Wire Less: Part 2, Belkin RockStar™

Posted on September 14, 2021 by pj

Well, as you might have guessed, your Apple iPad synthesis rig cannot be entirely wireless (at least inexpensively so). The audio part ain’t there yet. Bluetooth audio has too much latency and Apple does not provide any other built-in solution. So, it’s wires, again, folks.

In my first post, I discussed the MIDI part: an Apple iPad running Korg Module Pro communicating with a Korg Microkey Air 49 over MIDI BLE. The MIDI part works quite well and I haven’t had any issues.

As to the audio part, I absolutely detest the iPad 3.5mm phone connector. Any plug is exposed and even the slightest movement emits a nasty grind from a powered speaker or other downstream audio device. Buzzzzzz — literally. I simply wouldn’t risk this method in a high volume situation in front of a church congregation.

The Lightning connector, however, is relatively snug and secure. Like most people, I opt for a Lightning-based solution. With Bluetooth handling MIDI duties, one needs only an audio interfacing solution.

Sorry, USB-C people, I don’t have a USB-C iPad and don’t address USB-C solutions here.

By now, everyone knows that the Apple Lightning to USB Camera Adapter is not just for cameras. Thanks to the Camera Adapter, you can hook up a USB-based audio interface to your iPad (or iPhone). The Camera Adapter’s street price has dipped to $9USD, making this a very inexpensive solution. If — if — your bus-powered audio interface draws a small amount of power, the Camera Adapter will supply power although the iPad battery drains faster. If your bus-powered audio interface is a power hog, you will get the infamous “This accessory requires too much power” message and iPad will refuse to play along, shutting down the interface.

Enter the Apple Apple Lightning to USB3 Camera Adapter ($39USD). The USB3 Camera Adapter has two ports: a USB3 host port and a Lightning charge port. The USB3 port connects to your audio interface while the Lightning port connects to an AC adapter. The Lightning port both charges the iPad battery and supplies power to your audio interface.

At one time, I considered the Steinberg UR22C as a solution for both desktop use and mobile. The UR22C has the necessary functional features and sports its own external power port. This is definitely another way to go and I wish more manufacturers would provide an external power port and not rely solely on bus power. I decided to eschew “yet another box at the gig” in favor of an even smaller, lighter solution. (For desktop, I eventually chose the Yamaha AG06, BTW).

Belkin adapter and the tangled mess o’cables

For smaller and lighter, I went with the somewhat neglected Belkin 3.5mm Audio + Charge RockStar™ ($40). This Belkin adapter provides a more robust 3.5mm jack and a Lightning charge port. My only beef is the short iPad to adapter Lightning cable. The short cable is good enough for casual listeners, but I feel that it still requires too much stress on the 3.5mm jack. I added a 2 meter Lightning extension cable, letting me rest the adapter and 3.5mm plug on the floor. This arrangement reduces the physical stress on the iPad Lightning port, too. One flexible cable to the iPad makes it easier and safer to move the iPad during a gig.

A few fine points. I configure Korg Module Pro for MONO out and use a 3.5mm stereo to 1/4″ breakout cable (tip and ring) for the final audio connection. MONO is close enough for rock’n’roll. I realize this is audio religion to purists. 🙂

If you don’t want 3.5mm audio, Belkin offers the Belkin Lightning Audio + Charge RockStar™ ($45USD). It has a two Lightning ports: one for audio and one for charge.

Before closing, I want to mention an ultra-cheap, simple solution: an Apple Lightning to 3.5mm Headphone Jack Adapter plus an extension cable. You may already have one of these adapters! When Apple dropped the 3.5mm jack, it began selling these adapters so people could connect their headphones to the jackless iPhone. It’s ultra-inexpensive at $8USD (street).

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Random answer day (1)

Posted on September 9, 2021 by pj

Maybe it’s the first day of the regular NFL season or the phase of the moon. Here’s a recap of a few questions that came into the forums.

How are arranger/synth preset voices stored? First, one may ask, “How is a preset represented?” Typically, a preset voice consists of waveforms (AKA “samples”) and voice (meta-)data. The voice data control how the sample-playback engine applies filtering, amplitude envelope, modulation and so forth. The waveforms, of course, provide the basic digital audio data.

There is such a broad range of arranger/synth products at different price points, that the amount of storage and the kind of storage varies quite a lot.

The lowest of the low in the Yamaha range: PSS-A50, -E30, -F30, PSR-F51. Presets are stored on a 2MByte serial flash ROM and are loaded into the processor (SWLL) at start-up. The 2MBytes include code, too! Tone generation is integrated into the SWLL. Insanely small, and very low cost.

The highest of the high in the Yamaha range: Genos. Factory presets are stored in four 1GByte ONFI NAND flash devices. Expansion memory consists of two 1GByte ONFI NAND flash devices. Wave memory connects directly to external tone generators (SWP70).

I’ve looked at the diagrams for Genos and I’m not sure about the size and function of those memory units, especially Genos USER memory and expansion memory.

Yamaha confuses people when they speak of “user memory,” “internal memory,” etc. They are usually referring to logical, user visible storage.

When getting down to the hardware level, there are many different physical memory units. since we’re not discussing fairy dust or magic, the logical storage must be assigned to one or more physical memory units. And, of course, the physical memory units themselves may be composed of multiple integrated circuits. The other dimension is “what communicates to what.” Memory is passive and needs a processor to initiate reads and writes and to do something with all that data. At the physical level, a memory unit essentially belongs to a single processing unit (host computer, tone generator) and directly communicate with it.

