To make it short: I'll be back - probably in a different format than a blog.

And still there are Twitter, Facebook, Google+ and Audiotool for your entertainment.

Thanks for reading since 2005!

If you do not have a ticket yet - you should run.

I've updated the DemoSuperMario to run SoundSheets. The synth is decoding MP3 in runtime. Samples are taking from Reason. Could be better...

Listen to the piano.mp3 to get an idea how it works.

More updates at Audiotool

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I am happy to announce another Audio Code Clash Workshop, where I am teaching to work with Tonfall, an open-source library to create all kind of music and sound creations. Furthermore I will talk about sound creation and music in Flash in my session Pulsatile Crackle. Do not miss it. I won't! See you in Amsterdam.

My version is rather optimized for runtime purpose, does not create objects in runtime.

Press UP/DOWN to add/remove circles to the steiner chain.
Press RIGHT/LEFT to adjust the rotation speed.
Ensure keyboard focus (Click once in the Flash movie)

View this page in Romanian by courtesy of azoft.

More references
Wiki | Wolfram

Download

Pulsate was initially written for Flash

More information

Decoding a WAV results in the flash.media.Sound data format 44.1Khz,Stereo,32bit (float).
You can play it with the dynamic Sound API. Encoding can be done by simply passing the data you created in SAMPLE_DATA_EVENT to the encoder after specifying the target wav-format. You have to finalize the encoder after writing. Both processes are on the fly - processing block by block. They do not have to be processed in a single Flashplayer frame time.

Supported formats

• 44.1Khz/8bit/Mono
• 44.1Khz/8bit/Stereo
• 44.1Khz/16bit/Mono
• 44.1Khz/16bit/Stereo
• 44.1Khz/32bit/Mono
• 44.1Khz/32bit/Stereo

Please check out IWavIOStrategy, if you want to implement new wav-formats.

Browse source code

Decode Example

phase += frequency / samplingRate;
phase -= int( phase ); // [0,1]

amplitude = waveshaper( phase ); // [-1,+1]


Tonfall provides wave-shaping function like sinus, triangle, sawtooth and pulse (rectangle). The last one is well known by me from the C64 sound-chip and provided setting the pulse width. A normal pulse wave is 1.0 for all phase value lower than 0.5 and -1.0 for the rest of the phases. If you modulate the center (0.5) back and forth, the sound gets more wide and living. This is called Pulse-modulation but also reflects to phase-modulation. The effect on this simple waveform is the same.

For Audiotools Pulverisateur we wanted to implement such modulations for all kinds of waveshapes, even curves. So I needed to find a good and fast way to translate the phase value into a modulated value that could be passed into the wave-function.

amplitude = waveshaper( modulate( phase ) ); // [-1,+1]

These were the requirements

• Receive a value between zero and one and return a value between zero and one
• Ability to bend the value in both directions (symmetric)
• Be able to remain the input

Lets see what I came up with

The best part is, that there a parameter t to shape the curve. The upper curve (t>0) extends the output value and it compresses it later. The lower(t<0) is exactly doing the opposite. It is symmetric to y=x. Lets compare some values of t.

t: +/- 2

t: +/- 1

t: 0

That is another benefit. When you pass t=0, you get a straight line.
Meaning: output = input

t: +/-64

For really high values for t towards INFINITY you get in the end a right angle.

Costs
Curves like that are usually very expensive. I stumbled upon Super Ellipse, which distracted me for a couple of hours. It uses exponential functions, which I cannot run on every sample at 44100 Hz. I thought about taking a quadric bezier curve, but transforming the function to x leads to a sqrt-root. This solution presented here costs just one division and some simple algebra, which is okay for audio-dsp. I don't know, if I invented this thing. If you know, what it is, let me know!

Phase modulated sinus-wave

Function (for positive t)

Function (for negative t)

Actionscript

private function bend( x: Number, t: Number ): Number
{
if( t < 0.0 )
{
return x / ( t * x - t + 1.0 );
}
var tx: Number = t * x;
return ( tx + x ) / ( tx + 1.0 );
}


If you want to listen to the implementation, check out Audiotool, launch the application and create a Pulverisateur. Make sure to init the default patch and start with a single oscillator. Hope you like it!

