A few thingz
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.
Don't forget to backup the aforementioned folder before adding this file, so that you can easily remove this extension if necessary.
- You can customize the Launchpad or any other controller as much as you want by editing these "MIDI remote script" Python files!
(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:
- use an external USB soundcard (but sometimes you're not in your studio - maybe traveling - and it is not an option to have a soundcard in your backpack!)
- use a (free) driver named "ASIO4ALL"
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:
- an external sound editor, such as SoundForge, to edit a sample
- or your media player to listen to a MP3 (for example to compare the mix you're currently working on with another song)
- or Youtube in your browser, to get some inspiration from your favourite artist
... 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:
Contact the author of ASIO4ALL and ask if a multiclient version would be possible
WASAPI drivers: promising Microsoft audio low-latency API but unavailable in Ableton Live (even version 10)
Asio2Ks [asio2ks_beta.zip, 118 KB], apparently not working anymore on Windows 7 / 64 bit. Copying the .cpl file gave nothing, and it seems impossible to open it once in
Vidance AsioMulti [asiomulti.msi, 4073 KB] (another link here), not working for the setup (1), maybe it would work for 2 ASIO programs, but not for 1 ASIO + 1 WDM at the same time. It made SoundForge crash when starting Ableton + SoundForge at the same time
FL ASIO driver (coming with FL Studio 12 demo): you can have audio both in the DAW (FL Studio but also Ableton Live, etc.) and in another program, but unfortunately, on my standard i5 laptop, the latency with FL ASIO is much poorer than with ASIO4ALL and there are many "crackles" when playing a VST instrument, whereas with ASIO4ALL everything is smooth (no such problem even with the lowest latency value)
Jack: the famous virtual cable solution for Linux, ported for Windows. I tried it a few days, but finally I always ended with artefacts, audio glitches, etc.
Vincent Burel's VirtualAudioCable (yes there are two programs by two different editors having the same name), I maybe should retry it again.
- Hi-Fi CABLE & ASIO Bridge, by the same author, is even closer to what I'm looking for. (Feature request: it would be great if it could set its "Hi-Fi Cable Input" device as default Windows playback device on startup of the program, and restore the standard speakers on exit). Strangely, it has worked once, but now, on startup of the program, I have a
Buffer: 1024blinking in red, and no sound, never mind! This might be because the buffer 1024 is too small for this software, but then it's surprising - AsioLinkPro works with a 512 buffer-size. Edit: now fixed after a reboot with a 4096 buffer. After some further tests, it finally does not work for the setup (1): this program does a bridge WDM => ASIO (ASIO4ALL in my case), but then my soundcard is no more available for ASIO4ALL in Ableton Live (screenshot here), which is totally logical. Conclusion: another bridging method is required, see setup (2) below.
Now, promising solutions:
O-Deus AsioLinkPro 2.4.2: nearly working for the setup (1) described above (still some glitches when playing an audio file of < 200 ms. Why would you want to play such short files? This is useful when you browse your sample library: kicks, snares, etc. are usually short samples)
O-Deus AsioLinkPro 220.127.116.11: totally working, perfect solution, in combination with ASIO4ALL. It even sets its device as "default Windows playback device" when we launch the program, and automatically removes it on exit... Except that this software is discontinued and it's impossible to buy a new license... (This has changed, see update below)
Here is AsioLinkPro's clever idea: you still use ASIO4ALL as output, but this way:
Setup (2): Ableton Live (or any other DAW) --> ASIO: AsioLinkPro \ --- AsioLinkPro mixer --> ASIO4ALL Firefox or Chrome or SoundForge --> WDM: ASIOVADPRO virtual device / or MP3 player (AsioLinkPro)
Clever, because even if there are 2 programs producing sound, AsioLinkPro is the only one which speaks directly with ASIO4ALL (which would not support 2 programs).
It must have been tricky to code it because it requires to code a "WDM virtual speaker device" Windows driver + an ASIO driver, phew!
Even if it's discontinued, at least it gives an idea about how to do it. Let's write such a minimalist open-source tool?
