Linear slew limiting might be even better, but it's hard to do in analog circuitry; I'll have to play with coding it up in Csound and running the 1130's output through it. I thought at first that a built-in lag processor, or lowpass filter that could be offset according to the frequency, might have been a useful addition. Then again, there's already a ton of circuitry crammed into this module, and not much room for any more jacks or controls. And there is such a huge variety of lowpass filters available on the market that I realized it would not make sense to incorporate one into the DoubleDeka itself. Instead, leaving that choice to the performer makes sense.</div>
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I've found that there seems to be a bit of the unexpected in how the sliders interact with the processor. I'm not sure what it actually is. The first thing I noticed: if you set all ten sliders to the zero value, you would expect to get silence. Instead, there is a low-level buzzing noise. I wanted to see what this looked like on the scope display, so I cranked the gain way up at my mixer, and got this:</div>
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It's a digital-type noise that seems to be impossible to totally get rid of. Where does that come from? Is it something to do with the calibration of the sliders? I don't know. Another thing I noticed, possibly related to this, is that all of the sliders don't seem to behave in the exact same way. For example, on the A bank, if you set all the sliders to zero and then set the first (leftmost) slider in the bank all the way up, you get what you'd expect -- the sound of a narrow pulse wave. However, if you zero all the sliders and then turn the second slider way up, there's something of a difference: a prominent third harmonic that doesn't seem to show up on the scope, but you can hear it. Other sliders in the bank did one or the other to varying degrees. In the B bank, the same thing happens, but on different sliders. </div>
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<b>Exploring the Panel</b> </div>
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Anyway, let's move on to explore some of the other features of the 1130. And there are a lot of features here. Getting past the obvious: At the upper left are knobs for coarse and fine tune. These control the base frequency of the VCO and hence the frequency, modulo octave settings, of both banks of sliders. (It would have been cool if there were a way to offset the frequency of the B bank from the A bank, but that would have required a second VCO.) If you move the coarse tune knob slowly, it steps audibly; I'd say about 1/8 tone (25 cents) per step. This is the price to be paid for having a coarse tune control with a very wide range. The range of the fine tune control is about eight half steps. </div>
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The two slider banks each have an individual output, and then there is also an output for the mix of the two. There is a knob for controlling the mix, and also a control voltage (CV) input. With the knob full counterclockwise and no CV connected, the output is A only; at full clockwise on the knob the output is B only; in between the two are mixed. (Minor quibble: This knob should have been indexed on the panel with values ranging from -5 to +5, or some such, rather than 0-10 which is a bit confusing for a cross-mix control.) A positive voltage into the CV jack moves the mix more towards the B bank. This is one of the features added in this new version of the Double Deka that was not present in Ian Fritz's original design.</div>
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Here is a sound clip that illustrates using the mix CV to vary the mix between the A and B slider banks:</div>
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<a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/DD%20Wave%20Blending.mp3" target="_blank">Wave Blending</a></div>
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<b>Octave Selection and Control</b></div>
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Each bank has a slider that allows selection of one of six octaves. Twiddling the octave sliders, one notices right away that they are not slide switches but linear potentiometers. There is a reason for that: each bank also has a CV input for octave selection. The CV is added to the slider setting and then the sum is quantized to determine what octave the bank should play in. If you input a signal from an LFO into the CV input, and then move the slider up and down, you find that the pattern of octave changes varies continuously as you move the slider. That's why it isn't a slide switch: by careful setting of the slider, you can get sequences of octave changes that would not be achievable if the octave slider only had discrete values. This is an addition to the original DoubleDeka design, and a very nice design feature. Feeding low-audio-rate waveforms into the CV inputs can produce quite startling results. Here is an example:</div>
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<a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/DD%20Octave%20VC.mp3" target="_blank">Octave Control by Control Voltage</a></div>
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<b>Modulation Options</b></div>
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This is an area where the 1130 really stands out. The available modulation inputs are:</div>
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<li>1 volt/octave, standard scaling for playing the module in scale from a keyboard or a MIDI/CV interface.</li>
<li>Exponential FM, with a panel knob for controlling the ratio.</li>
<li>Linear FM, with a panel knob for controlling the ratio.</li>
<li>Sync input</li>
<li>"Ring" modulation input</li>
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The 1V/octave, expo FM, and linear FM inputs all do what you expect. The available FM modulation index values are very high if you crank the knobs up. The result tends to trend towards noise very quickly; I suspect this is another aspect of the stair-stepping in the output. It's a harmonically rich output and applying FM to it creates harmonic chaos pretty quickly. There is a a switch for the linear FM input that allows a DC-blocking filter to be switched in or out. A linear FM example:<br />
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<a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/DD%20Linear%20FM.mp3" style="text-align: left;" target="_blank">Linear FM</a><br />
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The "ring" modulation input is kind of strange, but also capable of doing some really neat things. It isn't really a conventional ring modulation circuit. Here's what it does: It "squares up" whatever input waveform is applied to the "ring" input jack, so the result is either "high" or "low". When the result is "high", the output is cut off. When the result is "low", the output passes through normally. A consequence of this is that a sine wave, triangle wave, and sawtooth wave of the same frequency will have pretty much the same effect. The most interesting results occur with pulse and multi-pulse waves at the ring input. I created a patch with a Synthesizers.com Q106 VCO feeding the ring input from its pulse wave output, and an LFO driving its pulse width modulation across nearly the full range. At one extreme, the output almost disappears, but at the other extreme you hear an un-modulated output for a moment. Ian calls this "digital ring modulation". Here is a rather long example of the ring modulation, starting with a low-frequency modulation and then moving into audio frequencies. Note how the modulation starts by simply cutting the audio in and out, and then begins to resemble more typical ring modulation as the modulation frequency increases:<br />
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<a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/DD%20Ring%20Mod.mp3" style="text-align: left;" target="_blank">Ring Modulation</a><br />
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The sync I haven't yet figured out. There are two modes, a "harmonic" mode and an "aharmonic" mode. There are some demos on Ian's web site, which I haven't yet managed to reproduce. I'll post something about it later. <br />
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<b>Access to the VCO Core</b><br />