Sometimes I think of the SWP70 as a parallel processor just like a GPU. The CPU/SWP70 is not exactly analogous to host CPU plus GPU, however. Graphics memory is shared between CPU and GPU. The SWP70 does not share its waveform memory with anybody — it’s dedicated to the tone generator. That’s why installing an expansion pack (voice library) is kind of slow and technically complicated, and why a Genos reboot is required.

Staying with Genos, Genos has two SWP70 tone generators: one handles factory presets and the other handles user expansion voices. The factory SWP70 has 4GBytes of flash memory while the expansion flash memory has 1GB of flash memory. That’s physical memory. Yamaha boosted the effective capacity to 3GB expansion through compression.

The SWP70s also have DSP RAM. As a user, you never know about this memory. It’s scratchpad memory for DSP effects. Physically, the DSP RAM is completely separate and independent from the waveform memory, and communicates with only its parent SWP70.

The host CPU has two kinds of memory (as determined by its bus interfaces): 1GB of working RAM on the CPU memory bus (EMIF) and two embedded eMMC memory devices that act like solid state storage drives (MMC0 and MMC1). As far as a user is concerned, the user never sees the 4GB eMMC drive (MMC0) just like you don’t see the DSP RAM; it’s hidden. The MMC0 drive contains the Linux operating system kernel and the root file system.

The user sees only part of the second 64GB eMMC drive (MMC1). The user sees the logical storage which Yamaha calls “Internal memory” or “USER drive.” What’s in the remaining 6GB? I don’t know — Yamaha haven’t left any clues.

What about Montage and its 5.67GByte waveform memory? 5.67GB is the capacity when the waveforms (samples) are compressed. Again, this is logical storage capacity.

Montage has two SWP70s. One SWP70 is dedicated to FM-X and it does not have waveform memory. The second SWP70 handles AWM2 synthesis (sample playback) and has waveform memory connected to it. The waveform memory consists of four 1GByte devices totaling 4GBytes. Thanks to Yamaha’s proprietary compression, Montage stores 5.67GBytes-worth of data in the physical waveform memory. The remaining space, 1.75GB physical, is available for user samples.

How does sample capacity relate to price? It doesn’t. Component cost is outweighed by manufacturing costs, software development cost and sound design cost.

If the memory components are so cheap, why isn’t there more waveform memory? If there was more, then you wouldn’t buy the Mark II model, would you? 🙂

I understand that E30/F30 do NOT offer velocity sensitivity. My question is about the internals. Is it confirmed that it’s a keybed with two switches per key, that just aren’t supported in software?

Yes, you need to be careful here. There are hardware model differences: E30 and F30 are not velocity sensitive. A50 is velocity sensitive.

There are two different keybed printed circuit boards (PCB). Yamaha part number VAY27800 for F30/E30 and VAY28500 for A50. The A50 PCB has the necessary diodes installed for velocity sense. The F30/E30 PCB does not have the diodes. Further, the A50 board has a 12-pin connector while the F30/E30 board has an 11-pin connector — perhaps to avoid assembly mistakes.

Yamaha Reface key switch matrix schematic

Is velocity sense worth the extra bucks? There may be other differences, too, but these differences are plainly visible.

And the usual caution/disclaimer — kiss the warranty good-bye! For the money, the PSS should be good mod-fodder. Korg probably sold a mess o’monotron that way.

Wire Less: Part 1, Korg Microkey Air 49

Posted on September 3, 2021 by pj

With the pandemic raging, I’m searching for ways to reduce my physical gig footprint and schlep factor. I thought I would share my adventure in battery-lowered, almost wireless keyboard-land.

Months ago, I had a good experience with Korg Module Pro. It has the range of high quality sounds that I need for my church gig. So, I decided to eschew battery-powered MIDI modules like the MidiPLUS miniEngine USB and go iPad and Korg Module Pro.

I tried a bunch of controller candidates. (See the end of this post for more info.) I had the best experience and minimal number of wires with built-in Bluetooth MIDI. The SHS-500 Sonogenic, in particular, is nearly ideal:

Pluses: Built-in Bluetooth, pitch bend and mod wheels, decent mini-keys, narrow depth is good for a lap-board.
Estimated battery life is OK (10 hours); AC adapter jack is well-placed and secure.
Minuses: 37 keys (3 octaves), no expression pedal input, mod wheel works backwards when played in one’s lap.

No, I am not playing the SHS-500 as a keytar. I find the whole keytar thing to be gimmicky and not appropriate for church. I intend to play the controller in my lap, thereby keeping my physical profile small. (Social distancing!) A lap-board lets me ditch the keyboard stand, minimizing schlep.

Mini-keys deserve comment. Mini-keys enable short, lap-held keyboards. They are very lightweight and easy to transport. If the basic key feel is good, I make peace with play-ability.

My trouble isn’t so much with key size. It’s that three octaves (37 keys) are too short. Many melody and bass lines require two octaves and a player needs two octaves below middle C and two octaves above. Otherwise, I do unnecessary mental and hand gymnastics in real-time to fit the music onto the keyboard. That ain’t right.

Just me? Watch Harry Connick Jr. rock a 3 octave Reface CP. Harry sez, “There’s not a lot of room here.” [Tonight Show: Jimmy Fallon, NBC, 1 September 2016, Playing starts at 3:00.]

