John Bowen Synth Design

Congrats on your pre-order 3rdConstruction. I use a mixture of analog and digital synths myself. The Solaris is the digital synth we've been waiting for. It has a great interface. It makes you want to turn knobs and touch the synth. It has the sound. Whether digital or analog is your thing, the Solaris has it all and will be able to add more synth features in the future.

So for someone like you who only knows analog synths, this will be very at home in your hands, but will also be the best introduction into digital synths. It has the power and flexibility of vst, with sonic character and hands on capability of classic analog synths.

whoops. as a point of order, i had picked up on the 24/96 processing from another discussion thread. but just noticed this on the solaris page:
[quote]Internally, all operations are 32-bit floating point and audio signals and busses run at 96 kHz for pristine audio quality.[/quote]

[quote="matocaster"]good luck finding any digital synth or va with specs this powerful[/quote]
specs quoted in terms of osc, filters, etc. are at least something i can understand. it was on the basis of those specs, plus the knob-laden interface plus the sound i heard on john's demos that convinced me to step up, sign up, & become a preorder customer. so i'm already sold! it was things like having 4 osc, the absolutely unique rotors, the filter resonance, the feedback capability, & all those modulation options that i could not resist.

my question about the digital specs comes from the fact that i'm an analogue geezer from pre-MIDI days, and i'm just now trying to rise from the dead & get back into electronic music making. solaris will be my first digital synth, ever. have never even tried a va or vst before. i never understood what bothered me when i listened to demos of things like the virus or roland's pseudo-junos, or vsts. well, i still don't, to be truthful. something seems to be missing. they sound a little flat (as in 2 dimensional) maybe more like a picture of a sound.

for reasons i also don't understand, the solaris sounds different, more real & alive, like an analogue synth, but different of course. and i'm just trying to understand what is that makes the solaris different. i suspect it's due to a number of things such as the internal processing rate, details of the mystical code, the D/A conversion, & who knows what else. i'm sure the 24/96 processing rate is an important piece in this.

i suspect this is why other companies don't quote bit rate/freq specs on their machines. but does anyone know?

[quote="3rdConstruction"]I have a question about the 24 bit 96 kHz internal processing of the Solaris. I am completely new to digital synthesis & am still learning. I've looked for similar specs on other digital synths like the Access Virus & Roland machines, but cannot find any similar specs reported. Can anyone tell me how the 24/96 processing rate of the Solaris compares to other digital synths, e.g. Virus (just as one example)? Digital specs do not seem to be quoted in a common language.

If this has been covered elsewhere, I apologize for missing it in my search. If you'd be good enough to just point me in the right direction, I would be grateful...[/quote]

I would say the Solaris is in an entirely different league then any Access or Roland synths or romplers. The processing power blows them all away. No comparison. As for sound quality for a digital synth, it will be unequaled. You can't find similar specs because they don't exist in hardware. You would have to find it in a Vsti.

http://forums.johnbowen.com/viewtopic.php?t=459

http://forums.johnbowen.com/viewtopic.php?t=413

http://forums.johnbowen.com/viewtopic.php?t=417

good luck finding any digital synth or va with specs this powerful:

Solaris
* 4 Oscs, 2 Rotors, 4 Mixers, 4 Filters, 4 Amp/Pan sections, 8 Envelopes, 5 LFOs, 4 External Inputs, Flanger/Chorus, Phaser, Delay, 3 band EQ, Overdrive
* 2 Vector Mixers
* 2 separate AM (amplitude modulation) sections, with Ring, AM, Rectify, & Clip algorithms
* Joystick and multi-touch Ribbon controllers
* Arpeggiator and control Step Sequencer provided, with MIDI output
* Performance buttons include 2 assignable switches, Arpeggiator Start, Sequencer Start, Hold, (Tap) Tempo, Unison
* Polyphony count - expected to be 24 voices with all 4 oscs, 4 filters, 4 mixers, envelopes, LFOs, etc. running
* 96 kHz internal processing rate
* Insert FX pre-/post each filter section, with BitCrush, Decimate, and Distortion
* 4 pair of outputs; Main pair for v1.0, additional outputs reserved for future use (when Multi-Timbre Mode is implemented)
* separate Headphone out

Detail per section:
Oscillators - each osc type selects from standard waveshapes (MultiMode Osc), wavetable (PPG) type, sample (WAV) playback, CEM (Prophet 5) type, or Prophet VS type. The MM (MultiMode) type provides the following waveshapes:
Sine, triangle, ramp, saw, pulse, noise, S&H, morphing sine-to-saw, morphing sine-to-square, and a stacked "supersaw" with varible detune (based on the Shape parameter).

