JASS, Just Another Synth...Sort-of, codename: Gemini (UPDATED TO V-1.0.1)

jass-v1.0.1( esp with midi fixes).zip

1.0.1-CHANGES:

• Fixed issues with midi routing, re the mode selector (mentioned below)
• Upgraded the midi mode "fetch" abstraction to be less granular
• Fix (for midi) so changing cc["14","15","16"] to "rnd" outputs a random wave (It has always done this for non-midi.)
• Added a midi-mode-tester.pd (connect PD's midi out to PD's midi in to use it)
• Upgrade: cc-56 and cc-58 can now change pbend-cc and mod-cc in all modes
• Update: the (this) readme

INFO: Values setting to 0 on initial cc changes is (given midi) to be expected.

JASS is a clone-based, three wavetable, 16 voice polyphonic, Dual-channel synth.

With...

• The initial, two wavetables combined in 1 of 5 possible ways per channel and then adding those two channels. Example: additive+frequency modulation, phase+pulse-modulation, pulse-modulation+amplitude modulation, fm+fm, etc
• The third wavetable is a ring modulator, embedded inside each mod type
• 8 wave types, including a random with a settable number of partials and a square with a settable dutycycle
• A vcf~ filter embedded inside each modulation type
• The attack-decay-release, cutoff, and resonance ranges settable so they immediately and globally recalculate all relevant values
• Four parameters /mod type: p1,p2, cutoff, and resonance
• State-saving, at both the global level (wavetables, env, etc.), as well as, multiple "substates" of for-each-mod-type settings.
• Distortion, reverb
• Midiin, paying special attention to the use of 8-knob, usb, midi controllers (see below for details)
• zexy-limiters, for each channel, after the distortion, and just before dac~

Instructions

Requires: zexy

for-entire-state

• O: Open preset. "default.txt" is loaded by...default
• S: Save preset (all values incl. the multiple substates) (Note: I have Not included any presets, besides the default with 5 substates.)
• SA: Save as
• TEST: A sample player
• symbol: The filename of the currently loaded preset
• CL: Clear, sets all but a few values to 0
• U: Undo CL
• distortion,reverb,MASTER: operate on the total out, just before the limiter.
• MIDI (Each selection corresponds to a pgmin, 123,124,125,126,127, respectively, see below for more information)
  • X: Default midi config, cc[1,7,8-64] available
  • M: Modulators;cc[10-17] routed to ch1&ch2: p1,p2,cutoff,q controls
  • E: Envelopes; cc[10-17] routed to filter- and amp-env controls
  • R: Ranges; cc[10-17] routed to adr-min/max,cut-off min/max, resonance min/max, distortion, and reverb
  • O: Other; cc[10-17] routed to rngmod controls, 3 wavetypes, and crossfade
  • symbol: you may enter 8 cc#'s here to replace the default [10-17] from above to suit your midi-controller's knob configuration; these settings are saved to file upon entry
• vu: for total out to dac~

for-all-mod-types

• /wavetable
  • graph: of the chosen wavetype
  • part: partials, # of partials to use for the "rn" wavetype; the resulting, random sinesum is saved with the preset
  • duty: dutycycle for the "du" wavetype
  • type: sin | square | triangle | saw | random | duty | pink (pink-noise: a random sinesum with 128 partials, it is not saved with the preset) | noise (a random sinesum with 2051 partials, also not saved)
• filter-env: (self-explanatory)
• amp-env: (self-explanatory)
• rngmod: self-explanatory, except "sign" is to the modulated signal just before going into the vcf~
• adr-range: min,max[0-10000]; changing these values immediately recalculates all values for the filter- and amp-env's scaled to the new range
• R: randomizes all for-all-mod-types values, but excludes wavetype "noise"; rem: you must S or SA the preset to save the results
• U: Undoes R