In the end, I broke down and bought a Korg Microkey Air 49. It is a good size for a lap-board and the Korg Natural Touch mini-keys ain’t too bad. The Microkey Air firmware was already at v1.04 when it arrived and it connected with Korg Module Pro under IOS 14.1 without a problem. [More on this in a future post.]

The Microkey Air 49 has an estimated 30 hour battery life. Good thing, because Bluetooth operation must use battery power (two AA batteries). Be sure to have two spare AA batteries at the gig; there isn’t a USB powered safety net.

The Microkey Air has a footswitch input. Expression input would be better. Of course, connecting a pedal to the Microkey Air adds a cable. Fortunately, Bluetooth pedals like the Airturn BT200-S4 get the job done. I have a BT200-S4 and found it easy to switch sustain, etc. via Bluetooth in Korg Module Pro. The BT200-S4 is small and light, not any worse than schlepping a wired sustain pedal.

I made a few advances with iPad wiring along the way. The Korg Microkey Air 49 is working out pretty well and I’m practicing with it every day. I have a few custom layers in Korg Module Pro and the day is coming when I’ll try out the rig in front of a congregation.

Going native

For completeness sake, I tried “going native” with sounds built into the Yamaha SHS-500 Sonogenic, Yamaha Reface YC, Yamaha PSS-A50 and Korg microKorg XL+ — all fine battery-powered instruments in their own right with sounds appropriate for rock, soul, jazz, and pop, but not church. I need good strings, reeds, classic organ and gospel B-3. Before moving on, I give props to the Reface YC as it is truly gig-worthy and have play it on the job.

Blooming BLU

I also tried using “the natives” as Bluetooth MIDI controllers. All of the candidates have USB and/or 5-pin MIDI DIN ports, and can be fitted with Yamaha UD-BT01 and MD-BT01 wireless MIDI adapters. The candidate keyboards are battery-powered, so what the heck!

Yamaha UD-BT01 (with AC adapter) and UD-BT01 Bluetooth MIDI

To make a long story short, all candidates worked well with the Yamaha adapters and with Korg Module Pro on iPad — even the lowly, dirt-cheap PSS-A50. A few specific observations:

The Yamaha UB-BT01 not only does Bluetooth MIDI, it supplies power to the PSS-A50. If you must add a cable to connect the A50 to the UD-BT01, you might as well get power, too, and save batteries. If you own a PSS-A50 and want to go Bluetooth MIDI, don’t hesitate!
The Reface YC has the added bonus of an expression pedal input. An expression pedal is a vital part of my gig toolkit. Korg Module Pro will connect simultaneously to more than one Bluetooth MIDI source (like the BT200-S4 previously mentioned). In one experiment, I used Reface YC as my expression source while playing the black and whites on the SHS-500. Neat. I might add the new Boss EV-1-L wireless expression pedal once it ships.
I looked into expected battery life. The Korg Microkey Air is the best at 30 hours estimated life. The other solutions are burdened by tone generation and DSP. The added power-burn is unnecessary if we’re not using the internal synthesis engines.

Even though you take a power hit, an internal engine is a good back-up in case there is a technical problem with Bluetooth, the iPad or Module Pro.

    Instrument     Estimated battery life 
    -------------  ---------------------- 
    Microkey Air          30 Hours 
    PSS-A50               20 Hours 
    SHS-500               10 Hours 
    Reface YC              5 Hours 
    microKorg XL+          4 Hours

In terms of key feel and play-ability, all candidates are acceptable. The Yamaha HD mini-keys are more synth- and organ-like, and are good for legato (especially organ). The Korg Natural Touch mini-keys are more piano-like — good for striking, not quite as good as Yamaha HD for legato. Unlike Microkey Air 49, the other candidates are 37 keys and are too short for unfettered play.

                           Key dimensions 
                        -------------------- 
    Instrument          Width  Length  Depth 
    ------------------  -----  ------  ----- 
    Reface HD            19mm    88mm    9mm 
    Korg Natural Touch   20mm    80mm    8mm 
    MODX                 21mm   133mm   10mm 
    Genos FSX            22mm   133mm   10mm

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PSS chorus: A dusty look back

Posted on September 1, 2021 by pj

Here’s a look into the past — and maybe, the present.

A PSR Tutorial Forum member inquired about the chorus effects in the PSR-E463. The PSR-E463 has the usual system chorus effect and the newer DSP chorus effect. I’m going to focus on the older system chorus effect.

The PSR-E series chorus system effect date back to the earliest days of Yamaha XG and arranger keyboards. These are low-cost entry-level keyboards and usually contain a single integrated circuit (IC) which integrates the main processor (CPU), tone generator and effect units. The most price- and cost-sensitive models integrate the wave memory (samples) on the IC, e.g., the SWLL (PSR-F51). Processors in the other models have an external wave memory, e.g., the SWL01 (PSR-E443) and SWX03 (PSR-E463).

Newer DSP effects aside, the E-series models share the same basic reverb and chorus effects. There are three chorus effects:

Chorus1 (MSB: 66 LSB: 17)
Chorus2 (MSB: 65 LSB: 02)
Chorus3 (MSB: 65 LSB: 00)

The LSB has varied, but they all refer to the same CHORUS (CELESTE) effect algorithm. The LSB just selects a set of preset effect parameters. Chorus1, BTW, falls into the XG CELESTE category, not CHORUS.

Due to hardware integration, the chorus effects likely share the same hardware. Since none of these processors have external DSP RAM, the chorus memory is integrated, too.