Note: not something very big and complex like Jack, but just a small WDM virtual speaker driver and an ASIO driver that both mix their content and send it to the ASIO4ALL output. (No GUI is even required).
Now, a working solution (update 2019):
FlexASIO 1.5 allows me to get 3.6ms latency in Exclusive Mode (non multi-client) and 20ms on Shared Mode with my laptop's built-in soundcard. A few months ago, I was unable to make it work when testing the version 0.1, but the development is currently very active, and it has evolved into a good solution. Here is my configuration file (
backend = "Windows WASAPI" bufferSizeSamples = 128 [input] device = "" [output] suggestedLatencySeconds = 0.0 wasapiExclusiveMode = true # change to false to have multiclient
FlexASIO works great, it unfortunately can only handle one input channel at a time.
- O-Deus AsioLinkPro 18.104.22.168 (link here) is now available for free (the family of the developer seems to have authorized the release, see this post for more details). A patcher is required, that can be found here.
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:
expensive samplers or expanders (800€+) that can do all of that, but they usually are not very portable
- cheap samplers like the Korg Volca Sample or the Akai MPX16, but they only offer a few MB of sound, and usually support no velocity layers, the loading time can be very long, etc.
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:
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...)
When making instrument sample sets (e.g. church organ sample sets used with Hauptwerk or GrandOrgue), we need to set looping points in WAV audio files:
such that when playing the part [a, b] in loop, we don't hear any click or pop when the sample reaches the end of the loop.
Example 1: bad loop with audible clicks
Example 2: seamless loop with no click, that's what we are looking for! The loop has a ~ 2.670 second period, can you hear where are the looping points?
Finding looping points can be done manually but this is a very long and tedious task. A few programs exist to do this process automatically such as Extreme Sample Converter (it has an excellent auto-looping algorithm), LoopAuditioneer (open source), Zero-X Seamless Looper, SampleLooper, etc.
Here we'll look at a home-cooked algorithm that works well to detect looping points.
First of all, let's load the audio file (downloadable here) with Python:
from scipy.io import wavfile import numpy as np import itertools sr, x = wavfile.read('060.wav') x0 = x if x.ndim == 1 else x[:, 0] # let's keep only 1 channel for simplicity, but we could easily generalize this for 2 channels x0 = np.asarray(x0, dtype=np.float32)
Let's say the audio file's sustain part (this is precisely where we're looking for a loop!) begins at t=2 sec and finishes at t=9 sec. We will now subdivide the time-interval [2 sec, 9 sec] into a 250 milliseconds grid: 2, 2.25, 2.5, 2.75, 3, 3.25, ..., 8.75, 9.
From this sequence, we now create "loop candidates" (a, b) of length at least 1 second, example: (2.5, 7.5), (3.25, 5.75), (6.0, 8.75), etc.
Then, for each loop candidate, we'll improve the loop (this is the core of the algorithm, it will be discussed in the next paragraph) and compute a distance
We finally keep the loop that has the minimal distance (among all loop candidates). Finished!
A = [int((2 + 0.25 * k) * sr) for k in range(29)] # the grid 2, 2.25, 2.5, ... 8.75, 9 dist = np.inf for a, b in itertools.product(A, A): # cartesian product: pairs (a, b) of points on the grid if b - a < 1 * sr: continue a, B, d = improveloop(x0, a, b, sr=sr) print 'Loop (%.3fs, %.3fs) improved to (%.3fs, %.3fs), distance: %i' % (a * 1.0 / sr, b * 1.0 / sr, a * 1.0 / sr, B * 1.0 / sr, d) if d < dist: aa = a BB = B dist = d print "The final loop is (%.3fs, %.3fs), i.e. (%i, %i)." % (aa * 1.0 / sr, BB * 1.0 / sr, aa, BB)
Finished? Not yet! We need to explain what we mean by improving a loop, as that's the crucial part of the algorithm. More precisely, we'll now explain how to transform a loop (3.25, 5.75) with points taken on the grid (this random loop probably "clicks" like in Example 1 before!) into a "good loop" (3.25, 5.831). Let's zoom on the junction point to understand what's going on:
How to measure if a loop is good or not? Ideally, if the loop (a, b) is perfect/seamless,
x[a:a+10 ms] should be very close to
Measuring how close two arrays
y are can be done by computing
sum((x[n]-y[n])^2), and if the sum is small,
y are close.