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The "HF Out" jack provides you with the raw waveform from the ultrasonic VCO core. You could use this to, for example, feed an octave divider and generate additional waveforms that could be used to generate audio signals to be mixed with the DoubleDeka's outputs, or to do weird sync tricks. (Don't run it directly to your amp -- it'll fry your speakers!) It would have been nice if there were also an input that overrode the VCO core's signal to the A and B divider banks. That would, for example, allow a second DoubleDeka to be slaved to the first, possible at some odd division of the first unit. <br />
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<b>Design and Contstruction</b><br />
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The 1130 appears to be well constructed. The panel is of the same mechanical design and same thickness of aluminum that Synthesizers.com uses. There are three circuit boards: one that carries all of the jacks, one that carries all of the sliders, and one that carries the VCO core and other circuitry. The slider board has some surface-mount components on it; I didn't look closely to see what they are, but they are probably multiplexors for stepping through the sliders. The VCO is <a href="http://electronicmusic.wikia.com/wiki/Tempco" target="_blank">tempco</a> regulated, and trimmers for adjusting scale of the 1V/octave input, and high frequency compensation, are available. I have so far not needed to touch the calibration. The rotary pots are from Alpha and feel very nicely damped. And I didn't have to re-index any of the knobs; they were all spot on. Max depth from the back of the panel is 2.75", or slightly less than 7 cm. <br />
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The panel is well laid out despite being crammed with stuff. I think this module wins the prize for most items on the panel -- 22 sliders, 5 pots, three toggle switches, and 12 jacks. The area around the mix knob could get busy if you are using all of the A and B octave CV inputs and the mix CV input. The A octave slider is a bit close to the coarse tune knob -- I bumped it a few times while tweaking that slider. But these are all minor quibbles. You might not think it's possible for a 4U width module's panel to be "full up", but this one is, and SSL did a fine job of laying out the panel so as to keep everything comfortable. The panel graphics are the same as what Synthesizers.com uses, and the silkscreening is high quality. <br />
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<b>Conclusions</b><br />
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You may have some doubts about whether you want to dedicate 4U of precious case space to one module. In this case -- do it. Everyone needs one of these; it's a great route to exploring and finding new timbres. Use the modulations, apply generous amounts of low pass filtering, and will be pleased by the results. I leave you with one more sound sample. This is a pulse wave from a Synthesizers.com Q106 VCO being fed to the Double Deka's ring mod input. The Q106 is being pulse-width-modulated by an asymmetrical sine wave from a Synth Tech MOTM-320 LFO. The range of the LFO is such that at one end it drives the Q106's pulse width completely into cutoff, and at the end of each sweep the un-modulated waveform peeks out for a moment.<br />
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<a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/DD%20Chaos%20Excerpt.mp3" style="text-align: left;" target="_blank">Noisy</a><br />
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And visit <a href="http://home.comcast.net/~ijfritz/projects/DD.htm" target="_blank">Ian Fritz's Double Deka page </a>for more demos. </div>
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-61492012410026153402012-11-28T22:54:00.000-06:002012-11-28T22:54:46.377-06:00SSL Double Deka VCO -- first look<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgPDWFZ9Wn4yqePjpHVYqlInNlX-4nkF3oL4vn8uqstGU1kWmuWUuDTDjRtnE9gGDxopbxPxvPk3fQwxE8lYsU6bjcinYDBPQzxCbKmW5CFu-dg-__cGEQO5aat8uwItID3kaQn3hb0klA/s1600/IMG_6931.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgPDWFZ9Wn4yqePjpHVYqlInNlX-4nkF3oL4vn8uqstGU1kWmuWUuDTDjRtnE9gGDxopbxPxvPk3fQwxE8lYsU6bjcinYDBPQzxCbKmW5CFu-dg-__cGEQO5aat8uwItID3kaQn3hb0klA/s1600/IMG_6931.jpg" /></a></div>
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This beast is the <a href="http://www.steamsynth.com/" target="_blank">Synthetic Sound Labs</a> Double Deka VCO, based on the original design by <a href="http://home.comcast.net/~ijfritz/" target="_blank">Ian Fritz</a>. SSL just completed a limited run, which was marketed through <a href="http://www.muffwiggler.com/forum" target="_blank">Muff's</a>. The module, as you can see, is a three-unit-wide module in <a href="http://electronicmusic.wikia.com/wiki/Dotcom" target="_blank">Dotcom</a> (aka MU) format. It takes power through a standard Dotcom 6-pin MTA-100 connector. </div>
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To be honest, I'm not yet fully up on how this beast actually works. As I understand it, the VCO proper runs in the ultrasonic range (around 100 KHz, I think), and it divides that down to produce square waves of varying ratios, which can be mixed in inverted or non-inverted via the banks of sliders that feature so prominently on the panel. There are actually two, but they share a set of frequency control voltage inputs -- 1V/octave, variable-ratio exponential, and linear inputs are all available. There is an octave selector for each bank, and a mix control for mixing the two banks. There are a bunch of other control voltage inputs that I'll detail in a review after I've had a bit more time with it.</div>
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Before it's installed, let's look at the back:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjrbch_dSjjw8VO5ECKKU04FY7ZuRc0TD8zGPQwXJlN5Uq8ts5lrIsAVWFMcGqvO6BkaiAk-IS2yglKxL3A1oe_ifkSx2ESpKcleMKFOxfil_2ccqG1whGS6kbU8xHZ3eQzO4G4QGwuzC4/s1600/IMG_6933.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjrbch_dSjjw8VO5ECKKU04FY7ZuRc0TD8zGPQwXJlN5Uq8ts5lrIsAVWFMcGqvO6BkaiAk-IS2yglKxL3A1oe_ifkSx2ESpKcleMKFOxfil_2ccqG1whGS6kbU8xHZ3eQzO4G4QGwuzC4/s1600/IMG_6933.jpg" /></a></div>
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We see three boards: the large board taking up most of the surface area of the panel appears to be the "main" board; it has some surface mount, and most of the panel controls are mounted to it. The long board at the bottom mainly carries I/O jacks. The extension board sticking out from the right edge is show in a better view below:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjTgwwc82BEtAI6ekvmBHTlQysCihjX3OQ5sxT5vig2HjBFwWJV37LdQFt21X6ZtaRVfvH7LuxGh9EDaH7UwsikaMlx6T4SdHIek-7qrbAhNF4FJMAVpat3jASVXO_FJoh2v1X61IV0DQs/s1600/IMG_6934.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjTgwwc82BEtAI6ekvmBHTlQysCihjX3OQ5sxT5vig2HjBFwWJV37LdQFt21X6ZtaRVfvH7LuxGh9EDaH7UwsikaMlx6T4SdHIek-7qrbAhNF4FJMAVpat3jASVXO_FJoh2v1X61IV0DQs/s1600/IMG_6934.jpg" /></a></div>
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Interestingly, this board appears to be carrying the actual VCO circuitry, since the tempco mounted to the expo-converting transistor array is obvious if you look closely (under the blue trim pot). The power input connector is also here; you can see it as the white component at the left edge in this picture. That location caused me a slight bit of trouble during installation since it's near the top edge, and I had to bend the power cable some to keep the top cover from pressing on it. </div>
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So how does it sound? Well, I've only had about 20 minutes so far... It's a VCO that has a lot of capability for generating various timbres. Adjusting the sliders has so far been a trial-and-error process; I'm still trying to find the design documentation (which doesn't seem to be present on Ian's Web site at the moment). So far what I've managed to do with random tweaking has been mostly pulse-sounding things, although I have hit a few times on combinations that had the effect of greatly emphasizing certain harmonics (particular the 2nd and 3rd). The fact that you have two banks, and that the mix between them is voltage controllable, creates possibilities for morphing between three timbres: the A bank, the B bank, and the timbre that results when the two are mixed. (You can get a lot of cancellation between them, to the point of the output almost disappearing with certain combinations.)</div>
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There are modes for syncing to an external oscillator, which I don't quite understand yet; the terminology used on the switch is not the standard hard/soft selections. There are control voltage inputs for making the banks change octaves, and a "ring in" jack that I haven't quite figured out -- it doesn't seem to be ordinary ring modulation. I'm thinking what it might do is square up the input signal and then XOR it against the output. </div>