As far as chorus is concerned, this is the way it has been since the 1990s! Let’s look back to the Yamaha QY-70 XG implementation (1995). I suspect that the current chorus effects are the same or very similar to the good old QY.

The QY-70 had one chorus and celeste effect algorithm:

Param#  Parameter            Value range 
------  -------------------  -------------------- 
   1    LFO Frequency        0.00Hz - 39.7Hz 
   2    LFO PM Depth         0 - 127 
   3    Feedback Level       -63 - +63 
   4    Delay Offset         0 - 127 
   5 
   6    EQ Low Frequency     50Hz - 2kHz 
   7    EQ Low Gain          -12dB - +12dB 
   8    EQ High Frequency    500Hz - 16.0kHz 
   9    EQ High Gain         -12dB - +12dB 
  10    Dry/Wet              D63;gt;W - D=W - D<W63 
  11    ... 
  15    Input Mode           Mono, Stereo 
  16

These parameters are laid down by the Yamaha XG specification.

The XG specification does not define the preset values, however. Here are the QY-70 preset chorus values:

Param#  Parameter            Chorus1 Chorus2 Chorus3 Chorus4 
------  -------------------  ------- ------- ------- ------- 
   1    LFO Frequency        0.25Hz  0.33Hz  0.16Hz  0.37Hz 
   2    LFO PM Depth         54      63      44      32 
   3    Feedback Level       +13     +0      +0      +5 
   4    Delay Offset         106     30      110     104

QY-70 Chorus3 has the same MSB/LSB as PSR-E Chorus2. QY-70 Chorus 1 has the same MSB/LSB as PSR-E Chorus3. Confusing? Yes, but these are probably the PSR values or close to it.

Next are the QY-70 preset celeste values:

Param#  Parameter            Celeste1 Celeste2 Celeste3 Celeste4 
------  -------------------  -------- -------- -------- -------- 
   1    LFO Frequency        0.50Hz   1.17Hz   0.16Hz   0.33Hz 
   2    LFO PM Depth         32       18       63       29 
   3    Feedback Level       +0       +26      -20      +0 
   4    Delay Offset         0        2        2        0

None of the QY-70 presets have the same MSB/LSB as PSR, so your guess is as good as mine.

Now, the really bad news. The PSR-E series, at best, is XGlite. XGlite implementations typically don’t support the XG messages that set effect parameters. Therefore, what you hear is that you get. In other words, the effect presets are hardwired.

The Yamaha PSS series with its minimal SWLL processor implements exactly one chorus and exactly one reverb preset. You get what you pay for!

PSS-A50 MIDI mod

Posted on August 23, 2021 by pj

Sometimes you get very lucky when searching the Web!

A Japanese blogger (darekasan_net) posted a review of the Yamaha PSS-A50 and a mod adding 5-pin MIDI OUT. [Google translation of the review]

There are a set of test pads in the upper left corner of the A50 main board (DM) as shown in the picture below. [Click image to enlarge it.]

Yamaha PSS-A40 MIDI signals and USB circuitry

The two larger rectangular pads (orange) are digital ground (DGND). The four smaller pads (blue) from left to right are:

MIDI_IN (RXD MIDI_IN)
MIDI_OUT (TXD MIDI_OUT)
3.3V
Digital ground

The blogger connected the MIDI_OUT signal to a 5-pin DIN connector, which is mounted nearby on the enclosure.

By the way, Yamaha conveniently mark test points with a circle (bullseye). You can see several test points for digital ground (DGND), the +3.3V digital rail, and the +5V digital rail. The USB interface chip is an NXP ARM microprocessor. The micro USB connector is in the upper right.

Direct connection is too trusting. The MIDI_IN and MIDI_OUT pads go directly to SWLL pin 55 (RXD) and SWLL pin 54 (TXD). I suggest adding a 220 ohm current limiting resistor in series with MIDI_OUT. Adding a signal buffer would be even better since you would rather blow up the buffer instead of the main processor (YMW830-V or SWLL) should someone radically misconnect the 5-pin MIDI port. A current limiting resistor on the +V MIDI pin wouldn’t hurt either.

If you get the urge to add 5-pin MIDI IN, you’ll need an optoisolator as shown in the schematic below.

Although the schematics indicate 5V, the circuits should work with 3.3V instead. Fortunately, the unpopulated test connector provides +3.3V as well as MIDI_IN and MIDI_OUT.

Here’s an idea. Instead of hacking in a 5-pin DIN connector alone, why not add a Bluetooth MIDI plug like the CME WIDI Master?

Our Japanese blogger considered adding a sustain input. The PSR-F50 dedicates SWLL pin 53 (PORTC0) to sustain. Unfortunately, the PSS-A50 has other ideas and SWLL pin 53 mutes the headphone output instead. You could put an external switch in parallel with the front panel sustain switch, but it toggles sustain and, thus, it doesn’t behave like a true piano sustain pedal.

As with any mods, make them at your own risk and kiss your warranty good-bye!

PSR-E473 speculation

Posted on August 17, 2021 by pj

Update: Yamaha PSR-E473 and PSR-EW425 announcement.

Yamaha have had a busy few years updating their entry-level models, most notably, the PSR-E373 and the DGX-670. We have yet to see the PSR-E473 and PSR-EW425 models which will replace the PSR-E463 and PSR-EW410, respectively.