Hard Sync is only available for MM saw, ramp, pulse, and the CEM osc models.
New types will be added as they are developed via an upgrade to the OS.

Individual "analog-style" glide is available for each oscillator.

There are 4 mod paths. Each one is freely assignable to select exponential frequency (normal pitch mod), Linear FM, or Shape as their destination. Mod Sources include any oscillator, any filter, the 4 external inputs, any of the lfos, envelopes, controllers, etc.. A 'sidechain modulation' function is provided for each path, using Controller (non-audio rate) signals. Controller signals are all lfos and envelopes, velocity, note, aftertouch, mod wheel, ribbon, joystick, select MIDI controllers, assignable CC knobs, etc.

Filters - 4 filters, each with selectable inputs.

Filter types include:
1) all pole possibilities for the MultiMode (MM1) filter, including 24 dB Lowpass, Highpass, and Bandpass, 12 dB Lowpass, Highpass, and Bandpass, and 6 dB Lowpass, Highpass, and Bandpass, along with some other combination modes, for a total of 23 variations.
2) 24 dB Lowpass modeled on the Prophet 5 Rev1 filter (SSM2040)
3) 24 dB Lowpass modeled on the Rev 3 Prophet 5 (CEM3320)
4) 12 dB Lowpass modeled on the Oberheim SVF
5) Comb/Tube filter (the "tube" is a comb with negative feedback)

New filter types will be added as they are developed via an upgrade to the OS.

For filter modulation, it's the same structure as the Oscillators - there are 4 mod paths. Each one is freely assignable to select Cutoff, Resonance, or Damping (if Comb/Tube is selected) as their destination. Mod Sources include any oscillator, any filter, the 4 external inputs, any of the lfos, envelopes, controllers, etc.. A 'sidechain modulation' function is provided for each path, using Controller (non-audio rate) signals. Controller signals are all lfos and envelopes, velocity, note, aftertouch, mod wheel, ribbon, joystick, select MIDI controllers, assignable CC knobs, etc.

ADSRs - there are 6 standard DADSRs. Each overall amount can be modulated by Velocity, and each segment can be individually modulated from Velocity, Note, Mod Wheel, and assignable Midi Controllers (CC1-CC5). Also, each segment can have a variable slope, from linear to exponential.

Looping Envelopes - there are also 2 looping envelopes, each with 8 Time&Level segments. There is overall modulation possible of Time and Level.

LFOs - there are 5 identical LFOs, with the fifth being permanently connected to the frequency of all oscs (therefore, it is called the Vibrato LFO). The LFOs have the standard waveshape types, and range from 0-524 Hz. There are parameters for Delay Start, Fade In, Fade Out, Rate, Waveshape, Retrigger, Phase, Level, MIDI Clocking, and Offset (offset provides a positive unipolar signal for the lfo outputs). There are 3 mod paths, similar to the Oscillator modulation structure. The destinations here are selectable for Rate or Level.

VCAs - There are several models implemented for the final output stage circuit. VCA Types include: Linear, Log, and Sigma (Minimoog style). There is 1 mod path for the VCA, and 1 for the Pan position.

Vector Synthesis - There are 2 Vector Mixer sections. The Joystick (non-spring loaded) in the leftmost section is normally connected to both Vector Mixers, but can be disabled.

AM Sections - 2 Amplitude Modulation sections, each of which have Carrier, Modulator, Algorithm and Shaper parameters. Ring Mod is one of the algorithms provided.

Effects - Initially available will be delay, flanger/chorus, reverb, EQ, Overdrive. As with the other sections, additional FX types such as a vocoder or resonant filter bank will be added as they are developed via an upgrade to the OS.

I have a question about the 24 bit 96 kHz internal processing of the Solaris. I am completely new to digital synthesis & am still learning. I've looked for similar specs on other digital synths like the Access Virus & Roland machines, but cannot find any similar specs reported. Can anyone tell me how the 24/96 processing rate of the Solaris compares to other digital synths, e.g. Virus (just as one example)? Digital specs do not seem to be quoted in a common language.

If this has been covered elsewhere, I apologize for missing it in my search. If you'd be good enough to just point me in the right direction, I would be grateful...

[quote="ThreeFingersOfLove"]John,

can you please tell us if any of the filters in the Solaris self-oscillates? If so how does it handle the resonance buildup close to 12 KHz? Do the filters use oversampling internally or is there something like a soft compressor to keep the buildup well-behaved?

Also can you please tell us how does the mixer behave when all the signals from the oscillators are at full level? Does it scale back the signal so there is no overload or does the user must decrease the respective levels manually so as not to have any overload?