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I think it's going to be a good module to have and to work with. By itself, it doesn't zingy outer-space sounds, although I did manage to get a few weird things with external modulation. There's undoubtedly a lot more timbral capability in the slider banks than I've uncovered so far. And SSL's service was great; it took a few months to build this batch, but once they were ready and I paid my balance, I received my module within a week. One minor complaint: no power cable was included. Not a big deal to me since I have the tools to make my own. Packaging was very good and my unit arrived in fine shape. </div>
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And yeah, there are those cool LEDs in the sliders. </div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgqbiaOv2YzWQO1gCjBPz7WYSvThuBLS9hyphenhyphen32H1pOAI79JBuxwGhxIf3wTCWpuo_ECn6aJIaYehlxJ0JD3-d4-3Ec7T4D6auLYggsuwWUxx1rFMs36IvDRRh8H5P3s899smi_zvwmNxlag/s1600/IMG_6935.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgqbiaOv2YzWQO1gCjBPz7WYSvThuBLS9hyphenhyphen32H1pOAI79JBuxwGhxIf3wTCWpuo_ECn6aJIaYehlxJ0JD3-d4-3Ec7T4D6auLYggsuwWUxx1rFMs36IvDRRh8H5P3s899smi_zvwmNxlag/s1600/IMG_6935.jpg" /></a></div>
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-90862270363212776662012-10-13T19:58:00.000-05:002012-10-13T19:58:15.436-05:00Discombobulator changes<div class="separator" style="clear: both; text-align: left;">
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Several months ago I purchases a <a href="http://electro-music.com/forum/forum-185.html" target="_blank">Rob Hordijk Designs</a> Phaser Filter. At the time, I didn't have a place to put it. Last week, I did a rearrangement of the Discombobulator, and I removed a few modules that I haven't been using much, and a couple that weren't working right. That left me space to install the Phaser Filter. But, when I went to connect the power cable (it can take the <a href="http://electronicmusic.wikia.com/wiki/MOTM" target="_blank">MOTM</a>-format or <a href="http://electronicmusic.wikia.com/wiki/Dotcom" target="_blank">Dotcom</a>-format power inputs, I found a small problem: The module uses a "sandwich" of two circuit boards, and the space between isn't quite large enough to get the female connector in over the ends of the male connector's pins:</div>
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<img border="0" height="240" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiHO27iOtl6fzNIDmX59nXsdutz95GGQwWvLVpry0QZ4QQ10zrAbJ7T8frDJq6yxI_848Yis_NnvLYzC1j5aF9zyyvGlwXlpGIYwQXZ0ydnMZrGmFz-QoaGxmoRMC13RSnL9dHWBbGjvuU/s320/IMG_6884.jpg" width="320" /></div>
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The fix was pretty easy, though: remove the two screws holding the rear board to the standoffs at that end, and loosen the two screws at the other end. That lets the rear board move away from the front board enough to get the connector in:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgBFQzJq2vFw3hctF7wHR-9T7aNSOb9c7OVnC5ViPk4_Oq8i1vz-zhjy-OjOx3zSD4eGQ_Qx0-bKlSUZ0cJ917g5kCqAYO4lm-_2tHXKqZz7myrYSoDsVGyP8IDLBKYY4HamPDbFcewJkg/s1600/IMG_6885.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="240" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgBFQzJq2vFw3hctF7wHR-9T7aNSOb9c7OVnC5ViPk4_Oq8i1vz-zhjy-OjOx3zSD4eGQ_Qx0-bKlSUZ0cJ917g5kCqAYO4lm-_2tHXKqZz7myrYSoDsVGyP8IDLBKYY4HamPDbFcewJkg/s320/IMG_6885.jpg" width="320" /></a></div>
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The Phaser Filter is a combination of a 5-stage phaser in series with a 3-stage lowpass filter. It's rather unusual in that, if no modulation signal is plugged into the mod in jacks for each function, the modulation defaults to the incoming audio signal. I'll have a review with a few audio examples up soon. Here's the module, ready for installation:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEis4lYEk32pZQLjSfXtDhRv7zlqGfZSj-2vDhiZnT7DCbwqKr2L0GZOVa8LaYwa2aVMl3cvgSbG_1TkmrPVr3R6tuaSEPyXbr4905EEmLFvfh86ssM8Un5GrbjutruFPz7Ch3IVysHLyfU/s1600/IMG_6888.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEis4lYEk32pZQLjSfXtDhRv7zlqGfZSj-2vDhiZnT7DCbwqKr2L0GZOVa8LaYwa2aVMl3cvgSbG_1TkmrPVr3R6tuaSEPyXbr4905EEmLFvfh86ssM8Un5GrbjutruFPz7Ch3IVysHLyfU/s320/IMG_6888.jpg" width="240" /></a></div>
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Here are the five blocks of the Discombobulator, brought together on top of the Hammond A100:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiUcKgn-ylGfDKY8fid2RPs0dDpctpVlsnEBEz2akFzEFSAiO3SHsoxmooP1p1ZT0eaNUBH2q7aDdAeAdzynT8KL9oMVpKxgtAPw9Bdk1uVY7D8Zj76I7fayNEDwycA5wnNVDQQQpQ6U4s/s1600/IMG_6893+19-55-33.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="240" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiUcKgn-ylGfDKY8fid2RPs0dDpctpVlsnEBEz2akFzEFSAiO3SHsoxmooP1p1ZT0eaNUBH2q7aDdAeAdzynT8KL9oMVpKxgtAPw9Bdk1uVY7D8Zj76I7fayNEDwycA5wnNVDQQQpQ6U4s/s320/IMG_6893+19-55-33.jpg" width="320" /></a></div>
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The current composition:</div>
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<b>Tethys </b>(upper left):</div>
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1. Synth Tech MOTM-310 micro VCO</div>
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2. Synth Tech MOTM-310 micro VCO</div>
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3. Synth Tech MOTM-310 micro VCO</div>
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4-5. Rob Hordijk Phaser Filter</div>
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6-7. Encore Electronics Frequency Shifter</div>
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8. open</div>
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9. Synth Tech MOTM-890 micro mixer</div>
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<b>Iapetus </b>(lower left):</div>
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1-2. Synth Tech MOTM-650 four-channel MIDI interface</div>
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3. Oakley EFG envelope follower / gate generator</div>
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4-5. Synth Tech MOTM-820 voltage controlled lag processor</div>
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6. Synthesizers.com Q128 A-B routing switch</div>
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7-8. Encore Electronics Universal Event Generator</div>
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9. Synthesizers.com Q108 VCA</div>
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<b>Dione </b>(upper right):</div>
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1. Synthesizers.com Q130 clipper / rectifier</div>
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2-3. Synth Tech MOTM-410 Triple Resonant Filter (modified)</div>
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4-5. Synth Tech MOTM-510 WaveWarper</div>
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6. Cynthia Steiner Filter (jacks converted to 1/4")</div>
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7-8. Synth Tech MOTM-440 OTA lowpass filter</div>
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9. Synth Tech MOTM-890 micro mixer</div>
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<b>Titan </b>(middle right): </div>
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1. Synthesizers.com Q141 oscillator aid</div>
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2-3. Synthesizers.com Q106 VCO</div>
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4. open</div>
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5. Synth Tech MOTM-190 VCA / ring modulator</div>
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6. Synthesizers.com Q109 ADSR envelope generator</div>
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7. Synthesizers.com Q123 voltage / frequency standards</div>
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<b>Rhea </b>(bottom right):</div>
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1-2. Synthesizers.com Q106 VCO</div>
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3. Synthesizers.com Q161 oscillator mixer</div>
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4-5. Synth Tech MOTM-320- voltage controlled LFO</div>
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6-7. Synth Tech MOTM-101 noise source / sample and hold</div>
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8-9. open</div>
