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Silent Way for JACK/Linux

The Expert Sleepers Eurorack products allow digital audio outputs from computers to be injected as analog control voltages and gates into Eurorack synthesizers. Several of their products allow many digital signals to be multiplexed onto a smaller number of S/PDIF or ADAT digital audio connections. On MacOS and Windows machines, there's a vendor-provided driver that manages the multiplexing. Not so for Linux.

Luckily the nice folks at Expert Sleepers pointed me to the code of a Pure Data external implementing the multiplexing: https://github.com/expertsleepersltd/externals

Based on the algorithms in that code, I'm implementing a standalone JACK client to sit between your signal-generating software and the audio interface that's connected to the Expert Sleepers device(s).

Note: It's not ready yet

I doubt that anyone will come across this repo for a while, but in case you do: I'll remove this text when I have something working. I'm still implementing, fleshing out this document, and just pushing to Github in case I spill coffee on my laptop.

Installation

This code uses "waf" as its build tool.

$ ./waf configure build install

I mostly develop on amd64 Debian Linux, so there are probably some implicit assumptions about that.

For OSX users, I have at least been able to waf build with the following packages brew-installed:

$ brew install jack glib argp-standalone

Run-time Configuration

The silentway client looks for a configuration file, called ~/.silentway by default (a different file can be used via command-line arguments). The config file defines what Expert Sleepers hardware you have and the way it is connected, so that the application's audio and MIDI ports can be set up correctly.

[silentway]             # global options
interface=es-5          # valid values: es-4, es-5, es-40

[GT1]                   # first expansion port 
device=esx-8cv          # valid values: esx-8gt, esx-8cv, esx-4cv, esx-8md 
port_0=signal           # default is an audio input converted to CV
port_1=midi(1, cc1)     # use midi channel 1 CC1 as the signal source 
port_2=midi(2, noteNum) # use midi channel 2 NoteOn note numbers
port_3=midi(2, noteOn)  # use midi channel 2 NoteOn as a gate 

Signal config options

The range of the digital audio outputs are (-1, 1), with Expert Sleepers hardware scaling this range to (-10V, 10V).

Ports that are of type signal can (optionally) specify a simple linear calibration adjustment as (offset, scale).

   sig_out = (sig_in + offset) + scale

So, for example, if you want the signal range (-1,1) to be mapped onto a control voltage from 0 to 5V, you would specify

port_0=signal(1.0, 0.25)

MIDI config options

MIDI inputs can be mapped to either CV voltages or gates. The (required) parameters are provided as (channel, source, transpose*, offset*, scale*). Parameters marked with * may be omitted if there are no following parameters with values.

Values for source:

  • CCs: cc1, cc2,
  • Note number: noteNum (convert note number to 12TET CV)
  • Note trigger: noteOn (convert note on to trigger)
  • Note gate: noteOnOff (convert note on/off to gate)

Silent Way overview

Here's what I have figured out about the Silent Way approach based on my reading of the source code and the Expert Sleepers documentation.

Signal acquisition

Digital audio enters the Expert Sleepers domain via either ADAT (ES-3) or S/PDIF (ES-4, ES-40).

Audio channels can be brought out directly to the front panel of an ES-3; those require no special software so I'm not concerned with them. An ES-4/ES-40 S/PDIF interface, or ES-3 channels sent to an ES-5 expansion interface treat the audio differently.

A single stereo pair of digital signals (either a stereo 20-bit S/PDIF connection or 2 channels of a 24-bit ADAT connection) is treated as a 40- or 48-bit-wide data stream running at the specified interface rate (generally 44.1 kHz or 48 kHz).

Stream splitting

The 40- or 48-bit stream is then split into 5 or 6 8-bit streams at audio rate. Each of these streams appears on the PCB of the device (ES-5, ES-4, ES-40) as a 10-pin header marked "GT1", "GT2", etc.

For the 48-bit stream from an ES-3 into an ES-5 expansion connector, each 24-bit word is split into 3 8-bit chunks. Channels 1, 2, and 3 are on the left channel, while 4, 5, and 6 are on the right.

For the 40-bit stream from an ES-4 or ES-40, channels 1 and 2 are unpacked from the highest 16 bits of the left input channel, channels 3 and 4 are unpacked from the highest 16 bits of the right input channel, and channel 5 is assembled from the lowest 4 bits of each (lowest 4 bits of left channel becomes highest 4 bits of channel 5, lowest 4 bits of right becomes lowest 4 bits of 5).

Expansion devices

The 8-bit expansion connectors can go to one of several "ESX" devices to produce gate or CV outputs:

  • ESX-8GT just directly outputs the 8 bits as 8 gate/trigger signals
  • ESX-4CV uses an encoding scheme to multiplex 4 12-bit CV signals at 1/8 the sample rate onto one 8-bit connection
  • ESX-8CV uses a different encoding scheme to multiplex a variable number of 12-bit CV signals (1-8) onto one 8-bit connection at a sample rate dependent on the number of connections
  • ESX-8MD breaks out 8 1-bit serial MIDI data streams just like an ESX-8GT but with different connectors and different software

Encodings

Audio to expansion ports

ES-3 --> ES-5: Each 24-bit digital audio signal is split into 3 8-bit segments.