for-each-mod-type

• mod-type-1: (In all cases, wavetable1 is the carrier and wavetable2 is the modulator); additive | frequency | phase | pulse | amplitude modulation
• mod-type-2: Same as above; mod-type-2 May be the same type as mod-type-1
• crossfade: Between ch1 and ch2
• detune: Applied to the midi pitch going into ch2
• for-each-clone-type controls:
  • p1,p2: (self-explanatory)
  • cutoff, resonance: (self-explanatory)
• navigation: Cycles through the saved substates of for-each-mod-type settings (note: they are lines on the end of a [text])
• CP: Copy the current settings, ie. add a line to the end of the [text] identical to the current substate
• -: Delete the current substate
• R: Randomize all (but only a few) substate settings
• U: Undo R
• cut-rng: min,max[0-20000] As adr-range above, this immediately recalculates all cutoff values
• res-rng: min,max[0-100], same as previously but for q
• pbend: cc,rng: the pitchwheel may be assigned to a control by setting this to a value >7 (see midi table below for possibilities); rng is in midi pitches (+/- the value you enter)
• mod-cc: the mod-wheel may be assigned to a control [7..64] by setting this value

midi-implementation

cc - basic (for all midi-configs)

cc - "X" mode/pgmin=123

cc - Modes M, E, R, O

Jass is designed so that single knobs may be used for multiple purposes without reentering the previous value when you turn the knob, esp. as it pertains to, 8-knob controllers.

Thus, for instance, when in Mode M(pgm=124) your cc send the signals as listed below. When you switch modes, that knob will then change the values for That mode.

In order to do this, you must turn the knob until it hits the previously stored value for that mode-knob.

After hitting that previous value, it will begin to change the current value.

cc - Modes M, E, R, O assignments

Where [10..17] may be the midi cc #'s you enter in the MIDI symbol field (as mentioned above) aligned to your particular midi controller.

In closing

If you have anywhere close to as much fun (using, experimenting with, trying out, etc.) this patch, as I had making it, I will consider it a success.

For while an arduous learning curve (the first synth I ever built), it has been an Enormous pleasure to listen to as I worked on it. Getting better and better sounding at each pass.

Rather, than say to much, I will say this:

Enjoy. May it bring a smile to your face.

Peace through love of creating and sharing.

Sincerely,
Scott

mda Delay.vst.zip

[vstplugin~] v0.2.0
WARNING: on macOS, the VST GUI must run on the audio thread - use with care!
searching in '/Users/boonier/Library/Audio/Plug-Ins/VST' ...
probing '/Users/boonier/Library/Audio/Plug-Ins/VST/BreakBeatCutter.vst'... failed!
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probing '/Library/Audio/Plug-Ins/VST/VCV-Bridge.vst'... failed!
probing '/Library/Audio/Plug-Ins/VST/WaveShell1-VST 10.0.vst'... failed!
found 0 plugins
searching in '/Users/boonier/Library/Audio/Plug-Ins/VST3' ...
found 0 plugins
searching in '/Library/Audio/Plug-Ins/VST3' ...
probing '/Library/Audio/Plug-Ins/VST3/TX16Wx.vst3'... error
couldn't init module

probing '/Library/Audio/Plug-Ins/VST3/WaveShell1-VST3 10.0.vst3'... error
factory doesn't have any plugin(s)

probing '/Library/Audio/Plug-Ins/VST3/Nektarine.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/Nektarine_32OUT.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/OP-X PRO-II.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/SPAN.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/Surge.vst3'... failed!
probing '/Library/Audio/Plug-Ins/VST3/Zebra2.vst3'...
[1/4] 'Zebrify' ... failed!
[2/4] 'ZRev' ... failed!
[3/4] 'Zebra2' ... failed!
[4/4] 'Zebralette' ... failed!
found 0 plugins
search done
print: search_done

edit: solved

Hi Folks,

I thought I’d share this patch in the hopes that someone might be able to help improve upon it. I am by no means even semi competent with PD and jumped into this task without actually bothering to learn the basics of PD or RPi, but nevertheless here we are: maybe you can share a better implementation.

Mods/experienced folks, if I am sharing irrelevant/wrong/confusing info, mea culpa and please correct me.