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Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-9506728850860153262012-09-17T22:20:00.002-05:002012-09-17T22:20:31.539-05:00First attempt at panel fabricationI decided to take a crack at fabricating a module panel on my own. I've got a couple of spaces in the <a href="http://home.hiwaay.net/~cornutt/Music/Web%20Page/modular.html" target="_blank">Discombulator</a> that are not quite wide enough for a module in any current 5U format. So I decided I'd make a <a href="http://electronicmusic.wikia.com/wiki/Multiple" target="_blank">multiple</a> that will be a one-half-<a href="http://electronicmusic.wikia.com/wiki/MOTM" target="_blank">MOTM</a>-unit width. "How hard can it be?", I told myself.<br />
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So I ordered some sheet aluminum from <a href="http://onlinemetals.com/">onlinemetals.com</a>. This place is nice because the sell sheet, bar, tube, etc. metal goods in retail quantities. I ordered a 12x24 inch sheet of .0125" 6061 aluminum. While I was waiting on it, I made a full-size template of the panel, with the locations for the jacks and mounting holes, as well as a switch. When the metal came in, I used this to mark the length and width of the piece that I wanted. I bought a metal cutting blade for my circular saw, and had it. <br />
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The results, as you can see, were not terribly satisfactory:<br />
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgeOgwn2pMT44KvfNqdKwRr8GjO3QrxTqR5-6Ws6MuFjOTHr67lE8COuhDnZTz6HFjZDLTptE3UG6gUpUPSontHnhXsqe5wMA3wWpKT9ZKB4kgu0KYf09prRpmwnnLCy9Y2NeaNMlpzKY8/s1600/IMG_6587.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgeOgwn2pMT44KvfNqdKwRr8GjO3QrxTqR5-6Ws6MuFjOTHr67lE8COuhDnZTz6HFjZDLTptE3UG6gUpUPSontHnhXsqe5wMA3wWpKT9ZKB4kgu0KYf09prRpmwnnLCy9Y2NeaNMlpzKY8/s320/IMG_6587.jpg" width="240" /></a></div>
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The printed template is show here next to cut metal. As you can see, the cut metal isn't a consistent width; it's wider in the middle. I found it difficult to cut a straight line with the circular saw because I had to go so slowly, and because the guide on my saw isn't that good. The burrs were also really bad -- it was like the saw melted some of the edge and it built up an irregular edge on both sides. I worked on it with a Dremel armed with a diamond-grinding bit, and that got a lot of that cleaned up, but eventually melted metal clogged up the bit. Nonetheless, I think I'll proceed with drilling and assembly and see how it goes. I haven't decided yet what to do for a finish. </div>
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Here is the template overlaying the metal:</div>
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<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgURTENZtUxALM-XIaqXyN7-Q921ZRMNTGQLKSK4reTpcyp8kIRjMqcwJBw2-XpYX7F-Atf59W5iAemfHfycixysMIcOR4xRvjWDKDU56CRMAkMGRbPAS3d5HBnheU8wb2_1Gd2UFaaWbw/s1600/IMG_6588.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgURTENZtUxALM-XIaqXyN7-Q921ZRMNTGQLKSK4reTpcyp8kIRjMqcwJBw2-XpYX7F-Atf59W5iAemfHfycixysMIcOR4xRvjWDKDU56CRMAkMGRbPAS3d5HBnheU8wb2_1Gd2UFaaWbw/s320/IMG_6588.jpg" width="240" /></a></div>
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-27571110844711151712012-09-11T22:47:00.000-05:002012-09-11T22:47:11.625-05:00What is electronic music?We pause today to ponder the question: what, exactly, is electronic music? What does it mean for music to be "electronic"? We ask this question in part just to be smart alecks, but also because it appears that electronic music is experiencing one of its <a href="http://www.vintagesynth.com/forum/viewtopic.php?f=6&t=68942" target="_blank">periodic brief bursts of fashionable-ness</a>. (Hat tip to "meatballfulton" at VSE.) Such are always fraught with potential danger: loss of creativity, hyper-inflation of gear prices, and annihilation of the universe by previously-unsuspected interactions between the electron neutrino and pretentious hipsters.<br />
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Back in the simpler days of the '60s and '70s, the answer to the question was easy: if electronics were employed in the creation of the various tones, the music was electronic music. This was because the early synthesizers, and the various pre-synthesizer electronic devices, were not designed to re-create familiar instrument sounds. In fact, they were designed (or improvised) to do the opposite: the ethos of the day was finding a way to create new sounds and timbres, not reproduced old, tired ones. (Okay, that last bit may have had some of its own age's hipsterism in it. Nonetheless, the dedication of early pioneers like Cage, Buchla, and the <a href="http://electronicmusic.wikia.com/wiki/BBC_Radiophonic_Workshop" target="_blank">Radiophonic Workshop</a> gang to pushing the sonic envelope had a lot to do, indirectly, with creating the sounds of a lot of popular music today.) <br />
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Then Yamaha created the <a href="http://electronicmusic.wikia.com/wiki/DX-7" target="_blank">DX-7</a>, and whether it intended this result or not, it irreversibly changed the nature of the synthesizer market, as well as defining the not-yet-existent market for things like soft synths and music analysis software. Before the DX-7, most synth performers created their own <a href="http://electronicmusic.wikia.com/wiki/Patch" target="_blank">patches </a>and sounds; after the DX-7, these people found themselves in a distinct minority. From the first bar-band keyboard player who glommed onto the DX-7 as a replacement for the venerable Rhodes electric piano, the purpose of synthesizers, as far as 90% of the market is concerned, is not to create new sounds but to reproduce existing ones. The DX-7 begat the <a href="http://electronicmusic.wikia.com/wiki/Rompler" target="_blank">romplers </a>which begat the modern "arranger workstation", such as Roland's Fantom line and Yamaha's Motif line. Although some of these have some fairly powerful sound-designing features, they are marketed using pitches along the lines of: "Thousands of sounds! More sound banks available! Every sound you need in the studio!" So much so that the ironic comment that <a href="http://electronicmusic.wikia.com/wiki/Modular_synthesizer" target="_blank">modular synth</a> owners often ask each other is: "How is the piano sound?" The arranger workstations have made the cost of producing music much lower: they can imitate almost any non-electronic instrument, without a huge investment in guitar wood, trumpet brass, or violin catgut. They work overtime and odd hours without issues. They never get sick and none of them belong to a union. Yes, it's true, they don't have down all of the style and mannerisms of a real player playing the actual instrument, but for most purposes in today's pop music, they are good enough. And anyway, as far as more accurately reproducing all of the nuances of the imitated instruments, the software smart guys are working on it.<br />
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So this leads us to a dilemma: there is a lot of music these days for which electronic instruments were used in its production, but there is no sense that anything in the music is "electronic", except for maybe a certain sense of something being not quite right. The intent is that it not sound "electronic"; in other words, it carefully avoids that territory that the likes of Buchla and Moog wanted to explore back in the '60s, or Jarre or Fast in the '70s. The 10% of the synthesizer user market who actually use synths as synths, and not as lower-cost replacements for the <a href="http://en.wikipedia.org/wiki/The_Wrecking_Crew_%28music%29" target="_blank">Wrecking Crew</a>, are stuck: when they describe what they do as "electronic music", the first thing that comes to mind for the average music listener is the latest pop-tart album, as opposed to, say, <a href="http://en.wikipedia.org/wiki/Boards_of_Canada" target="_blank">Boards of Canada</a>. The point being that there is probably nothing in the pop-tart stuff that could not have been done without electronic instruments -- it's just cheaper to produce that way. If we can agree that this does not meet the definition of "electronic music" since it does not take advantage of any of the unique capabilities of the hardware and software, then what shall we call electronic music?<br />