Bitmask (L R)       Channel 
0xff0000 0x000000   1
0x00ff00 0x000000   2
0x0000ff 0x000000   3
0x000000 0xff0000   4
0x000000 0x00ff00   5
0x000000 0x0000ff   6

ES-4/ES-40: Five channels are split across 40 bits (S/PDIF values arrive as 24-bit words in most systems, with the lower 4 bits always 0):

Bitmask (L R)       Channel 
0xff0000 0x000000   1
0x00ff00 0x000000   2
0x000000 0xff0000   3
0x000000 0x00ff00   4
0x0000f0 0x0000f0   5

Expansion port to ESX-8GT

The 8 bits of the expansion port signal are connected directly to the 8 gate outputs. Each word of digital output immediately affects the gate outputs, so they can change at audio rate (44.1 kHz/48 kHz)

Bit   1  2  3  4  5  6  7  8
Gate [1][2][3][4][5][6][7][8]

Expansion port to ESX-8CV

The 12 bits of each CV signal are encoded into 3 8-bit expansion port words. You may notice that this is using 24 bits to encode 12 bits of data, which seems a bit inefficient. Also included is the destination CV port number and some sync data to make the connection robust and allow for the best possible sample rate for the number of desired signals (from r/3 to r/24 depending on the number of connected signals)

Word 1:

Bit   Value  Meaning 
0x80     1   This is the first word
0x40     0   Not the second word
0x20     0   Or the third word
0x10    D5   
0x08    D4
0x04    D3
0x02    D2
0x01    D1 

Word 2:

0x80     0   Not the first word
0x40     1   This is the second word
0x20     0   Not the third word
0x10   D10   
0x08    D9
0x04    D8
0x02    D7
0x01    D6 

Word 3:

0x80     0   Not the first word
0x40     0   Not the second word
0x20     1   This is the third word
0x10    A2   CV address bit 2
0x08    A1   CV address bit 1
0x04    A0   CV address bit 0
0x02   D12
0x01   D11

I believe that the CV outputs arei each basically just a sample-and-hold with a lowpass filter after it. Each data point addressed to a certain CV output is immediately available on that output once the frame containing it is written. This allows for the interesting property of the variable sample rate of the output CVs, but it also means that this is probably not going to be a very high-fidelity output for signals changing at more than a few Hz.

Expansion port to ESX-4CV

I'm not sure of the history, but I believe this was the first iteration of representing a 12-bit value on an 8-bit channel. The advertising material says that the sample rate of each port a 1/8th the audio rate, meaning that it takes 8 8-bit words to represent 4 12-bit samples. That means 16 bits (2 words) for each 12-bit value, with 4 extras: 2 bits for sync and 2 bits for DAC address. Thanks to Expert Sleepers for confirming this bit mapping in the support forum:

Word 1

Bit   Value  Meaning 
0x80     1   This is the first word
0x40    A1   CV address bit 1 
0x20    A0   CV address bit 0 
0x10   D12   (data >> 7)
0x08   D11
0x04   D10
0x02    D9
0x01    D8

Word 2

Bit   Value  Meaning 
0x80     0   This is the second word
0x40    D7   (data & 0x7f) 
0x20    D6   
0x10    D5   
0x08    D4
0x04    D3
0x02    D2
0x01    D1 

I would guess that, like the ESX-8CV, frames are requested but not required to cycle through all 4 CV addresses in a cycle.

Expansion port to ESX-8MD

Each of the 8 MIDI DIN connectors on the ESX-8MD has just one data bit; that's the nature of MIDI's serial protocol. So from an electrical and signal routing perspective, the ESX-8MD is basically the same thing as the ESX-8GT, just with bigger connectors.

The software is the biggest difference. MIDI is a serial protocol operating at 31250 bits per second, whereas the audio-rate signals entering the ESX-8MD are 44100 or 48000 bits per second (for each port) and the MIDI device drivers at the operating system level treat it as a stream of 8-bit bytes going over a channel that's about 3100 bytes per second. Reconciling this mishmash is what the Silent Way software does.

I've never seen the source code to the SW software, so I'm having to guess at how they make all this work, but it's not a huge leap. Basically we rely on the receiving MIDI device following the spec and just bit bang our signal to meet its timing expectations.

The key thing to note is that most serial devices use a word-oriented protocol with a fixed "idle line" condition, followed by a "start bit", followed by data and sometimes a "stop bit". MIDI is no different. The line is normally held in the "high" or "1" condition, then the start of a word (and the start of the timer for reading each bit) is indicated by the transition to the "0" start bit state. Every 32 microseconds after that the line is read to get the next bit.

So to fool a genuine MIDI device into reading our data, we just have to turn a stream of data bytes into a series of 320 microsecond signals, where 10 bits of data (start + 8 payload + stop) are turned into 15 samples (at 44.1 kHz) or 16 samples (at 48 kHz) by strategically repeating a bit to stretch out the timing to match the receiver's expectation.

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