I wanted to make a patch for PD in Raspberry Pi that would do 3 things:

• Get the abl_link~ temp data over wifi
• Create a midi clock output using a 5-pin midi adapter (I have one of the cheapo usb-to-midi cable things here)
  -simultaneously create an audio pulse ‘clock’ output such as those used by volcas, Teenage Engineering Pocket operators, and the like (I am not sure if such an audio signal over a 3.5mm jack would be hot enough to be considered a CV pulse too, maybe you can help clear that up?)

As I say, after much struggles I have globbed something together that sort of does this.

A couple of things for newcomers like myself:

The abl_link~ object in the patch isn’t initially part of the standard pure data install as I write. I was able to use deken (ie the code that powers the ‘help/find externals’ bit of PD) to look for abl_link~. Search for it. At the time of writing there is a version for Arm7 devices like the Raspberry Pi 3 which was put together by the illustrious mzero with code from antlr. Go ahead and install the abl_link~ object. (Possibly you may have to uncheck the ‘hide foreign architectures’ box to get the arm7 version to show up. This is usually a safeguard to stop users from trying to install versions of externals that won’t work on their systems. So long as you see ‘arm7’ in the description it should hopefully be the one you want) PD will ask where you want to store the external, and I would just leave it at the default unless you have a special reason to do otherwise.

To get the patch to hook up to your preferred audio and midi outputs by default you may have to take certain steps. In my version of it I have deemed the built in audio and my cheapo USB midi output to be good enough for this task.

[As part of my troubleshooting process I ended up installing amidiauto which is linked to here: https://community.blokas.io/t/script-for-launching-pd-patch-with-midi-without-aconnect/1010/2
I undertook several installations in support of amidiauto which may be helping my system to see and link up my USB midi and PD, but nothing worked until I took the step in the following paragraph about startup flags in PD. (It may also be that I did not need to put in amidiauto at all. Maybe I’ll try that on another card to see if it simplifies the process. I’m saying you might want to try it without amidiauto first to see).]

Midi: - (ALSA is the onboard audio and midi solution that is part of Raspbian). To have PD use ALSA midi at the start I made the following setting in the preferences/startup dialog - within that window there is a section (initially blank) for startup flags. Here you can set instructions for PD to take note of when it starts up. I put in -alsamidi to tell it that alsamidi will be my preferred midi output. (I also took the step of going to file/preferences/midi settings, then ‘apply’ and ‘ok’ to confirm the Alsa midi ports that showed up. Then I went back to file/preferences/save all preferences. This seems to have (fingers crossed) saved the connection to my USB midi output.

Audio: I used the terminal and sudo raspi-config to set my audio out to the internal sound card (advanced options/audio/3.5mm jack). Since I had a fairly unused installation of PD I’d never asked it to do anything but work with the system defaults so getting audio out was fairly simple.

[nb I initially stuck this patch together on my Mac where everything worked pretty trouble free in terms of audio and midi selection]

About the patch. Obviously it is sort of horrible but there it is. It is a combination of stuff I cribbed from the demo example of abl_link~ in the example, and two example patches created by users NoDSP and jpg in this forum post https://forum.pdpatchrepo.info/topic/9545/generate-midi-clock-messages-from-pd/2
As well as some basic synthesis to make the bip bip noises I learned from LWMs youtube channel
https://www.youtube.com/channel/UCw5MbnaoDDuRPFsqaQpU9ig

Any and all errors and bad practice are mine alone.
The patch has some comments in it that doubtless expose my own lack of understanding more than anything. Undoubtedly many users can do a better job than I can.

Some observations on limitations/screwups of the patch:

• If you disconnect from the stream for a bit, it will attempt to catch up. There will be a massive flurry of notes and/or audio bips as it plays all the intervening notes.

• It doesn’t seem to be too fussy about where in the bar it is getting started (It will be "on" the beat but sometimes the ‘1’will be the ‘2’ etc. This is okay if I’m using internal sequencers from scratch (in the volca, say) but not if there is an existing pattern that I am trying to have come in 'on the 1'.

• My solution to more detailed subdivision of bars was to make a big old list of numbers up to 32 so that abl_link~ can count up to more than 4. There’s probably a better solution for this. If you find that you need even more subdivisions because you are making some sort of inhumanly manic speed gabba, add even yet more numbers and connections.

