
Compressor
Type : Hardknee compressor with RMS lookahead envelope calculation and adjustable attack/decay References : Posted by flashinc[AT]mail[DOT]ru
Notes : RMS is a true way to estimate _musical_ signal energy,
our ears behaves in a same way.
to making all it work,
try this values (as is, routine accepts percents and milliseconds) for first time:
threshold = 50%
slope = 50%
RMS window width = 1 ms
lookahead = 3 ms
attack time = 0.1 ms
release time = 300 ms
This code can be significantly improved in speed by
changing RMS calculation loop to 'running summ'
(keeping the summ in 'window' 
adding next newest sample and subtracting oldest on each step)
Code : void compress
(
float* wav_in, // signal
int n, // N samples
double threshold, // threshold (percents)
double slope, // slope angle (percents)
int sr, // sample rate (smp/sec)
double tla, // lookahead (ms)
double twnd, // window time (ms)
double tatt, // attack time (ms)
double trel // release time (ms)
)
{
typedef float stereodata[2];
stereodata* wav = (stereodata*) wav_in; // our stereo signal
threshold *= 0.01; // threshold to unity (0...1)
slope *= 0.01; // slope to unity
tla *= 1e3; // lookahead time to seconds
twnd *= 1e3; // window time to seconds
tatt *= 1e3; // attack time to seconds
trel *= 1e3; // release time to seconds
// attack and release "per sample decay"
double att = (tatt == 0.0) ? (0.0) : exp (1.0 / (sr * tatt));
double rel = (trel == 0.0) ? (0.0) : exp (1.0 / (sr * trel));
// envelope
double env = 0.0;
// sample offset to lookahead wnd start
int lhsmp = (int) (sr * tla);
// samples count in lookahead window
int nrms = (int) (sr * twnd);
// for each sample...
for (int i = 0; i < n; ++i)
{
// now compute RMS
double summ = 0;
// for each sample in window
for (int j = 0; j < nrms; ++j)
{
int lki = i + j + lhsmp;
double smp;
// if we in bounds of signal?
// if so, convert to mono
if (lki < n)
smp = 0.5 * wav[lki][0] + 0.5 * wav[lki][1];
else
smp = 0.0; // if we out of bounds we just get zero in smp
summ += smp * smp; // square em..
}
double rms = sqrt (summ / nrms); // rootmeansquare
// dynamic selection: attack or release?
double theta = rms > env ? att : rel;
// smoothing with capacitor, envelope extraction...
// here be aware of pIV denormal numbers glitch
env = (1.0  theta) * rms + theta * env;
// the very easy hard knee 1:N compressor
double gain = 1.0;
if (env > threshold)
gain = gain  (env  threshold) * slope;
// result  two hard kneed compressed channels...
float leftchannel = wav[i][0] * gain;
float rightchannel = wav[i][1] * gain;
}
}

Comments
Added on : 10/06/04 by musicdsp[ AT ]umminger[ DOT ]com Comment : My comments:
A rectangular window is not physical. It would make more physical sense, and be a lot cheaper, to use a 1pole low pass filter to do the RMS averaging. A 1pole filter means you can lose the bounds checks in the RMS calculation.
It does not make sense to convert to mono before squaring, you should square each channel separately and then add them together to get the total signal power.
You might also consider whether you even need any filtering other than the attack/release filter. You could modify the attack/release rates appropriately, place the sqrt after the attack/release, and lose the rms averager entirely.
I don't think your compressor actually approaches a linear slope in the decibel domain. You need a gain law more like
double gain = exp(max(0.0,log(env)log(thresh))*slope);
Sincerely,
Frederick Umminger
Added on : 30/07/04 by xeeton[AT]gmail[DOT]com Comment : To sum up (and maybe augment) the RMS calculation method, this question and answer may be of use...
**********
musicdsp@shoko.calarts.edu writes:
I am looking at gain processing algorithms. I haven't found much in the way of reference material on this, any pointers? In the level detection code, if one is doing peak detection, how many samples does one generally average over (if at all)? What kind of window size for RMS level detection? Is the RMS level detection generally the same algo. as peak, but with a bigger window?
The peak detector can be easily implemented as a onepole low pass, you just have modify it, so that it tracks the peaks and gently falls down afterwards. RMS detection is done squaring the input signal, averaging with a lowpass and taking the root afterwards.
Hope this helps.
Kind regards
Steffan Diedrichsen
DSP developer
emagic GmbH
**********
I found the thread by searching old [musicdsp] forum posts. Hope it helps.
Added on : 06/11/06 by graue[ AT ]oceanbase[ DOT ]org Comment : How would you use a 1pole lowpass filter to do RMS averaging? How do you pick a coefficient to use?
Added on : 08/11/06 by scoofy[ AT ]inf[ DOT ]elte[ DOT ]hu Comment : Use x = exp(1/d), where d is the time constant in samples. A 1 pole IIR filter has an infinite impulse response, so instead of window width, this coeff determines the time when the impulse response reaches 36.8% of the original value.
Coeffs:
a0 = 1.0x;
b1 = x;
Loop:
out = a0*in  b1*tmp;
tmp = out;
 peter schoffhauzer
Added on : 20/11/08 by txutao[ AT ]163[ DOT ]com Comment : I am looking at gain processing algorithms£º
There are too such sentences :
double att = (tatt == 0.0) ? (0.0) : exp (1.0 / (sr * tatt));
double rel = (trel == 0.0) ? (0.0) : exp (1.0 / (sr * trel));
can you tell me something about the exp (1.0 / (sr * tatt))?
New day ~~
thanks
Added on : 28/04/10 by pak[ DOT ]nine[ AT ]gmail[ DOT ]com Comment : This is a useful reference, however the RMS calculations are pretty dodgy. Firstly there is a bug where is calculates the number of samples to use:
int sr, // sample rate (smp/sec)
...
double twnd, // window time (ms)
...
// samples count in lookahead window
int nrms = (int) (sr * twnd);
The units are mixed when calculating the number of samples in the RMS window. The window time needs to be converted to seconds before multiplying by the sample rate.
As others have mentioned the RMS calculation is also really expensive, and in my tests I found it was pretty innacurate unless you use a LOT of samples (you basically need a (sample rate)/2 window of samples in your RMS calculation to accurately measure the power of all frequencies).
I ended up using the 1 pole low pass filter approach suggested here, and it is a good cheap approximation of power. I did, however, end up mulitplying it by root(2) (the RMS of a sine wave, which seemed like a reasonable normalisation factor) in order to get it between 0 and 1, which is a more useful range.
Another slightly more accurate way to caculate the RMS without iterating over and entire window for each sample is to keep a running total of the squared sums of samples.
for( i = 0; i < NumSamples; ++i )
{
NewSample = Sample[i];
OldSample = Sample[i  RMSWindowSize];
SquaredSum = SquaredSum + NewSample * NewSample;
SquaredSum = SquaredSum  OldSample * OldSample;
RMS = sqrt( SquaredSum / RMSWindowSize );
// etc...
}
Calculating the power in the signal is definately the awkward part of this DSP!

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