Or just open a template I made on Audiotool.com

Circles.pbk

Update: There is a useful application for that!

Tonfall is not meant to be a fully implemented audio engine but you may learn a lot about signal processing and sequencing in common. It covers most questions I am receiving by email and is not that far away from the audio engine we use in Audiotool.

Features

• Holding a list of processors and time information
• SignalProcessor for audio generation taking TimeEvents (like notes) into account
• Design Pattern to process polyphonic audio generation (e.g. synthesizer)
• Static Delay as a template for other effects
• Note to frequency mapping
• WavFile Encoder
• Samples of the Casio RZ1 drumcomputer
• AmenBreak sample
• Super Mario playing Sequencer
• Fully launchable Metronome (Hello World of Audio DSP)
• Simple implementation of ToneMatrix

Tonfall is built with FDT 4.0, which I highly recommended. I checked-in the project files as well to let you compile right away.

I also highly recommend to check out Justin Windle's AS3 Particle Node Sequencer, which is based on Tonfall and a very nice implementation of what you can do with it. Thanks again Justin!

My session this time is called Pulsatile Crackle and will combine audio synthesis with physic simulations. And of course I give you a quick overview of what our cute baby Audiotool is already capable of. Never forget: Audio is the next big thing in Flash

Early bird price for this event ends Friday July 2nd, so get your tickets NOW. and use the code 'andre-michelle' for an extra 10% off!

If you don't have your ticket yet and are able to take a trip to Minneapolis, do not hesitate!

Visit Flashbelt.com

This will be some conference and I will also have my session Pulsatile Crackle!
I am really excited about all this. If you don't have tickets yet, go sack them quickly. I am pretty sure, that they are leaving John's hands right away.

Also, John was asking me how I find myself being Marilynized.
Ah! That's it! Marilynized! Hell yes, I like that ;) That's what makes FOTB so special, right?

Workshop Audio Code Clash
I am sure, attendees will learn a lot more than I can usually cover in my sessions.
Besides the workshop, I will show some pretty nice stuff on audio and physics called "Pulsatile Crackle".

See you at Flash on the beach.

Is it possible to loop MP3 without gaps?
Yes, but to answer the question I like to explain, why gap-less MP3 looping was actually not possible before Flash10 (Except for Flash-IDE encoded MP3 or ugly hacks).

Encoding Audio
Every audio encoder works by subdivided the audio stream into frames. Each time-frame contains information about the current frequency bands in use. You can easily follow, why an algorithm needs at least a couple of amplitudes (called a window) to analyze the spectrum. A single amplitude does not have any information about frequencies. Imagine a snapshot of a speakers membrane. There is no way to say anything about its velocity, not even its sign.

MP3 loss-less
MP3 also encodes audio with a certain amount of samples each frame. The length of a frame however is not free to choose. That is why encoder add more data (encoder delay) to the audio stream then actually necessary to ensure minimal frame length (depending on algorithm). Unfortunately the engineers of MP3 back then forgot to add the information how much silence they added to the encoded file. To make it absolutely clear: This information is lost for all times. You may write algorithms to estimate the delay, but it is not possible to recover exactly.

Btw: FlashIDE encoded MP3 loop gap-less, cause the SWF contains the original amount.

Does Lame help?
Lame offers embedding this missing information in a hidden MP3 frame. But to get this information you have to load the MP3 as a flash.utils.ByteArray, parse all MP3 frame-headers (no decoding necessary) and do some calculations. But afterward you need to load it again into a flash.media.Sound object to make use of the extract method. So better store this information of each loop in your database or simply somewhere in the Actionscript.

Solution
Once you have access to the original amount of samples you can use the Flash10 Playback API to playback your loop and wrap it seamlessly on low-level. You only have to take the encoder delay into account. If you extract from the very beginning you read nothing else than zero amplitudes. Check out the magic-number provided in the source code. This number works perfectly with Lame encoded MP3. This number is not Magic of course. It simply combines the amount of samples LAME added in front of the MP3 - plus - the amount of samples the Flash decoder thinks it were added. You can actually loop waveforms now.

Example

Track Void Panic | wav | mp3 | 124417 samples | Sourcecode

Still to complicated?
Vote for native Vorbis support!