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The above statement leads to a possible definition: <i>Electronic music is music that employs electronics to produce tones and timbres that could not have been produced without electronics</i>. This admittedly is a bit squishy -- does something qualify as electronic music simply because it was run through a flanger? -- but it does leave open the inclusion of music for which the original sound source was non-electronic, but it was processed sufficiently with electronics that the resulting sound is something that could not been done without electronics. A fair number of modular synth users use their modulars this way part or all of the time, using the synth to mangle things like electric guitars and horns. I think that qualifies. But it leaves open another issue, that arises mainly with the users of drum machines, and it goes like this:<br />
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There are a lot of guys who are absolute demons at designing drum patterns. And even though they may be using sampled sounds of real drums, the patterns take the sounds way beyond what a human player would be capable of, such as the familiar hyper-fast rolls in which individual drum hits are so fast that the next one actually chops the end off of the previous one. Taken to a far enough extreme, it actually becomes a tone rather than a series of individual hits. And this can be done with any sound originating from an electronic source, not just a drum machine. The patterns can be replicated and combined as much as desired, with timing as accurate or as inaccurate as the performer desires. This then leads to a second definition of electronic music: <i>Electronic music is music that employs electronics to produce patterns and sequences of sounds that could not have been played by an unaided human performer</i>. Like the above, this is a bit squishy -- if you had 20 drummers playing different parts at once, could they copy a complex drum-machine pattern? Maybe they could, but getting 20 drummers and drum kits together in a studio all at once, and teaching them the pieces and expecting them to synchronize perfectly without hours and hours of practice, is pretty impractical. <br />
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There's a problem with this, though. We may have let something leak in that we didn't intend. Consider the current trends for how most popular music is recorded these days. What happens to drum tracks? They use quantizing software on the drum track to pull all of the drum hits to exact beats. Then they use something like Sound Replacer to replace all of the recorded hits with sampled ones. When the track is done, every hit is on a perfect beat boundary, and every accented or non-accented hit of a particular drum sounds exactly alike every other hit of that same drum. Levels are all perfect; toms don't ring, kick pedals don't squeak, and cymbals are never cracked. Similar for other tracks: bass is probably recorded as individual notes which are sampled and then played by a sequencer to produce the track, again with every note perfectly intonated and exactly on the beat. Even vocals aren't spared: they get Autotune so that every sung note is perfectly on pitch, and envelopes are manipulated so that every sung note is at the exact right level and of perfect duration. Take this all together, and what do you have? Music that could not have been produced without electronics!<br />
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And yet, the mind reels at classifying this as "electronic music". Why is that? Really, it's a matter of what does and doesn't become cliched due to overuse. This is an area that electronic music, and its audience, have always been very sensitive to. It's a holdover from that '60s/'70s ethos: back in the day, the audience expectation was that a performer's new album would always be more daring, more mind-blowing, and contain more unconceived-of sounds and ideas than the previous album. The idea survived into the '80s; when synth pop began, it almost by definition was fresh and unique in itself. Synth pop died when its practitioners ran out of ideas in the late '80s. Hip-hop turntabling was the same; the very idea was unique when it began. Same for the '90s first wave of electronic dance music. It's an aesthetic that demands constant improvement. In electronic music, sitting on one's laurels is not tolerated by the audience for very long. <br />
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And maybe that's the real answer: electronic music is the creation of original music and sounds, using electronics. <br />
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com2tag:blogger.com,1999:blog-279251740880432906.post-59091514405920638872012-09-04T23:13:00.000-05:002012-09-04T23:13:03.440-05:00VirusedNot my computer... me! Last winter I contracted a viral infection that caused me problems for months afterwards. It's called "pitiriasis rosea" and it gives you a skin condition like a very bad sunburn. It makes it uncomfortable to wear clothes or lay on a bed. It wasn't until March that I finally got rid of it, and then afterwards for several months I had Epstein-Barre-like symptoms. It's only been since July that I've really been myself again. That's why no posts in so long -- it was all I could do to go to work and do basic personal maintenance, and my synths went untouched for several months. But I'm getting back to it now. <br />
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As you can see, the format of Sequence 15 has changed drastically. It was time for that green to go, and the old template was not working well with the latest Blogger software and had developed a lot of formatting bugs, which required a lot of hand-editing of the HTML for every post. Plus, I never really liked how much screen real estate that template wasted. I'm still tweaking on this template, so don't be surprised to see more changes over the next few weeks. <br />
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I'm noticing that several of the modular manufacturers are asking customers to fund development of new modules, via the placing of deposits or through Kickstarter. There's been a certain amount of resistance to this; customers are leery of long delivery times and seeing their money tied up in modules that they have not received. However, the economy is bad and I understand the manufacturers not being able to take the risk of designing and producing a module that doesn't sell. The Great Modular Revival has now been ongoing for nearly 15 years; users are getting more sophisticated and are asking for more complex and capable modules that require more up-front money for design and manufacturing setup. And as this occurs, surface mount is becoming more popular in the modular world, which more or less eliminates the one guy slaving over his workbench cranking out modules via hand inserting and soldering; to go big time surface mount takes the facilities of a real assembly line with pick-and-place robots and reflow equipment. The one-man-shop can contract for that service, but it takes a minimum order and setup money up front. So let's have some patience with the designers and manufacturers who have been cranking out some great module designs. On the other hand, the days when customers could be expected to pay up front and then wait two years for their order has passed. Some of the culprits are realizing that they can't work this way anymore and are changing the way they operate, for the better. The others (and they know who they are) will probably find themselves out of business before too much longer. <br />