I haven’t tested this much. And since it’s taken me the better part of 18 months to do this at all, I’m really not your guy to make it work any better. I’m posting here so that wiser souls can do a better job and maybe share what I think has the potential to be a useful midi sync tool.

I plan to revisit https://community.blokas.io/t/script-for-launching-pd-patch-with-midi-without-aconnect/1010/3
for some pointers on setting this up to launch the patch at startup to give me a small, portable midi Link sync device for 5-pin and audio-pulse clocked devices.

This is my first ever bit of quasi productive input to any technical community (mostly I just hang around asking dumb questions… So be kind and please use your giant brains to make it better) I look forward to spending some time learning the basics now. link-sync.pd

Apologies for cross-postings

Fifth Annual Web Audio Conference - 2nd Call for Submissions

https://www.ntnu.edu/wac2019

The fifth Web Audio Conference (WAC) will be held 4-6 December, 2019 at the Norwegian University of Science and Technology (NTNU) in Trondheim, Norway. WAC is an international conference dedicated to web audio technologies and applications. The conference addresses academic research, artistic research, development, design, evaluation and standards concerned with emerging audio-related web technologies such as Web Audio API, Web RTC, WebSockets and Javascript. The conference welcomes web developers, music technologists, computer musicians, application designers, industry engineers, R&D scientists, academic researchers, artists, students and people interested in the fields of web development, music technology, computer music, audio applications and web standards. The previous Web Audio Conferences were held in 2015 at IRCAM and Mozilla in Paris, in 2016 at Georgia Tech in Atlanta, in 2017 at the Centre for Digital Music, Queen Mary University of London in London, and in 2018 at TU Berlin in Berlin.

The internet has become much more than a simple storage and delivery network for audio files, as modern web browsers on desktop and mobile devices bring new user experiences and interaction opportunities. New and emerging web technologies and standards now allow applications to create and manipulate sound in real-time at near-native speeds, enabling the creation of a new generation of web-based applications that mimic the capabilities of desktop software while leveraging unique opportunities afforded by the web in areas such as social collaboration, user experience, cloud computing, and portability. The Web Audio Conference focuses on innovative work by artists, researchers, students, and engineers in industry and academia, highlighting new standards, tools, APIs, and practices as well as innovative web audio applications for musical performance, education, research, collaboration, and production, with an emphasis on bringing more diversity into audio.

Keynote Speakers

We are pleased to announce our two keynote speakers: Rebekah Wilson (independent researcher, technologist, composer, co-founder and technology director for Chicago’s Source Elements) and Norbert Schnell (professor of Music Design at the Digital Media Faculty at the Furtwangen University).
More info available at: https://www.ntnu.edu/wac2019/keynotes

Theme and Topics

The theme for the fifth edition of the Web Audio Conference is Diversity in Web Audio. We particularly encourage submissions focusing on inclusive computing, cultural computing, postcolonial computing, and collaborative and participatory interfaces across the web in the context of generation, production, distribution, consumption and delivery of audio material that especially promote diversity and inclusion.

Further areas of interest include:

• Web Audio API, Web MIDI, Web RTC and other existing or emerging web standards for audio and music.
• Development tools, practices, and strategies of web audio applications.
• Innovative audio-based web applications.
• Web-based music composition, production, delivery, and experience.
• Client-side audio engines and audio processing/rendering (real-time or non real-time).
• Cloud/HPC for music production and live performances.
• Audio data and metadata formats and network delivery.
• Server-side audio processing and client access.
• Frameworks for audio synthesis, processing, and transformation.
• Web-based audio visualization and/or sonification.
• Multimedia integration.
• Web-based live coding and collaborative environments for audio and music generation.
• Web standards and use of standards within audio-based web projects.
• Hardware and tangible interfaces and human-computer interaction in web applications.
• Codecs and standards for remote audio transmission.
• Any other innovative work related to web audio that does not fall into the above categories.