No doubt, supply chain and global shipping problems have delayed product launch. Prior PSR-E4xx models employ digital-to-analog converters (DAC) including DACs from Asahi Kasei Microdevices (AKM). The AKM plant in Nobeoka city suffered a major fire on October 20, 2020. Japanese authorities just recently gave AKM permission to clean-up and rebuild. The AKM fire caused a mass shortage of its digital-to-analog and analog-to-digital converters. The shortage affects other major audio and digital musical instrument manufacturers, too, not just Yamaha.

In addition, Yamaha is ramping up production at its new OneHub Chennai (India) manufacturing plant. The Chennai plant products acoustic guitars and portable keyboards for the Indian market and for export. According to a May 2019 press release, the initial goal is to make 200,000 acoustic guitars and 150,000 portable keyboards, including the PSR-I500 and the PSS series. Yamaha eventually wants to raise the goal to 400,000 acoustic guitars and 300,000 keyboards per year. Roughly 50% of production will be for export.

My PSS-A50 was manufactured in India. Given kinship to the PSR-I500, I would expect Chennai to make the new PSR-E473 and PSR-EW425. Export data indicate that a few E473s already have been run off and exported. Prototypes? Development? Testing?

Of course, the delayed launch has intensified interest among enthusiasts. The E373 received substantial feature upgrades: Super Articulation Lite (S.Art Lite) voices and new DSP effect types which once could only be found on mid- and upper-range arrangers. A few of the new effects are top-of-the-line: dual rotary speaker, Real Distortion guitar amp effects, vintage stereo phaser, compression and harmonic enhancer. One fully expects to see the same upgrades in the E473 and EW425.

S.Art Lite voices behave somewhat differently than their mid- and upper-level cousins. A dedicated articulation button triggers the articulation effect. The cousins transparently employ software scripting which reacts to player gestures, e.g., legato, intervals, and so forth.

My first thought was “That looks quite professional,” not just a home keyboard. The live control knobs are re-located to the upper left. This decision will be controversial! The lighted buttons look pleasant (light blue color) and the screen is black and white monochrome. Still only four registrations per bank.

Yamaha did majorly swizzle around a bunch of front panel controls with respect to the E463. The keypad to the right has been significantly redesigned.

The quadrant to the right of the display has a 3×4 button matrix for voice and style selection by category. The buttons above the data wheel control selection mode: voice or style. I wonder if one of the mode buttons turns the matrix into a numeric keypad? The FUNCTION and PORTABLE GRAND buttons are below the matrix along with some kind of BOOST button.

The control groups running above the keyboard are (left to right): master volume, SONG/STYLE control, TRACK control, registration memory, and quick sampling. The large light blue button between the volume knob and the SONG/STYLE control group may be the ARTICULATION button.

The Quick Sampling feature got more real estate. Quick Sampling has several buttons: LOOP HOLD, A, B, C, D, and CAPTURE. I wonder if it’s possible to capture four waveforms? Did Yamaha re-think sample control including sample zones? Do the lighted A, B, C, D buttons reflect sample status like a pad controller? Can we play the pads? Are they velocity sensitive?

The live control knobs are further away from the keyboard in the quadrant to the left of the display. I can’t tell if there is an additional row or not. E463 has five live control rows; EW425 has 6 or 7?

The rest of the buttons in the upper left quadrant must be record, metronome, tap tempo, melody suppressor, voice control and all that miscellaneous stuff. The legends in the picture are too distorted to read.

Yamaha is still using an LCD with pre-defined, fixed icons and legends. Do they really save that much money versus a full graphic, pixel addressable display? What do I know? It probably simplifies the software, but it seems so 90s.

Should be interesting finally to see the specs, and not just guess.

Review: Mystic Circuits 0HP Envelope

Posted on August 4, 2021 by pj

As promised, here is a review of the Mystic Circuits 0HP Envelope module. I won’t go through build details, etc., since I discussed these aspects of the 0HP line in my review of the Mystic Circuits 0HP 0ttenuator.

Mystic Circuits 0HP Envelope and 0ttenuator modules

The 0HP modules all share the same micro form factor. The 0HP Envelope has four jacks:

IN: Gate or audio input
MULT: Duplicates the IN signal for patches
TIME: Release time control voltage (CV) input
OUT: Envelope output

Given a gate signal, the 0HP Envelope module generates a simple envelope. Given an audio signal, the Envelope module is an envelope follower, i.e., it generates an envelope which tracks the audio amplitude.

Schematic

Mystic Circuits have not posted a schematic for the 0HP Envelope. So, I drew one up. [Click images to enlarge.]

The 1uF capacitor is key to understanding the Envelope’s operation. The IN signal charges the 1uF capacitor while the phototransistor in the optoisolator (KTV816) discharges the capacitor to ground. The charge on the capacitor determines the OUT voltage. If you put audio into the Envelope, the capacitor smooths out the audio, leaving only the amplitude envelope.

The (release) TIME signal controls the brightness of the LED in the optoisolator. The brightness then controls the gate of the phototransistor. When the LED is brighter, the gate turns ON harder, more current flows, and the capacitor discharges faster.

Initial use

I had hoped to use the 0HP Envelope as an envelope follower such that I could process audio through the littleBits Filter module. Short story: A minor fail, but not the fault of the 0HP Envelope.

I connected the output of a Yamaha PSS-A50 to the audio input of the littleBits Filter. I sent the audio — in parallel — to the IN jack of the 0HP Envelope. No joy, or at least, not much happiness. I could not discern an audible difference in the filter cut-off sweep.

This minor failure motivated a few quick experiments to better understand the 0HP Envelope.

Oscilloscope: Audio envelope follower

When puzzled, measure!

