A few thingz


Joseph Basquin


01/08/2021

An improved Launchpad for Ableton Live

The Novation Launchpad is a great controller for Ableton Live, especially because it allows you to play / jam / record without having to look at the computer screen.

Except for one thing: you can play a clip, record a clip, stop a clip... but you cannot delete a clip. This limitation can be annoying, because sometimes when playing with your synth/guitar/whatever you need to record many takes before having the right one, and you also want to be able to immediately delete the bad recordings. Strangely, this was not possible with the Launchpad (now possible with Launchpad Pro, but it is more than twice as expensive).

So here is a "MIDI remote script" (just a little .py file) that you can copy in C:\ProgramData\Ableton\Live 10 Suite\Resources\MIDI Remote Scripts\Launchpad (for Windows, or the equivalent folder on Mac), that adds this feature to the Launchpad: the bottom right button will be a "delete the currently selected clip" button.

Bonus: this script also transforms the last row of the Launchpad into "stop clip" buttons, which is quite useful.

Download: MainSelectorComponent.py

Enjoy!

PS:

Low latency audio on a Windows PC with the built-in soundcard

(Edited 05 November 2019)

So you want to use your music production software, with low latency on your PC/Windows laptop?

You have basically two options:

Michael Tippach's ASIO4ALL is incredibly useful for the PC music community since more than 10 years, because it turns your cheap computer's built-in soundcard into a low-latency one! With ASIO4ALL, you can plug a MIDI keyboard and play piano or synth with no "delay". Without it, the delay of more than 50 ms between the keypress and the sound makes it nearly impossible to play.

But ASIO4ALL has one major drawback: it's not multi-client. This means that if your DAW is open with ASIO4ALL as sound driver, then, if you open:

... then it won't work: the audio is not available for them: your DAW and ASIO4ALL have locked your soundcard.

This is really annoying and I can't count how many hours of my life I wasted since 10 years to find a solution for this (every few months/years I retried and retried and benchmarked every new method). (Ok switching to Mac would have been a faster solution...)

The real difficulty is that we would like to use

Setup (1): A music software in ASIO + a standard application like Firefox using the so-called Windows WDM driver

Here is a list of things I tried, unsuccessfully:

Now, promising solutions:

Now, a working solution (update 2019):

Interested for future evolutions and other (audio) tools?

 

 

SamplerBox - a hardware sampler (to be continued)

This is in my Top10 biggest projects.

For many years, I had been looking for a small portable inexpensive sampler that can be used to play a good 1 GB piano sample set (with many velocity layers and supporting sustain pedal), a 500 MB Rhodes piano sampleset, drums, etc. and all sorts of sounds.

There was nothing like this on the market, but instead:

So I started brainstorming about this (this was my first sketch, I always search ideas with a pen + paper):

 

 

... and came to the idea that this could be possible with a Raspberry Pi. I then started to code a mixing engine in Python. Everybody told me "This is impossible, you won't be able to have a decent polyphony with Python, it's not fast enough...". The challenge was appealing for this reason and also because the only solutions available on Raspberry Pi at that time were 6-voice polyphony software samplers.

I spent days on this, and finally released both open-source code and hardware (rough) schematics:

https://www.samplerbox.org

Here is the blog: https://www.samplerbox.org/blog

I was happy to achieve a polyphony of ... nearly 100 voices at the same time, so my mixing engine in Python+Cython was not too bad ;)

Then it's a long story, many people contacted me, a commmunity growed on the forum (I should do a post about the software powering the forum, it's handmade too), etc.

(to be continued...)

Working with audio files in Python, advanced use cases (24-bit, 32-bit, cue and loop markers, etc.)

Python comes with the built-in wave module and for most use cases, it's enough to read and write .wav audio files.

But in some cases, you need to be able to work with 24 or 32-bit audio files, to read cue markers, loop markers or other metadata (required for example when designing a sampler software). As I needed this for various projects such as SamplerBox, here are some contributions I made:

  1. The Python standard library's wave module doesn't read cue markers and doesn't support 24-bit files. Here is an updated module:

    wave.py (enhanced)

    that adds some little useful things. (See Revision #1 to see diff with the original stdlib code).

    Usage example:

    from wave import open
    f = open('Take1.wav')
    print(f.getmarkers())

    If you're familiar with main Python repositery contributions (I'm not), feel free to include these additions there.

  2. The module scipy.io.wavfile is very useful too. So here is an enhanced version:

    wavfile.py (enhanced)

    Among other things, it adds 24-bit and 32-bit IEEE support, cue marker & cue marker labels support, pitch metadata, etc.

    Usage example:

    from wavfile import read, write
    sr, samples, br, cue, cuelabels, cuelist, loops, f0 = read('Take1.wav', readmarkers=True, 
        readmarkerlabels=True, readmarkerslist=True, readpitch=True, readloops=True)
    write('Take2.wav', sr, samples, bitrate=br, markers=cue, loops=loops, pitch=130.82)

    Here is a Github post and pull-request about a hypothetical merge to Scipy.

Here is how loop markers look like in the good old (non open-source but soooo useful) SoundForge:


Lastly, this is how to convert a WAV to MP3 with pydub, for future reference. As usual, do pip install pydub and make sure ffmpeg is in the system path. Then:

from pydub import AudioSegment
song = AudioSegment.from_wav("test.wav")
song.export("test.mp3", format="mp3", bitrate="256k")

will convert a WAV file to MP3.


Interested for future evolutions and other audio programming tools?

Get the reverb impulse response of a church

I recently recorded an impulse response of the reverb of a 14th-century church (more or less the footprint of the sound ambiance of the building). Here is how I did it.

Quite a lot of reverb, that's exactly what we want to catch with an IR!

Then, of course, we can do some cleaning, fade out, etc.


But what is this useful for? You can use this Impulse Response in any music production software (the VST SIR1 is quite good and freeware) , and make any of your recordings (voice, instrument, etc.) sound like if they were recorded in this church. This is the magic of convolution reverb!


Useful trick when you record your own IR: play sweep0.wav in the building instead of sweep.wav. The initial "beep" is helpful to see exactly where things begin. If you don't do that, as the sweep begins with very low frequencies (starting from 20 Hz), you won't know exactly where is the beginning of your microphone-recording. Once your recording is done, you can trim the soundfile by making it begin exactly 10 seconds after the short beep.

Some related reading in this topic, and this blog post.

My personal blog,
Joseph Basquin (Orléans, France)

twitter
email
github

Data / AI / Python consulting and freelancing.

Articles about:
#music
#photo
#opensource
#all

Some of my projects:
BSQN Digital
BigPicture
SamplerBox
Ojourdui
Jeux d'orgues
YellowNoiseAudio