<br />Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-67817386071908210432012-02-10T23:31:00.000-06:002012-02-10T23:43:55.176-06:00Fizmo Project progress reportAbout two years ago, I launched something called the Fizmo Project, which was an attempt to decode the Ensoniq Fizmo's patch dump format and identify where all of the parameters are. I've posted on it here before, but after working on it for a while, I gave up in frustration because identifying individual parameters in the 3097-byte-long patch dump proved to be quite a challenge. However, that wasn't the only problem; the other issue was that none of the changes that I could see in the patch dumps seemed to correlate to the actual values of the parameters in any way that made sense. <br /><br />Well, last summer I finally devised a way to partially automate the process, using the Unix "tr" and "diff" tools in Mac OSX. With this, I can now compare two dumps and it will find all the differences and list them out for me. And the mystery started unraveling. Some info: The patch memory structure on the Fizmo is a bit unorthodox. It stores 64 "sounds", each of which consists of two layers. Each layer is basically one complete sound path, with a Transwave oscillator, a filter, a VCA, three envelope generators, an LFO, and some other modifier sources. Within a sound, the two layers can be overlaid or split across the keyboard. Each sound has its own name.<br /><br />The Fizmo also stores 64 patches, each of which consists of a combination of the 64 sounds, plus a few other parameters. A sound can appear in more than one patch. Each patch has its own name, in addition to the names of the four sounds selected in the patch. A patch also contains a global effect; it can choose one of about 20 available effects, and each effect has its own set of parameters. <br /><br />When you do a patch dump, you get a big ugly lump of 3097 bytes. There's a header area, followed by four blocks of 640 bytes, one for each sound loaded into the patch. All four sound blocks have the same layout. The parameter layout is really, really ugly. Don't listen to anybody who tells you that it is similar to the layout used by the Ensoniq MR and ZR models. Not even close. Very few parameters start or end on a byte boundary; most span across bytes, and most bytes that contain any data at all contain pieces of two parameters. There seem to be unused bits and bytes scattered about everywhere at random. Some parameters which take up seven bits go from 0-127, like you'd expect, but others only range from 0-100. The character strings which contain the patch and sound names appear to be stored in reverse order, last character first. There's a lot of dead space. I haven't yet made any effort to identify where the effects parameters are, so it's possible that a lot of the apparent dead air is taken up by them, but we'll see.<br /><br />Following is a screen shot of a file that I'm building to describe the dump layout in detail:<br /><br /><a onblur="try {parent.deselectBloggerImageGracefully();} catch(e) {}" href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhli-1TPs9XPDRenSFnQBfLsEmCMkXWP_imG_nsPAs_1GZ_7UloZfLR0bp6F-HxDn4r2zlb2MWdo2sJ9kseNH9IERqcPKIwS6nRSji680O_e233158gydEVdOcAWhYsMeBfuOCNKTUV7Cc/s1600/Fizmo+Byte+Listing.png"><img style="cursor:pointer; cursor:hand;width: 320px; height: 246px;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhli-1TPs9XPDRenSFnQBfLsEmCMkXWP_imG_nsPAs_1GZ_7UloZfLR0bp6F-HxDn4r2zlb2MWdo2sJ9kseNH9IERqcPKIwS6nRSji680O_e233158gydEVdOcAWhYsMeBfuOCNKTUV7Cc/s320/Fizmo+Byte+Listing.png" alt="" id="BLOGGER_PHOTO_ID_5707731254033880946" border="0" /></a><br /><br />This file is available on my Web site <a href="http://home.hiwaay.net/%7Ecornutt/Music/Web%20Page/FizmoByteListing.ods">here</a>. It's an Open Office spreadsheet. Open Office is support on both Windows and Mac OSX platforms; you can download it for free from openoffice.org. The row number is the number of the byte in the dump, where the sysex beginning-of-exclusive byte (F0) is byte 1. Column A lists which of the four sounds the parameter belongs to, if it is specific to a sound. Column B shows which parameter or parameters appear in the byte, and which bits. The rightmost bit is bit 0, and the leftmost bit is bit 6 (remember, only 7 bits are available in a MIDI data byte). Column C provides any additional notes, such as parameter range or bit encoding. The next seven columns with the color-coded blocks will match up (in most cases) with the colors used in the text, to show which bits in the byte each parameter is using. Looking at the screen grab above, you can see how much empty space there is, and how many parameters wrap across the byte boundaries.<br /><br />For those who didn't know, the Fizmo has a huge number of parameters which are not editable (or even viewable) from its own panel; you have to use an external editing device to see or change them. Given that there are about 200 parameters for each sound, the only practical way to do this is to use a computer editor. However, the only editor that has ever been available for the Fizmo is a bastardized version of Sounddiver that Ensoniq made available to Fizmo buyers, before Creative Labs (the company that bought out Ensoniq) shut them down. My experience has been that this version of Sounddiver is incredibly buggy; it never quite does the same thing twice, and nearly every time I run it, I have to try to figure out what set of magic incantations is required that day in order to get it to basically work. The librarian features are untrustworthy and I've trashed many of the patches on my Fizmo in the process of doing this investigation (fortunately, I didn't have anything on it that I particularly wanted to keep). <br /><br />Given all this, there is strong motivation to work towards creating an open-source editor. That's one reason I'm doing this project. The Fizmo has the potential to be a very powerful synth, but most of its power can't be unlocked until there is a reliable and supported patch editor for it. I'm maintaining <a href="http://www.vintagesynth.com/forum/viewtopic.php?f=1&t=64073&start=15&hilit=fizmo">a thread</a> concerning the project on VSE; I'd like to get some people interested in starting to write some code. <br /><br />The main things remaining to do with the patch dump are: (1) figure out how the parameter values work for the parameters that specify wave selecting and offsets into the wave table (they appear to be memory addresses, but I haven't poked at it much yet), and (2) start working out the locations of the effects parameters. After that, although it isn't an absolute necessity for a basic patch editor, I'd like to start documenting the single-parameter sysex messages that Sounddiver uses to transmit individual parameter edits to the synth.Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com12tag:blogger.com,1999:blog-279251740880432906.post-61040391036816037362012-01-08T15:07:00.001-06:002012-01-08T15:10:40.502-06:00Realizing the RealizerThis post was inspired by <a href="http://www.bigcitymusic.com/">Big City Music's</a> announcement this week that they are <a href="http://bigcitymusic.com/index.php?main_page=product_info&cPath=2_55&products_id=735">offering a PPG Realizer for sale</a>. The Realizer, as many synth enthusiasts know, was the legendary attempt by <a href="http://wolfgangpalm.wordpress.com/">Wolfgang Palm</a>'s <a href="http://electronicmusic.wikia.com/wiki/PPG">PPG </a>company to pioneer the soft synth concept using 1980s technology. However, the development costs became the final straw for the financially struggling PPG, and led to the company liquidating itself in 1988. The Realizer never went into production; rumor has it that two prototypes were built. (If this is true, I don't know where the other one is; I've checked the listings of the <a href="http://www.audities.org/audities/index.html">Audities Foundation</a>, the <a href="http://www.synthmuseum.com/nesm/">New England Synthesizer Museum</a>, and the <a href="http://www.eboardmuseum.com/100e_eboardmuseum_eboardmuseum.html">Eboard Museum</a>. None of them list a Realizer, but that doesn't mean they don't have one in their collection somewhere.<br /><br /><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgiAaizp5huy_0bPCvfQWpD1ARU2GMxPcNl5ZrphKne2Q2_BVQmdgZFy6h4vIRBsCaAoq0j5mRpAe0FzjabZ_7NWr3rbkS34t_adIeO-8y6baCMIJ9KN53U-qf7o0O3AgdlGZwnwbuGCFo/s1600/realizer.jpg"><img style="cursor:pointer; cursor:hand;width: 320px; height: 226px;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgiAaizp5huy_0bPCvfQWpD1ARU2GMxPcNl5ZrphKne2Q2_BVQmdgZFy6h4vIRBsCaAoq0j5mRpAe0FzjabZ_7NWr3rbkS34t_adIeO-8y6baCMIJ9KN53U-qf7o0O3AgdlGZwnwbuGCFo/s320/realizer.jpg" alt="" id="BLOGGER_PHOTO_ID_5695366330301931266" border="0" /></a><br /><span style="font-style: italic;">Realizer control desk -- from </span><a style="font-style: italic;" href="http://theppgs.com/index.html">ThePPGs.com</a><br /><br />So what was the Realizer actually? Modern legend has it that it was the first <a href="http://electronicmusic.wikia.com/wiki/Virtual_analog">virtual analog</a> synthesizer. Actually, however, it was both more than that, and possibly less than that. According to Palm, the Realizer was an early attempt to build what we now call a "workstation"; it would have been capable of synthesis, multi-track recording, processing, and mixing. The unit you usually see in pictures, which PPG called the "control desk", is not the whole system; it's only the control unit and user interface. The control desk could interface with a combination of sound modules that actually did the audio processing, and hard disk units (HDUs) which provided audio and data storage, up to 8 units total. The sound modules and HDUs were intended to be somewhat mix-and-match with other PPG products, such as the Waveterm B.