k such that
np.sum(np.abs(x0[a:a+W1]-x0[k+b:k+b+W1])**2) is minimal can be obtained by noting that
(x[n] - y[n+k])**2 = x[n]**2 - 2*x[n]*y[n+k] + y[n+k]**2
and by using numpy.correlate. We can now define this function:
def improveloop(x0, a, b, sr=44100, w1=0.010, w2=0.100): """ Input: (a, b) is a loop Output: (a, B) is a better loop distance (the less the distance the better the loop) This function moves the loop's endpoint b to B (up to 100 ms further) such that (a, B) is a "better" loop, i.e. sum((x0[a:a+10ms] - x0[B:B+10ms])^2) is minimal """ W1 = int(w1*sr) W2 = int(w2*sr) x = x0[a:a+W1] y = x0[b:b+W2] delta = np.sum(x**2) - 2*np.correlate(y, x) + np.correlate(y**2, np.ones_like(x)) K = np.argmin(delta) B = K + b distance = delta[K] return a, B, distance
That's it, in less than 50 lines of Python code!
This audio file
(looped 4 times here but we could loop it forever) has been obtained with the algorithm described here. Not too bad, n'est-ce pas?
Example of output:
Loop (2.000s, 3.000s) improved to (2.000s, 3.009s), distance: 1003724800 Loop (2.000s, 3.250s) improved to (2.000s, 3.340s), distance: 839278592 Loop (2.000s, 3.500s) improved to (2.000s, 3.559s), distance: 1281863680 [...] Loop (2.000s, 8.500s) improved to (2.000s, 8.544s), distance: 1092337664 Loop (2.000s, 8.750s) improved to (2.000s, 8.789s), distance: 964747264 Loop (2.000s, 9.000s) improved to (2.000s, 9.004s), distance: 2488913920 [...] Loop (7.750s, 9.000s) improved to (7.750s, 9.004s), distance: 1167093760 Loop (8.000s, 9.000s) improved to (8.000s, 9.001s), distance: 1710333952 The final loop is (6.750s, 8.322s), i.e. (297675, 366989).
Note: Wouldn't it be possible to save these loop markers inside the WAV file's metadata instead of just printing them on screen? Sure it is, but as Python's standard library doesn't support WAV markers editing, you'll have to use these techniques to do this.
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:
that adds some little useful things. (See Revision #1 to see diff with the original stdlib code).
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.
The module scipy.io.wavfile is very useful too. So here is an enhanced version:
Among other things, it adds 24-bit and 32-bit IEEE support, cue marker & cue marker labels support, pitch metadata, etc.
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) print 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) print read('Take2.wav', readmarkers=True, readmarkerlabels=True, readmarkerslist=True, readpitch=True, readloops=True) write('Take3.wav', sr, samples, bitrate=br, markers=cuelist, loops=loops, pitch=130.82)
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.
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.
- First I installed a loudspeaker (a studio monitor Yamaha HS-80M) in the church, quite high from the ground. I played, rather loud, a sound called a frequency sweep, that contains frequencies from 20Hz to 20000Hz, i.e. the entire human hearing range.
- Then, in the middle of the church, I recorded this with 2 microphones. Here is what I got:
Quite a lot of reverb, that's exactly what we want to catch with an IR!
Now, let's use some Digital Signal Processing to get the IR. All the source code in Python is here. If you're into math, here is the idea:
ais the input sweep signal,
hthe impulse response, and
bthe microphone-recorded signal. We have
a * h = b(convolution here!). Let's take the discrete Fourier transform, we have
fft(a) * fft(h) = fft(b), then
h = ifft(fft(b) / fft(a)).
- Here is the result, the Impulse Response of the church:
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.