Submission Tracks

We welcome submissions in the following tracks: papers, talks, posters, demos, performances, and artworks. All submissions will be single-blind peer reviewed. The conference proceedings, which will include both papers (for papers and posters) and extended abstracts (for talks, demos, performances, and artworks), will be published open-access online with Creative Commons attribution, and with an ISSN number. A selection of the best papers, as determined by a specialized jury, will be offered the opportunity to publish an extended version at the Journal of Audio Engineering Society.

Papers: Submit a 4-6 page paper to be given as an oral presentation.

Talks: Submit a 1-2 page extended abstract to be given as an oral presentation.

Posters: Submit a 2-4 page paper to be presented at a poster session.

Demos: Submit a work to be presented at a hands-on demo session. Demo submissions should consist of a 1-2 page extended abstract including diagrams or images, and a complete list of technical requirements (including anything expected to be provided by the conference organizers).

Performances: Submit a performance making creative use of web-based audio applications. Performances can include elements such as audience device participation and collaboration, web-based interfaces, Web MIDI, WebSockets, and/or other imaginative approaches to web technology. Submissions must include a title, a 1-2 page description of the performance, links to audio/video/image documentation of the work, a complete list of technical requirements (including anything expected to be provided by conference organizers), and names and one-paragraph biographies of all performers.

Artworks: Submit a sonic web artwork or interactive application which makes significant use of web audio standards such as Web Audio API or Web MIDI in conjunction with other technologies such as HTML5 graphics, WebGL, and Virtual Reality frameworks. Works must be suitable for presentation on a computer kiosk with headphones. They will be featured at the conference venue throughout the conference and on the conference web site. Submissions must include a title, 1-2 page description of the work, a link to access the work, and names and one-paragraph biographies of the authors.

Tutorials: If you are interested in running a tutorial session at the conference, please contact the organizers directly.

Important Dates

March 26, 2019: Open call for submissions starts.

June 16, 2019: Submissions deadline.

September 2, 2019: Notification of acceptances and rejections.

September 15, 2019: Early-bird registration deadline.

October 6, 2019: Camera ready submission and presenter registration deadline.

December 4-6, 2019: The conference.

At least one author of each accepted submission must register for and attend the conference in order to present their work. A limited number of diversity tickets will be available.

Templates and Submission System

Templates and information about the submission system are available on the official conference website: https://www.ntnu.edu/wac2019

Best wishes,
The WAC 2019 Committee

@whale-av: thats interesting.

Details about my system and configuration:

OS.: arch
pd-version: Pd-0.49.0 ("") compiled 17:22:56 Sep 26 2018
pd config:

audioapi: 5
noaudioin: False
audioindev1: 0 2
audioindevname1: JACK
noaudioout: False
audiooutdev1: 0 2
audiooutdevname1: JACK
audiobuf: 11
rate: 41000
callback: 0
blocksize: 256
midiapi: 0
nomidiin: True
nomidiout: True
path1: /home/samuel/.sgPD
npath: 1
standardpath: 1
verbose: 0
loadlib1: sgCLib/sgScriptLib
loadlib2: sgCLib/sgInputC
loadlib3: zexy
nloadlib: 3
defeatrt: 0
flags: 
zoom: 1
loading: no

sound-server: jackd
jack config:

> jack_control dp
--- get driver parameters (type:isset:default:value)
            device: ALSA device name (str:set:hw:0:hw:0)
           capture: Provide capture ports.  Optionally set device (str:set:none:hw:0)
          playback: Provide playback ports.  Optionally set device (str:set:none:hw:0)
              rate: Sample rate (uint:set:48000:44100)
            period: Frames per period (uint:set:1024:256)
          nperiods: Number of periods of playback latency (uint:set:2:2)
             hwmon: Hardware monitoring, if available (bool:notset:False:False)
           hwmeter: Hardware metering, if available (bool:set:False:False)
            duplex: Provide both capture and playback ports (bool:set:True:True)
          softmode: Soft-mode, no xrun handling (bool:set:False:False)
           monitor: Provide monitor ports for the output (bool:set:False:False)
            dither: Dithering mode (char:set:n:n)
        inchannels: Number of capture channels (defaults to hardware max) (uint:notset:0:0)
       outchannels: Number of playback channels (defaults to hardware max) (uint:notset:0:0)
            shorts: Try 16-bit samples before 32-bit (bool:set:False:False)
     input-latency: Extra input latency (frames) (uint:notset:0:0)
    output-latency: Extra output latency (frames) (uint:notset:0:0)
       midi-driver: ALSA MIDI driver (str:notset:none:none)

root-priority: Hmm. I guess no. I'll try your suggestion, and see if it helps!

