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Playing around with the new Flash player 10 audio processing functionality the need for optimization becomes very apparent when you want to apply effects to several tracks of audio.

With a sample rate of 44100 and a dozen stereo tracks we are talking over a million samples to be processed per second where each process you apply will have probably at least some 30 operations. All of a sudden the great performance of AVM2 becomes quite limiting.

So it's important to squeeze out every drop of performance you can by optimizing the code.
First of all I have been benchmarking the performance of running code inside the processing loop, in a function, in an external class and inside a loaded swf (would have been neat for the possibility to plug in effects without recompiling the main swf).

The code I used for testing was to process a value and return it like this (obviously without function enclosure when doing the processing in the local scope):

public function calculate(num:Number):Number
{
	return num * 1.01;
}

The time needed in ms when calling the function 10 000 000 times:

• Locally: 46
• Calling a function in the same class: 213
• Calling a function in a separate class: 213
• Calling a function in an externally loaded swf: 2347

Not so surprising results.
Having processing code in an external swf is obviously not an option. I tried with both simply sticking a function in the swf or in a class which I retrieved by applicationDomain.getDefinition and both methods performed equally bad.
Doing processing locally instead of in a separate function or class is a lot faster, but obviously that could easily becomes very cumbersome and ugly.
At least there is nothing lost on having the function in a separate class compared to having a function in the same class.

Something that does surprise me a bit is that when just calling the function once and having the loop inside the function instead the resulting time was 75ms.
That's about 30 ms added for just one function call so it seems like the first call is a lot more expensive.

One would think that the conclusion is that the best approach when processing audio if one like to avoid placing the code inside the samplesCallbackEvent loop seems to be calling the processing code once and then iterate over the size of the buffer in the class for the effect.
This is exactly what I was suggested by Spender when I posted a 3-band EQ example.

The problem there and why my attempt at implementing his suggestion failed at making an improvement is that reading and writing floats in a ByteArray is slower than the function calls.
Testing to writeFloat 10 000 000 times to then read loop through them to read them again takes 1727 ms. So compared with the 213 ms doing the same amount of function calls it's clear that function calls actually is comparatively cheap. A Vector fares a bit better then the ByteArray with 1239 ms.

So the optimal approach seems to be to only do samples.readFloat once then use the returned value doing function calls for each process you like to apply before you do samplesCallbackData.writeFloat

Tags: AS3, Astro, Audio, benchmark, flash player 10

I have had reason to play around with some of the new functionality of Flash Player 10 and the vectors is just awesome.
On top of the benefits of the strict typing they are about 50% faster than Arrays according to my tests.

Being completely new to the concept, how to create multidimensional vectors was not completely obvious since you need to type every dimension when declaring the bottom level dimension:

var v1:Vector.<Vector.<Vector.<int>>>= new Vector.<Vector.<Vector.<int>>>();
var v2:Vector.<Vector.<int>> ;
var v3:Vector.<int>;
for (var i:int = 0; i < 10; i++) {
	v2 = new Vector.<Vector.<int>>();
	for (var ii:int = 0; ii < 10; ii++) {
		v3 = new Vector.<int>();
			for (var iii:int = 0; iii < 10; iii++){
				v3[iii] = iii;
			}
		v2[ii]=v3;
	}
	v1[i]=v2;
}

So far I have mostly been experimenting with the new samplesCallbackData to create a little mixer.
It seems like one needs to bypass the mixer in the Flash Player if one want to write to the output buffer because creating several sounds and then doing samplesCallbackData.writeFloat() on them will not work.
Of course each channel doesn't have its own output buffer, so you can only write to the master output.
The problem I'm having with this is that if one would like to have a level meter for each individal track I cannot figure out a way to determine what sample is currently being output.
Here is a simplified version of how I implemented the mixing in the SamplesCallbackEvent handler:

var i:int = 0, l:int = _sounds_vect.lengt;
while (i < l) {
	samples = new ByteArray();
	samples.position = 0;
	snd = _sounds_vect[i];
	snd.extract(samples, _bufferSize);
	_samples_vect[i] = samples;
	_samples_vect[i].position = 0;
	i++;
}
while (c < _bufferSize) {
	left = 0;
	right = 0;
	i = 0;
	while (i < l) {
		valL = _samples_vect[i].readFloat();
		valR = _samples_vect[i].readFloat();
		left += valL;
		right += valR;
		i++;
	}
	valL = left;
	valR = right;
	_out_snd.samplesCallbackData.writeFloat(valL);
	_out_snd.samplesCallbackData.writeFloat(valR);
	c++;
}

So the audio is mixed in chunks the size of the buffer and then written to the buffer using samplesCallbackData.writeFloat().

