You probably know that generating some real random data is not so easy to do with a computer. How to design a good Random Number Generator (or a pseudo-random one) is a math topic that you can work years on ; it's also something very important for real-life applications such as security/cryptography, for example when you need to generate strong passwords.
Usually (and this is true in general in cryptography), designing your own algorithm is bad, because unless you're a professional in this subject and your algorithm has been approved by peers, you're guaranteed to have flaws in it, that could be exploited.
But here, for fun (don't use it for critical applications!), let's try to generate 100 MB of true random data.
1) Record 20 minutes of audio in 96khz 16bit mono with your computer's built-in microphone. Try to set the mic input level so that the average volume is neither 0 dB (saturation) nor -60 dB (too quiet). Something around -10 dB looks good. What kind of audio should you record? Nothing special, just the noise in your room is ok. You will get around 20*60*96000*2 ~ 220 MB of data. In these 220 MB, only the half will be really useful (because many values in the signal - an array of 16-bit integers - won't use the full 16-bit amplitude: many integers "encoding" the signal might be for example of absolute value < 1024, i.e. will provide only 10 bits)
2) Now let's shuffle these millions of bits of data with some Python code:
from scipy.io import wavfile import numpy as np import functools sr, x = wavfile.read('sound.wav') # read a mono audio file, recorded with your computer's built-in microphone #### GET A LIST OF ALL THE BITS L =  # list of bits for i in range(len(x)): bits = format(abs(x[i]), "b") # get binary representation of the data # don't use "016b" format because it would create a bias: small integers (those not using # the full bit 16-bit amplitude) would have many leading 0s! L += map(int, bits)[1:] # discard the first bit, which is always 1! print L.count(1) print L.count(0) # check if it's equidistributed in 0s and 1s n = 2 ** int(np.log2(len(L))) L = L[:n] # crop the array of bits so that the length is a power of 2; well the only requirement is that len(L) is coprime with p (see below) ### RECREATE A NEW BINARY FILE WITH ALL THESE BITS (SHUFFLED) # The trick is: don't use **consecutive bits**, as it would recreate something close to the input audio data. # Let's take one bit every 96263 bits instead! Why 96263? Because it's a prime number, then we are guaranteed that # 0 * 96263 mod n, 1 * 96263 mod n, 2 * 96263 mod n, ..., (n-1) * 96263 mod n will cover [0, 1, ..., n-1]. (**) # This is true since 96263 is coprime with n. In math language: 96253 is a "generator" of (Z/nZ, +). p = 96263 # The higher this prime number, the better the shuffling of the bits! # If you have at least one minute of audio, you have at least 45 millions of useful bits already, # so you could take p = 41716139 (just a random prime number I like around 40M) M = set() with open('truerandom', 'wb') as f: for i in range(0, n, 8): M.update(set([(k * p) % n for k in range(i, i+8)])) # this is optional, here just to prove that our math claim (**) is true c = [L[(k * p) % n] for k in range(i, i+8)] # take 8 bits, in shuffled order char = chr(functools.reduce(lambda a, b: a * 2 + b, c)) # create a char with it f.write(char) print M == set(range(n)) # True, this shows that the assertion (**) before is true. Math rulez!
truerandom file should be truly random data!
The only issue I can see happen right now is if the ADC (analog-to-digital-converter) electronic component of your soundchip is highly biased (please drop me a message if you have such a device).
This code here is unoptimized, it took 2 minutes for 1 minute of audio. There's surely a better way to work with arrays of bits in Python, comments/improvements are welcome!
- How to test the randomness quality of this file? This is a complicated task, and here are some references to do that. This is very far from being a rigorous way to do it, but it can be a first step (quote from the linked page): I've seen winzip used as a tool to measure the randomness of a file of values before (obviously, the smaller it can compress the file the less random it is). If you do it on the file generated here, you get exactly the same size (or even a bit more) after zip-compressing the file! Idem with rar, 7z (which usually yield a far better compression ratio, especially for audio data), the compression ratio is 1:1.
Quick tip: here is how to create symlinks in Windows without using the command line tool
1) If you have Python installed, create
import win32clipboard # pip install pywin32 if you haven't installed it already import sys, os, subprocess fname = sys.argv win32clipboard.OpenClipboard() filenames = win32clipboard.GetClipboardData(win32clipboard.CF_HDROP) win32clipboard.CloseClipboard() for filename in filenames: base = os.path.basename(filename) link = os.path.join(fname, base) subprocess.Popen('mklink %s "%s" "%s"' % ('/d' if os.path.isdir(filename) else '', link, filename), shell=True)
Create a key named
HKEY_CLASSES_ROOT\Directory\shell. In this key create a subkey
commandcontaining the string
"C:\Python27\pythonw.exe" "C:\pathto\mklinkgui.py" "%1".
- Create a key named
HKEY_CLASSES_ROOT\Directory\Background\shell. In this key create a subkey
commandcontaining the string
"C:\Python27\pythonw.exe" "C:\pathto\mklinkgui.py" "%v"(please note the
How to use it?
First click on the file(s) or folder(s) that you want to create a symbolic link to. Do
Copy. (It works with multiple files!)
- Right click on the folder where you want to drop a link, choose
Mklink here, done!
When making instrument sample sets (e.g. church organ sample sets used with Hauptwerk or GrandOrgue, see my project Jeux d'orgues), 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.