Configuration is pretty standard, I guess.

My impression is that certain events in my patch cause some messages being sent to multiple destinations, which causes pd to get out of sync.
It's slightly difficult to isolate the problem(s) into a small example, since I am using my own library of abstractions for parsing/generating recursively structured messages and automatically/dynamically connecting objects. I use this e.g. to let one object track properties of another one. It's nothing too fancy, all just native pd.
The amount and length of messages is still very moderate. It's disappointing to see pd struggle already.
(e.g. a "sample recorder" would record audio and update the "sample" object, by sending a "length" of the recording in ms, which in turn also distributes this message to ~3 "sample player" objects).
I will try to make a small example, that shows the issues I am facing.

Thank you for your help so far!
Any suggestions are welcome

Hi!
Well first of all I should say that I'm very new to Pure Data, and this is the first time I'm using it on a real project. It's still a bit confusing for me, and there is a kind of complicated setup for what I'm trying to do,

Short question (my take on the issue):
Is there a way to assign a Pd patch to separate tracks in a DAW,? Like making the patch into a plugin.
Or is there an audio device that I can use with my DAW that supports multiple output channels, so I can assign each track to the desired patches within Pd.

Long explanation:
For this project I'm building a live music+visuals set with a band. I use a DAW to receive the sound of a full band via mic and line inputs on an USB interface. I also use a built-in sampler and synthesizer to play on certain songs and interact with the band, while sometimes also playing plain audio files that are too on separate tracks.
For the visuals I use Resolume Arena. As I kind of also play in the band (with the synth and the sampler), I need to be able to operate the visuals in a very simple yet interesting way, so audio-responsive effects are a nice option. I want to be able to assign as many parameters as I can to be responsive to the individual audio tracks that come from the band and my instrument. So my idea is to use the MIDI mapping feature on Resolume, which gives me many options.
On Pd I found a patch that takes audio signals and interprets them via [fiddler~] giving out MIDI. Now I only need to assign individual instances of the patch to individual tracks, and then restrict each of their MIDI output signals to a different note range so I can map them to Resolume.

In a perhaps less confusing way, this is what I need:
Starting point: live audio feed and audio files on DAW tracks
Ending result: A distinct MIDI signal for each track.

I hope this is understandable. If you can come up with any other way of achieving the main goal of this proyect (make audio responsive visuals on Resolume via MIDI mapping) and would like to share it, I would really appreciate your ideas.

Thanks in advance for your help!

I've been doing a bunch of experiments with PD on a Raspberry Pi, with custom-built MIDI control via a Teensy microcontroller. I've been using Raspbian lite with no GUI. This was working really well until recently.

For various reasons I updated my Raspberry Pi to the latest Raspbian (Stretch) which also allowed me to get a slightly more decent build of PD, 0.47.1.

Since doing that I can't seem to get any MIDI input in PD, no matter what startup flags I use. Most tellingly, if I run pd -nogui -listdev I get the following list:
audio input devices:

1. bcm2835 ALSA (hardware)
2. bcm2835 ALSA (plug-in)
3. Teensy MIDI (hardware)
4. Teensy MIDI (plug-in)
   audio output devices:
5. bcm2835 ALSA (hardware)
6. bcm2835 ALSA (plug-in)
7. Teensy MIDI (hardware)
8. Teensy MIDI (plug-in)
   API number 1

no midi input devices found
no midi output devices found

--
I find it very odd that it lists my Teensy MIDI device as an audio input and output, and also says that no midi input or output devices have been found. It is somewhat understandable that my patches will not therefore recognise any midi activity, but I don't understand why PD isn't seeing the MIDI devices.