Keeping the PSS-A50 as my audio source, I monitored the incoming audio signal and outgoing envelope using a Gabotronics USB oscilloscope.

The 0HP Envelope is, essentially, a passive device. It doesn’t require power and it doesn’t have any active electronics to amplify the incoming audio (or gate) signal. The phototransistor in the optoisolator acts like a variable resistor. Thus, what comes out of the envelope depends on what goes into it.

I immediately had to crank the oscilloscope gain to compensate for the small peak-to-peak audio signal going into the 0HP Envelope. The audio signal from the PSS-A50 does not have much voltage swing. The incoming signal limits the sweep of the outgoing envelope signal.

The oscilloscope traces above are a short piano note and the envelope produced by the 0HP Envelope module. Please note the voltage per grid unit; it’s very small. The small peak-to-peak audio signal does not produce a very large envelope voltage swing. The small swing was probably not enough to produce an audible filter cut-off sweep in the littleBits case.

The oscilloscope traces below are an oboe note and its envelope.

The traces look quite noisy thanks to the high oscilloscope gain. I suspect that some noise is to due to the crummy USB ground from the PC.

Oscilloscope: Gated envelope

To verify my hypothesis concerning signal level in affecting signal level out, I connected the Arturia Keystep Gate output to the 0HP Envelope module and monitored both the gate and envelope signals with the oscilloscope.

As expected, I needed to adjust the oscilloscope gain down to accommodate the relatively higher gate voltage. Given a strong gate signal, the envelope swing is much wider as shown in the oscilloscope traces above.

The 0HP Envelope stays high while gate is asserted. The example below shows the envelope produced by a longer note. When the TIME input is unused (0 Volts), the release time is rather long. You will most likely need to adjust the release time in practice unless you want a long release!

Envelope (bottom) stays high while gate (top) is high

Summary

Overall, the 0HP Envelope does what Mystic says. It’s a reasonable envelope generator (or follower) as long as you understand its behavior: big signal in, big signal out; small signal in, small signal out.

Here are further notes taken from the Mystic Circuits video about the 0HP Envelope.

When driven with a Gate, the Envelope will stay ON while the Gate is high. Then it will slowly release to zero when the Gate goes low.
Release time is controlled by the RELEASE control voltage (CV) input. With no control voltage going into RELEASE, the Envelope release time is at its longest. Release time decreases as a positive CV is applied. When driven with audio, the Envelope will give you a voltage which is proportional to the amplitude of the incoming audio. Sensitivity is controlled by the RELEASE CV input. When driven with a variable voltage, the Envelope will glide when the input voltage falls. It will increase quickly when the input voltage rises. Glide time is controlled by the RELEASE CV input. This only works on positive voltages.
The Envelope module has an on-board MULT to the input which allows chaining of multiple units. This is useful when you want to use an active envelope generator to produce more complex shapes and use the 0HP to produce simpler shapes. You can also route incoming audio back into your patch.

Review: Mystic Circuits 0ttenuator

Posted on August 2, 2021 by pj

I discovered Mystic Circuits while browsing Patchwerks. Mystic Circuits make and distribute a line of analog synthesis modules including its “0HP” micro-modules. Time to take a look.

But, first. Patchwerks? Patchwerks was a great little find, too, and a synthesizer lifeline during the pandemic. Patchwerks has a small Seattle-based brick-and-mortar retail store as well as its Web store. I have yet to step into their physical store, but I have ordered a number of small boards and toys on-line. Each time, their fulfillment has been fantastic: good packaging, same day shipping and quite frequently, over-night delivery, thanks to our Seattle metro locations. Watch for their seasonal sales. Highly recommended!

The 0H modules get their name because they don’t take up space in your modular rack. Each module implements one or more utility and synthesis functions that don’t require a rack slot. Just patch ’em in. Mystic Circuits offer 0HP modules both fully assembled and in kit form. Initially, I was searching for an envelope follower block and in the course of that search, I discovered the entire 0HP product line. I bought the 0HP Envelope module (fully assembled $36USD) and the 0ttenuator module (kit form $18USD). I focus on the 0ttentuator in today’s post.

Mystic Circuits 0HP 0ttenuator

The 0HP 0ttenuator is one of those modules that you can’t live without. It performs four functions:

Single passive signal attenuator
Dual, independent passive signal attenuators
Two input passive mixer
One-to-two signal splitter

I can’t count the number of times when I needed a simple signal attenuator (e.g., knocking a headphone level down to LINE), or a 2-input mono mixer. The 0ttenuator does the job and then some.

Even though Mystic Circuits call it the “simplest kit,” I wouldn’t recommend it for beginners. The resistor pads are really dinky and it would be easy to make a solder bridge to other, larger pads. You need a really good soldering tip to nail it. I suggest checking your work with a magnifying glass. Further, Mystic don’t identify the resistor values (or color codes). The resistors are so small, I can’t accurately read the color bars! Whip out a digital meter and the resistors measure as 22K ohms. The potentiometers (B104) are 100K ohms.

If I have to ding Mytic Circuits, I find their documentation to be thin and sketchy. Useful information is buried in video, including build and operating instructions. Nothing is written down. Frankly, I don’t have time to watch a video when I can read bullet points in a few seconds. Although Mystic provide Eagle “sch” files on their github site, most people aren’t set up to display Eagle. I ran the sch file through schematics.io and captured the rendering (below).

Hey, I’ve seen this circuit somewhere before! Nonetheless, it’s a very flexible design and the 0HP module is well-made. Operation depends upon the switched input jacks which configure the circuit for attenuation, mixing and splitting.