<br /><br /><br /><br />The sound module was the piece of the Realizer that did most of the work. Like many of today's digital synths, it contained a bank of digital signal processor ICs -- in this case, eight of the TMS 32010 -- and a Motorola 68010 that managed everything. (The HDUs also contained a pair of 32010s, but it appears that they were not used in the Realizer configuration.) PPG developed three synthesis packages that ran on the sound modules. The one that everyone talks about in regard to the Realizer was the "Minimoog" virtual analog software that reproduced not only the features of the Mini, but also its panel layout. However, there was also application software for additive synthesis and for the sampling and wave scanning method of synthesis that PPG was know for. Further, according to Palm, the software allowed individual processing functions to be mixed and combined in the style of a modular synth. (You could say that PPG invented Reaktor in 1986!)<br /><br /><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg2FL_dish_rlGL6bA-epDJyoXazSej5oWAtAUtFGryCxFQ-97tBYBrz6XSxY4tuIyIvw87L9mVpSDHrCV47xSrgutxbaqZhhubvYPcETT9o-rgoZzetGUtXgLckWkw8glNI6bwEPFoRMo/s1600/ppghdu01.gif"><img style="cursor:pointer; cursor:hand;width: 300px; height: 197px;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg2FL_dish_rlGL6bA-epDJyoXazSej5oWAtAUtFGryCxFQ-97tBYBrz6XSxY4tuIyIvw87L9mVpSDHrCV47xSrgutxbaqZhhubvYPcETT9o-rgoZzetGUtXgLckWkw8glNI6bwEPFoRMo/s320/ppghdu01.gif" alt="" id="BLOGGER_PHOTO_ID_5695370120828751634" border="0" /></a><br /><span style="display: block; font-style: italic;" id="formatbar_Buttons"><span onmouseover="ButtonHoverOn(this);" onmouseout="ButtonHoverOff(this);" onmouseup="" onmousedown="CheckFormatting(event);FormatbarButton('richeditorframe', this, 8);ButtonMouseDown(this);" class="" style="display: block;" id="formatbar_CreateLink" title="Link"><img src="http://www.blogger.com/img/blank.gif" alt="Link" class="gl_link" border="0" /></span></span><span style="font-style: italic;">PPG HDU with stand-alone control unit (in foreground). From </span><a style="font-style: italic;" href="http://www.synthmuseum.com/nesm/">Synthmuseum.com</a><br /><br />Further, the Realizer also incorporated the functions of what we now call a digital audio workstation. It was capable of multi-track recording and editing using the HDUs as storage. It had "plug-ins" for adding common studio effects (although, according to Palm, the reverb software was troublesome and was never completed to a satisfactory degree). And it could do mixdowns, producing a digital master that could presumably be transferred directly to CD mastering systems, although it is not clear how that would have actually worked.<br /><br />The control desk is the part of the Realizer that everyone has seen. It contains a monitor (which was to have been color in the production version, but the prototypes had green-phosphor monochrome), 31 knobs, 6 faders, a data entry wheel, a keypad with numeric and function keys, and a graphics tablet. Each of the knobs and faders had line graphics drawn on the panel from the control to the edge of the screen opening; the software generated lines on the screen that led from the screen edge to the parameters on the screen, and by that method, the user got a visual indication of which knob or fader controlled what on a given screen. The graphics pen would have been used for drawing, and probably also for selecting commands from menus, in the fashion of the Waveterm. The best photo I've been able to find that shows a screen and illustrates the layout and the knob graphics is the following, taken from Palm's old blog on Myspace. Palm describes this photo as being a photo of an early mock-up, which accounts for the crude-looking panel.<br /><br /><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgZHXQsDBiKX93Th9wH6EJHPfzvLO5aI5nThJ_hpYTenWjmXgDJRkMOL80nHuzde9TZRoXxEL06Iw3crwj6bdEScnFFPDsKYCobJRSMnvy1XBBtnaEnMIk8FSYF23siXbcoxMcKBOmi7dQ/s1600/Realizer+Screen.jpg"><img style="cursor:pointer; cursor:hand;width: 320px; height: 217px;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgZHXQsDBiKX93Th9wH6EJHPfzvLO5aI5nThJ_hpYTenWjmXgDJRkMOL80nHuzde9TZRoXxEL06Iw3crwj6bdEScnFFPDsKYCobJRSMnvy1XBBtnaEnMIk8FSYF23siXbcoxMcKBOmi7dQ/s320/Realizer+Screen.jpg" alt="" id="BLOGGER_PHOTO_ID_5695348877438192450" border="0" /></a><br /><br /><br />So what made the Realizer so expensive to develop, enough that it killed the company? One thing that Palm mentions is that in the timeframe when the Realizer (and presumably the Waveterm B also) was being designed, PPG had decided that laying out the circuit boards by hand, which was how they had done all of their previous products, wasn't going to work given the density of the boards they wanted to design. So they purchased an electrical CAD software package (which would have been very expensive in 1986), and then leased a computer system to run it. Unfortunately, apparently the leased computer didn't have sufficient performance, and the system ran very slowly, harming productivity. There probably would also have been a learning curve for the engineers. Palm also mentions, in his account of the Waveterm B development, that the Waveterm B and Realizer were PPG's first products to use 16-bit sampling, and that they had a hard time getting their 16-bit analog-to-digital converters working properly. (In order to build the initial sample library to be shipped with the Waveterm B, they hacked a Sony F1 digital tape system and used its converters.)<br /><br />However, I'm guessing that the real killer was the software. Even though there was probably some commonality with PPG's other products, there would have been a ton of new software to be written. They had to write a lot of new software for the control desk displays and user interfaces; the Moog emulation and the additive synthesis was new, and they had implemented a lot of improvements to the sampling which required new software. (Plus, knowing how things were done back in the day, the 12-bit sample handling software used in the Wave keyboards probably did a lot of "tricks" with unused data bits, which would have had to be re-written to handle 16-bit samples.) The sound modules required a bunch of new software to manage all of the DSPs, not to mention the actual DSP processing code. Also: they were writing all of this in assembly language. Palm states that a C compiler was not available for their systems at the time -- a statement which puzzles me, since C compilers were readily available for most processor families by 1986, and Sun, to name one, had one for the 68000-series CPUs.<br /><br />The final factor was that, due to a combination of circumstances, PPG found themselves without a lot of money to spend on R&D. In 1986 they had invested money in moving production to a larger facility, in part due to robust sales of the Wave, but at about the time the new factory opened, Wave sales began to decline. The Wave, especially by the time of the 2.3 revision, had very sophisticated capabilities for wave scanning and manipulation of samples, but a lot of users didn't care about that -- they only wanted basic sampling and playback, or just playback of canned sample libraries, and so they gravitated towards less expensive samplers like the Emulator or the Ensoniq Mirage. The Wave was PPG's bread-and-butter product, so when sales declined, the company's revenues suffered. A few Waveterm B's were sold, and a few HDUs were sold as stand-alone products, but the Realizer never reached production before Palm and the other founders realized that they were going to run out of money. They liquidated rather than continue and be forced into bankruptcy.<br /><br />Going back to the sale by Big City Music, I don't know exactly what it is that they are selling. The ad shows only the control desk. Although that item no doubt has significant collection value by itself, the point remains that if the goal of a buyer is to get the system actually running again, it won't do anything without at least one sound module and one HDU. Perhaps Big City has these items and is including them in the sale; the ad copy doesn't say.<br /><br />Below is a link to a 1987 demo, from Palm's Myspace page. The Realizer's control desk can be seen on the left for much of the video. (The device that the demonstrator is holding appears to be a stand-alone control for an HDU, and not part of the Realizer configuration.) Note that the demonstrator never touches the Realizer control desk, which suggests that the software was still not stable at this point.