Thanks,
Yannis[/quote]Hi Yannis,
All filters in the Solaris do self-oscillate. As far as any internal control for build-up - well I don't know, since I didn't code them, but I do not find any problem of a "resonance buildup' around 12 KHz (or anywhere else). For the mixers, as you say - the user has to decrease the levels to avoid overload, especially if you are using all sine waves for each input, but this is also dependent on where the Master Volume is set.
On the other hand - if you want the overload, it is there...
-john b.

96khz internal processing was the right decision IMO. With the solaris, sound quality should be #1. And there IS a difference in SQ. I testet with difference between 44.1 and 96khz and synth plug-in sound with my pulsar2 cards, and it was quite noticable.

John,

can you please tell us if any of the filters in the Solaris self-oscillates? If so how does it handle the resonance buildup close to 12 KHz? Do the filters use oversampling internally or is there something like a soft compressor to keep the buildup well-behaved?

Also can you please tell us how does the mixer behave when all the signals from the oscillators are at full level? Does it scale back the signal so there is no overload or does the user must decrease the respective levels manually so as not to have any overload?

Thanks,
Yannis

So the final spec is 6 AD21369 DSP (so half the power of the XITE-1) ?

So to say it is equivalent to a full 50 old Scope DSP system.

CheerZ

Totally agree with this design.
Anyone here ever heard that Viscount VA version of the Oberheim?
Being an old Oberheim user I was interested but noticed the 12 voices of polyphony.
Immediately that translated into 12 weak voices.
When I played the first chord I confirmed my suspisions.

I have never played a VA synth I liked yet except Solaris.
I am happy that JB chose to use 6 X ADP21369 chips and 16 voices.
If I wanted a synth with 500 presets and 96 voices I would have made another choice, and then probably had to layer patches to get a giant sound, which in my experienc3e sounds like Gooo and lacking any definition.

This synth is a surgeons tool, that's why I am buying one.

+1 on sound quality. There are a lot of awful sounding vst's and other stuff. Good sound is critical.

+1 on predictable DSP allocation.

I think the decision logic there is really good. Just wanted to offer a positive agreement on that.

[quote="ThreeFingersOfLove"]I agree with Septic Studio. If sound quality is to be compromised it's better to leave it at 96 KHz despite the extra voices. Given the fact that there are 4 analog outputs, it makes sense to have a 4 part multi-timbrality, no? Each part gets its own output and everyone is happy. John, the JP-8000 has a sorta user defined voice allocation. For instance, it has 8 voices, 2 parts multi-timbrality so if you have set up something like a bass and a pad sound you can explicitly allocate 1 voice to the bass and 7 to the pad. I don't know if you have something like that in your mind for future OS updates, but I think it works ok. I would be very happy to see the voice allocation saved per patch! Also, if the onboard FX is disabled will the DSP (and hence the extra 3 voices) be available?[/quote]We did discuss some fixed allocation of voices for parts, as you suggest. And yes, 4 part Multitimbre has been the plan all along....
As to disabling the onboard FX to free up the DSP - I have no idea if this could work. I'll have to ask.
-john b.

[quote="SepticStudio"]a 48/96 internal switch would be okay i think, but I personally gonna favor the 96 HiQ output even if it costs some voices.[/quote]Yes, but as I tried to explain, this would actually require different code inside, one set for 48 kHz, and another for 96 kHz. He would have to re-write much of the code, which is basically doubling the amount of work needed.
-john b.

I agree with Septic Studio. If sound quality is to be compromised it's better to leave it at 96 KHz despite the extra voices.

Given the fact that there are 4 analog outputs, it makes sense to have a 4 part multi-timbrality, no? Each part gets its own output and everyone is happy.

John, the JP-8000 has a sorta user defined voice allocation. For instance, it has 8 voices, 2 parts multi-timbrality so if you have set up something like a bass and a pad sound you can explicitly allocate 1 voice to the bass and 7 to the pad.

I don't know if you have something like that in your mind for future OS updates, but I think it works ok. I would be very happy to see the voice allocation saved per patch!

Also, if the onboard FX is disabled will the DSP (and hence the extra 3 voices) be available?

Regards,
Yannis

a 48/96 internal switch would be okay i think, but I personally gonna favor the 96 HiQ output even if it costs some voices (well half of it i think)

15 voices will be more than sufficient for mono timbral work and even with 2 parts is is very good doable, more than 8voice poly a part i never use myself. You could set up like this:

1 x 8 voice part
1x 6 voice part
2 x 1 mono part ( if the coder can get that extra voice from DSP 6)

that will be more than sufficient for me...I really like the non-compromise aproach of the Solaris. Like that 96 khz feature.

But a switch can be a welcome addition i think..especially if one wants to make a whole track with the Solaris only.