Mystic Circuits 0HP case top and bottom (assembled)

I’m glad that I purchased a fully assembled Envelope module along with the 0ttentuator kit. The fully assembled module showed me how to assemble the case. Again, the Mystic Circuits site does not have instructions for assembling the new 0HP cases. The video build instructions show the old plexiglass covers and spacers, not the new PCB-ish case. That’s the drawback of putting everything in videos; if the product changes, it’s a pain to revise the existing videos to reflect product changes.

The final niggle has to do with the metallic graphic design on the case bottom. I fear it will make unwanted contact with the bottom of the printed circuit board (PCB). I put a thin layer of electrical tape over the inner surface of the bottom case cover. (The design is etched onto both sides of the case bottom.)

If you buy a 0HP 0ttentuator module — and I recommend it — here are the missing operating instructions:

To use one attenuator:
1. Plug incoming signal into IN jack
2. Plug outgoing signal into OUT jack
3. Turn knob to change signal level
To send an incoming signal to two places:
1. Plug incoming signal into PAN jack
2. Plug one outgoing lead into the OUT jack
3. Plug the other outgoing lead into the MIX jack
4. Use knobs to set outgoing levels
To mix two signals together:
1. Plug one incoming signal into the IN jack
2. Plug the other incoming signal into the PAN jack
3. Plug the outgoing lead into the MIX jack
4. Use knobs to set the outgoing level
To use each attenuator separately:
1. Plug first signal into the IN jack and the corresponding output lead into the OUT jack
2. Plug second signal into the PAN jack and the corresponding output lead into the MIX jack
3. Use knobs to set levels for each separate signal

Mystic Circuits 0HP Envelope

I want to test the 0HP Envelope with a littleBits Filter. Given time constraints, I’ll address that subject in a future blog. In the meantime, here is a little more information about the 0HP Envelope.

Here is a description of the 0HP Envelope, paraphrased from the Mystic Circuits video.

When driven with a Gate, the Envelope will stay ON while the Gate is high. Then it will slowly release to zero when the Gate goes low. Release time is controlled by the RELEASE control voltage (CV) input. With no control voltage going into RELEASE, the Envelope release time is at its longest. Release time decreases as a positive CV is applied.
When driven with audio, the Envelope will give you a voltage which is proportional to the amplitude of the incoming audio. Sensitivity is controlled by the RELEASE CV input. When driven with a variable voltage, the Envelope will glide when the input voltage falls. It will increase quickly when the input voltage rises. Glide time is controlled by the RELEASE CV input. This only works on positive voltages.
The Envelope module has an on-board MULT to the input which allows chaining of multiple units. This is useful when you want to use an active envelope generator to produce more complex shapes and use the 0HP to produce simpler shapes. You can also route incoming audio back into your patch.

Unfortunately, the Mystic Circuits github area does not have an Eagle schematic. So, I drew one (shown below). Hope I got it right!

The 1uF capacitor is key to understanding the Envelope’s operation. The IN signal charges the 1uF capacitor while the phototransistor in the optoisolator (LTV816) discharges the capacitor to ground. The charge on the capacitor determines the OUT voltage. If you put audio into the Envelope, the capacitor smooths out the audio, leaving only the amplitude envelope.

The RELEASE signal controls the brightness of the LED in the optoisolator. The brightness controls the gate of the phototransistor. When the LED is brighter, the gate turns ON harder, more current flows, and the capacitor discharges faster.

Yamaha PSS-A50: Look inside

Posted on July 27, 2021 by pj

Let’s take a quick tour of the Yamaha PSS-A50.

Yamaha PSS-A50 top and bottom [Click images to enlarge]

The A50 has two main boards: the digital and analog electronics board (DM) and the front panel board (PN). After removing nine screws — don’t forget the screw hidden in the battery compartment — the A50 splays into two halves: the bottom half containing the battery compartment, DM board and keybed, and the upper half containing the speaker and PN board. The battery connects to a JST XH connector on the DM board. Ribbon cables connect the keybed and the panel board to the DM board.

The PN board has traces for the front panel buttons. The buttons are arranged into a 3 by 8 switch matrix: 3 drive lines and 8 sense lines. The power Standby/ON switch has two dedicated lines. The eight sense lines are shared with the three digit LED display. A further 3 lines are devoted to the display (for a total of eight lines). In addition to the front panel switch matrix, the PN board conducts audio signals to the speaker through two wide PCB traces.

I dare to say that the A50, PSS-E30 Remie and PSS-F30 have the same panel board. Only the front panel graphics and software differentiate the models in that regard.

Yamaha PSS-A50 main electronics board (DM)

The DM electronics board is tiny and is packed with surface mount (SMT) components. Impressive! The main digital components are:

Yamaha YMW830-V: Processor and tone generator (IC101)
Winbond 25Q16JVS1M: 16Mbit Serial flash memory (IC102)
74VHC273: 8-bit latch for display data (IC301)
NXP LPC11U13F/201: USB interface (IC401)

The YMW830-V is also known as “SWLL” and is a Yamaha proprietary system on a chip (SOC). The A50 has separate amplifiers for the speaker (IC701) and headphone output (IC601):

TI TPA6132A2RTER: Headphone amplifier (IC601)
Rohm BD27400GUL: Mono class-D power amplifier (IC701)
NJR NJM2740M: Dual operational amplifier (IC501)

The dual operational amplifier is part of the post-DAC low pass filter. Finally, the power-related components are:

TI TLV74333PDBVR: 3.3V regulator (IC001)
TI TPS63060DSCR: Switching regulator (IC004)
TI TPS25200DRVR: 5V eFuse/power switch (IC006)

The A50 must choose and switch between +5V USB power and battery power. That’s the role of the eFuse/power switch component.