If I run aconnect -o I can see that the Raspberry Pi recognises the device:
client 14: 'Midi Through' [type=kernel]
0 'Midi Through Port-0'
client 20: 'Teensy MIDI' [type=kernel,card=1]
0 'Teensy MIDI MIDI 1'

and if I run aseqdump -p 20 the MIDI data comes streaming through normally. I'm interpreting this to mean that the MIDI device is working, and the alsamidi system is working on the rPi. My only explanation is that something has changed in PD 0.47.1 to create this bug?

I am thinking about starting from scratch and installing Raspbian Jessie instead to test and see if this works, but I'd like to avoid that if possible! Any ideas?

Some more details, in case anyone else is able to jump in on this...running PD 0.47.1 on my Raspberry Pi headless on the latest raspbian stretch lite.

similar to @francis666, if I run sudo pd -nogui -listdev I get the following list:
audio input devices:

1. bcm2835 ALSA (hardware)
2. bcm2835 ALSA (plug-in)
3. Teensy MIDI (hardware)
4. Teensy MIDI (plug-in)
   audio output devices:
5. bcm2835 ALSA (hardware)
6. bcm2835 ALSA (plug-in)
7. Teensy MIDI (hardware)
8. Teensy MIDI (plug-in)
   API number 1

no midi input devices found
no midi output devices found

I'm hoping to use Teensy MIDI as a USB midi device. If I run aconnect I can see the MIDI input coming in so it's definitely working, it's just not getting recognised by PD. I've tried a number of different startup flag combinations, but there's something fishy about the "no midi input devices found" message that I'm getting there. What's going on? Any ideas? I've definitely done this a number of times in the past, just using the command sudo pd -nogui -midiindev 1 midwest.pd but that now doesn't seem to work.

The main recent change has been updating to Stretch and PD .47 from .46, is there any way that could be causing the problem?

Hi Matthieu,

I see your post is a little bit old but I'm experiencing the exact same problem now with my setup.
I'm using a Windows 7 machine with a MOTU 828x sound card connected via USB to the PC and Pd 0.48.1 vanilla.

Here what I've done:

1. I've checked the "Use Stereo Pairs for Windows Audio" inside the "MOTU Audio Console";

2. opened PD and selected "standard MMIO" as driver from the "Media" menù;
3. now here's the list of outputs as it appears from the drop-down menu of "Media/Audio Setting.../Output device":

• MOTU Analog 3-4
• Loudspeakers (devide High ...
• MOTU Main-Out 1-2
• MOTU ADAT optical A 3-4
• Digital Output MOTU Audio
• MOTU ADAT optical A 1-2
• MOTU Analog 1-2
• MOTU ADAT optical A 7-8
• MOTU ADAT optical B 3-4
• MOTU Analog 7-8
• MOTU Analog 5-6
• Digital Output
• MOTU SPDIF 1-2
• MOTU ADAT optical B 1-2
• MOTU ADAT optical A 5-6
• MOTU Phones 1-2

As you see this list is pretty messed up and the names of logical consecutive output channels are not consequential. I would like to have 8 analog outputs from my MOTU so I selected the first item on the list (MOTU analog 3-4) then specified a total of 22 channels.

I'm obliged to set 22 as the total number of output channels because in my list MOTU Analog 5-6 are the last analog elements present. Because items in the list represent pairs of channels, this item corresponds to logical channel 21 and 22.

4. Then I created the dac object this way:

[dac~ 13 14 1 2 21 22 19 20]

Here's an image

This way I'm able to hear sound on all analog outputs of the MOTU even if I'm experiencing variuous 'clicks' and a series of "resyncing audio" messages inside the PD console...

I confess, this method is the only way I'm able to make this setup work but it seems to me to be pretty messy and not intuitive at all.

What seems to be even worse is that analog audio outputs inside the device list seems to change their order at each computer restart, so every time I have to restart from scratch.

• Is there some easier solution to this problem?
• Maybe a preference file I can create for PD to load at each startup containing all these settings?
• or there may be a way to programmatically select correct "analog outputs" from the device list in my patch (even if string parsing doesn't seem to be so easy in PD to me).
• Would launching PD from console, maybe from an ad-hoc script, solve the problem?

Thank you so much for your support
M