<br /><br /><a href="http://mediaservices.myspace.com/services/media/embed.aspx/m=,mr=60628615,t=1,mt=video">http://mediaservices.myspace.com/services/media/embed.aspx/m=,mr=60628615,t=1,mt=video</a><br /><br />Finishing up with a historical curiosity: The astute observer may have noticed that in the photo of the control desk, following the first paragraph of this post, the desk does not have the data entry wheel. I don't know if this implies that prototypes were built both with and without the data wheel, or if it was added to the pictured unit after the photo was taken.Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0tag:blogger.com,1999:blog-279251740880432906.post-23298147992993470892012-01-01T22:30:00.001-06:002012-01-01T22:36:50.491-06:00Thoughts on panel graphicsHappy New Year to synthesists everywhere! Looking back at the last year, I see that for a while I've been focusing almost exclusively on gear. That was not really my intent. I'll let you in on a little secret: One of the reasons I created this blog was to use it, in a sense, as my own notebook; a lot of the things that I post are things that I want to be able to refer back to myself. For instance, since I did the <a href="http://sequence15.blogspot.com/2010/03/still-more-on-motm-650-arpeggiators.html">detailed description of the MOTM-650 MIDI-to-CV interface</a> back in 2010, I've referred back to it several times when I needed to sort out some parameter or other.<br /><br />However, I'm not doing this just for myself. If I was, I'd just keep a notes file on my computer, and not bother with a blog. My New Year's resolution, in regard to Sequence 15, is to share thoughts of all sorts with regard to synthesis and electronic music. Just blogging about gear is too limiting, and accounts for the dearth of posts.<br /><br />So here goes... In the ongoing discussion about the differences in the user community between modular synth performers who prefer the Eurorack format, and the performers who prefer the "5U" formats (<a href="http://electronicmusic.wikia.com/wiki/MOTM">MOTM</a>, <a href="http://electronicmusic.wikia.com/wiki/Dotcom">Dotcom</a>, and <a href="http://electronicmusic.wikia.com/wiki/Modcan-A">Modcan-A</a>), one thing that's often debated is the style of panel graphics commonly seen on Eurorack modules, versus the usual style of 5U modules. Let's compare: Here is a (rather murky) photo of my <a href="http://encoreelectronics.com/cont_ueg1.html">Encore Universal Event Generator</a>:<br /><br /><br /><a onblur="try {parent.deselectBloggerImageGracefully();} catch(e) {}" href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhyr_wmC-S3uHteRf3a3fvquySQ5jLRLyJtCbDztYF8M21ADeFFXp96re7sDKx2XT9TXEh2No8ufPEHbo0pZe7vxCLu076TFQ0DQh5LqjvUlMLVi3C8VNu_Gsum2vUK7FUa6MHbR2Jwe5w/s1600/IMG_5269.jpg"><img style="cursor:pointer; cursor:hand;width: 150px; height: 320px;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEhyr_wmC-S3uHteRf3a3fvquySQ5jLRLyJtCbDztYF8M21ADeFFXp96re7sDKx2XT9TXEh2No8ufPEHbo0pZe7vxCLu076TFQ0DQh5LqjvUlMLVi3C8VNu_Gsum2vUK7FUa6MHbR2Jwe5w/s320/IMG_5269.jpg" alt="" id="BLOGGER_PHOTO_ID_5692868415460847826" border="0" /></a><br /><br /><br />By comparison, here, from <a href="http://www.analoguehaven.com/">Analogue Haven's</a> Web site, is a photo of the <a href="http://www.makenoisemusic.com/PATCHPAL.html">Makenoise Maths</a>:<br /><br /><a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjoJ3ZJ08QYpclds3MZwEzV2uPdGLUuZ-0NXijjbA2m4xvNgP-RCGxldD_5VPbiCdClkCSejG4N7y6uQffCeH7fW0BI18Yh8dwkv-ex64rXN_wkF0Ng9Eeuyo2Ht-rX_fY33JhFXNaoVDY/s1600/mathss.jpg"><img style="cursor:pointer; cursor:hand;width: 200px; height: 250px;" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjoJ3ZJ08QYpclds3MZwEzV2uPdGLUuZ-0NXijjbA2m4xvNgP-RCGxldD_5VPbiCdClkCSejG4N7y6uQffCeH7fW0BI18Yh8dwkv-ex64rXN_wkF0Ng9Eeuyo2Ht-rX_fY33JhFXNaoVDY/s320/mathss.jpg" alt="" id="BLOGGER_PHOTO_ID_5692874593937994610" border="0" /></a><br /><br />The Encore UEG clearly maintains the tradition of the vintage Moog modulars: white graphics on a flat back background, and controls in neat rows and columns. (Further, it follows the MOTM format convention of putting the I/O jacks at the bottom, although it doesn't really conform to the whole MOTM standard grid due to the large number of knobs.) All the controls and jacks are labeled in a clean font, and the panel has index marks for the knobs. Line graphics are used to indicate associations between controls. Most MOTM, Dotcom, and Modcan-A format modules follow this pattern; in the world of 5U, Modcan's B-series modules (which are MOTM format) are considered a bit radical for having black graphics on a white background. There have been a few other makers of large-format panels who have used colored text and line graphics, but even they tend to stick to the black background and standard fonts.<br /><br />Now let's compare with the Maths. White background with a red border around the edge of the panel. (And that's considered conservative in Euro-land.) Zig-zaggy graphics that show the flow of signal through the module. There's four input jacks; they are at the top of the module, and you have to read the manual to realize that they are the four jacks pointed at by the small arrows. Knobs and jacks scattered hither and yon, although the panel is symmetrical. (It has two processing channels; the two outside ones do basically the same thing, and the same goes for the two inside ones.) Functions of some of the jacks are indicated only by the signal flow graphics. You have to look rather closely to see the little math operator symbols that label some of the controls. The knobs don't have any indexing, and there are two illuminated pushbutton switches whose purpose is not indicated at all. And I don't know where the hell Makenoise came up with that font; maybe they made it themselves.<br /><br />If that sounds like I'm ragging on Makenoise, I'm not intending to be. If you go to Makenoise's Web site and look through the descriptions of their modules, you realize that Makenoise has its way of doing things, and once you've studied it and gotten into that groove, most of those panel markings make intuitive sense. Where you start to run into problems in Euro-land is when you realize that the Makenoise way of doing things is not the same as the <a href="http://theharvestman.org/">Harvestman </a>way of doing things, which is not the same as the <a href="http://www.wmdevices.com/">WMD </a>way of doing things, etc.<br /><br />Euro users put up with this, in part because it looks cool. But I think there is also more of an aesthetic in the Euro world of being more willing to patch something up, turn some knobs, and see what happens, where in the 5U world, users tend to want things to be more precise (or "anal" if you prefer). This is just a general statement based on anecdotal data; it certainly doesn't apply universally. However, I do note that there are a few small makers in the Euro world who are willing to silkscreen something on a panel that has nothing to do with a panel's function, or just leave a panel blank; almost no one in the 5U world would ever do that. I do note that even in Euro land there has been a bit of a reaction to some of the more excessive panel designs. <a href="http://pittsburghmodular.com/">Pittsburgh Modular</a> makes a wry comment on it with their 1960s-embossed-label aesthetic. <br /><br />Other aspects of small vs. large format have been discussed to death already: 5U takes up a lot more space; Euro/Frac knobs are too small for large fingers, 5U panels cost more to make, 3.5mm jacks break off too easily, etc. However, I think there's one other, very practical concern. It's been noted that 5U users tend to be, on average, older than Euro users. Here's the other reason us 5U guys like things nice and clean: when we look at something like the Maths panel above, <span style="font-style: italic;">we can't see the panel!</span> Our eyes aren't as good as they used to be. If we had a Maths, we'd have to get a magnifying glass out every time we wanted to use it. Panels like that give us headaches. We have to stick with nice high-contrast panels with clean labeling that we can see.<br /><br />And besides, we like the laboratory-test-equipment aesthetic. Our moms all say it reminds them of their father's ham radio gear.Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com2tag:blogger.com,1999:blog-279251740880432906.post-41470744043840510212011-12-28T23:03:00.001-06:002011-12-28T23:05:05.821-06:00WashingtonA new Statescape... named Washington, and done entirely with the Solaris. More <a href="http://home.hiwaay.net/%7Ecornutt/Music/Web%20Page/music.html#Washington">at the Web page</a>.Dave Cornutthttp://www.blogger.com/profile/17769989714705003390noreply@blogger.com0