Yamaha PSS-A50 USB interface (NXP ARM MCU)

The NXP LPC11U13F is a bit of a surprise to me. It is an ARM Cortex-M0 32-bit microprocessor (MCU) with 24KB of flash memory. The SWLL sends and receives MIDI through its UART RX/TX ports. The ARM LPC converts simple MIDI from the SWLL to MIDI over USB. Using an ARM MCU to do the job seems like over-kill. It goes to show how far we have come as an industry when an MCU can be dedicated to such a mundane task!

Yamaha PSS-A50 CPU (Yamaha YMW830-V SWLL)

The SWLL (YMW830-V) has many of the specs that we’ve come to know about Yamaha’s entry-level CPUs. The external crystal resonates at 16.9344MHz. The SWLL internal clock is 33.8688MHz and generates a 67.7376MHz master clock. If these numbers look odd to you, simply note that they are even multiples of 44,100Hz, the basic sample rate:

    67.7376MHz = 44,100Hz * 1,536

When an external DAC is used, the master clock provides the bit serial audio clock. 1,536 can be subdivided in all sorts of interesting ways depending upon sample word length.

The SWLL integrates host CPU, memory, tone generation, serial MIDI communication, keyboard and front panel scan ports, and display ports. The digital to analog converter (DAC) is also integrated into the SWLL. The SWLL is truly Yamaha’s low-cost system on a chip solution.

The SWLL loads its software and samples from a 16Mbit serial flash ROM. 2MBytes for software and samples is not much, so one wonders if the SWLL has a preprogrammed flash memory of its own?

With the exception of the ARM LPC chip, the A50, PSS-E30 Remie and PSS-F30 electronics are identical. The software and samples determine the product personality. Such a high degree of commonality allows Yamaha to manufacture PSS keyboards (in India) and sell them at a dirt cheap price. Hats off — the amount of technology at this price — less than $100USD — is simply astounding.

PSS-A50: Power to the people

Posted on July 22, 2021 by pj

Today’s topic — power — may seem rather mundane. To a modder, though, power gives our circuits life.

I’m going to make a few comments of general interest before diving into details that are relevant to the Yamaha PSS series keyboards, including the PSS-A50 and PSS-E30 Remie.

Most of us don’t think too much about keyboard power. Sure, we know where the AC adapter connects or how to insert batteries. The internal details are hidden from us.

However, did you really read the fine print in the Owner’s Manual? The front panel power button may be labelled “Standby/ON” instead of “OFF/ON”, and the difference is important. The PSS-A50 Owner’s Manual states, “Even when the Standby/On switch is in standby status (display is off), electricity is still flowing to the instrument at the minimum level.”

Yes, that Standby/ON switch is really a “soft” power switch. It does not physically disrupt the flow of electrical current from the AC adapter (battery or USB port). In the PSS series (and other keyboards, too), the Standby/ON switch sends a signal to the keyboard’s processor telling the software to change the current power state. For the technically inclined, the Standby/ON switch pulls one of the processor pins to ground and software detects the ACTIVE LOW signal.

The rest of the story gets complicated fast depending upon power saving techniques supported by the hardware. Let’s assume that we’re changing from ON to Standby. The processor generates a separate signal which switches off the power amplifier — a major drain on battery or external power. Software turns off the display, another power hog. Finally, software places the processor in a low-power state and waits for the Standby/ON switch to be pressed again. Going from Standby to ON, software turns everything back on.

From the user’s perspective, the transition from Standby to ON is fast. No waiting and let’s get playing! The constant low current flow does affect battery life, however. Ever wonder why the batteries drained sooner than expected even though you haven’t turned your keyboard on for a few weeks? The low current flow eventually drains the batteries.

Power management has implications for people intending to mod an instrument. I’m planning to add an audio delay or filter circuit to the A50. The add-on circuit will need to draw power. Ideally, I would like to switch the add-on circuit on and off with the front panel switch. But, where should I take power from the existing design? Is there a PCB pad or trace that is big enough for soldering? Is voltage regulated at that point? Getting power is not a no-brainer!

If you don’t have the instrument’s service manual and schematic, this analysis gets really hairy and uncertain. For the E30/A50, I’ve been working from the PSR-F50 manual available from Elektrotanya. The PSS series keyboards are a revamped PSR-F50 design.

I’m considering a Synthrotek Dev Delay for add-on. The Dev Delay has a 5V regulator and runs on battery power. My thought is to connect the Dev Delay directly to the A50’s batteries through its own power on/off switch. That way I don’t add to the standby drain on the batteries. It just means turning the delay on and off separately.

PSS-E30 Remie main board (battery connector at right)

Even better, the A50 main board (DM) has a removable battery connector. If I rustle up a compatible cable and connectors, I can tap into existing battery power without soldering. I was already planning to use a short 3.5mm patch cable to jump the headphone OUT to the Dev Delay IN. Again, no soldering to SMT traces, etc. I like “reversible” mods!

I had enough headaches and scars from soldering mod chips to game console boards back in the day. 🙂

I hope this discussion provided some useful advice — no matter what you mod.

Sand, software and sound

Electronics and computing for the fun of it