For the main output I can create a level meter using:

_out_chn = new SoundChannel();
_out_chn = _out_snd.play();
function onEnterFrame(e:Event):void {
	_masterSlider.drawLevel(_out_chn.leftPeak, _out_chn.rightPeak);
}

But for the individal channels I will never issue a play() and hence cannot find a way to get a really consistently behaving level meter or spectrum.
I'm sure there is some clever way to do it that is escaping me.

What I currently do is to get the value in the mixing loop like so:

while (i < l) {
	valL = _samples_vect[i].readFloat();
	valR = _samples_vect[i].readFloat();
	left += valL;
	right += valR;
	if (c == 0){ // tried with (c == _bufferSize - 1) and (c == _bufferSize/2) as well
		_levelR_vect[i] = valL;
		_levelL_vect[i] = valR;
	}
	i++;
}

I then use the _levelL_vect and levelR_vect values in my onEnterFrame handler to draw the bars, but the result is a lot less accurate than what is possible using my_chn.leftPeak and far from satisfactory.
I guess what I would need is a way to be able to tell what sample from the output buffer that is playing a certain moment in time.

Apart from that small issue it's great to have the functionality to generate and process audio and I think we will see some very cool applications appearing eventually.

Tags: AS3, Astro, Audio, FlashPlayer 10

Finally Flash will have built in ability to access the sound output buffer when using FlashPlayer 10 that just has been released.
Tinic Uro have posted a little information about the implementation.
So no more relying on complicated hacks, this is all the code you will need to generate a sine wave (snipped from Tinics post):

  var sound:Sound = new Sound();
  function sineWavGenerator(event:SamplesCallbackEvent):void {
    for ( var c:int=0; c<1234; c++ ) {
      var sample:Number = Math.sin(
               (Number(c+event.position)/Math.PI/2))*0.25;
      sound.samplesCallbackData.writeFloat(sample);
      sound.samplesCallbackData.writeFloat(sample);
    }
  }
  sound.addEventListener("samplesCallback",sineWavGenerator);
  sound.play();

Tags: AS3, Astro, Audio, Flash

I added a little application to the download section.
It takes a Gregorian date and converts it to a dreamspell date and then calculates the kin, guide, analog and antipode to generate the affirmation for that day.
For those of you not familiar with the dreamspell calendar it's a reinterpretation of the Mayan calendar by Dr. José Argüelles.
It's based around 20 "glyphs" and 13 "tones" which make up the 260 day year called the Tzolkin.
The idea behind it is to create a new calendar system that actually is in tune with natural cycles, unlike the Gregorian system we use today, but mostly it's used to cast horoscopes.
Here is one resource for more information about the calendar.

My wife wanted to have an neat application for displaying the affirmation of the day and to look up a persons birth "kin", so I put this one together for her:

If this sort of stuff interest you are free to hotlink to the swf using the following URL:
http://www.resonantearth.com/ingrid/dreamspell.swf
You can download the swf here.

To get the source visit the download page.

Tags: AS3, Source

Many thanks to Guy Watson who confirmed that the FFTMode for computeSpectrum doesn't work yet.
I was hoping to be able to post a proper spectral analyser for Flash, but that has to wait until the FFT is sorted.

So for now you can check out the code for the example waveform display I made.
It's pretty much the same as what was posted on www.richapps.de but I used the readFloat method to access the data to make the wave display correctly.

Tags: AS3

I was very excited to notice that you now with AS3 can retrieve amplitude and spectrum information. I been hoping for this for a long time, and now it's there
Problem is that it doesn't seem to work properly.

I looked at the example on www.richapps.de and tried it out.
The problem is that accessing the values of the byteArray with normal array access brackets will not give the correct values.
It will return the byte value, ie a value between 0 and 256. According to the documentation for AS3 the computeSpectrum should return a decimal value between -1.0 and 1.0.
So since the length of the byte array is 2048 and the number of values should be 2x256 the values is stored as a single-precision 32-bit value.
So that gives that each value should be stored in 4 bytes and that readFloat would be the appropriate method to retrieve the values.
So my code to read the array for the right channel looks like this:

while(i>1024){
   spectrum.position=i;
      sprites_array[Math.round(i/4)].scaleY=(spectrum.readFloat()*100);
   i=i+4;
}

That works fine when FFTMode is false.
It will display the raw wave as you can see in this example.
(requires flash player 8.5, click to start sound).

But what you usually want to do to display a spectrum is to use FTT.
That will analyse the wave and actually make it into spectral information.
So, I try with just setting FFTMode to true....but no luck
Here is the swf with FFT applied.
As you can see there is no spectral information. With a sine sweep like this what you should see is a thin spike travelling from right to left.

Am I doing it completely wrong, or is indeed the FFTMode not working as it should?
Someone managed to get it to display a proper spectrum graph yet, and if so how?

Tags: AS3