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iLBC AS3 Wrapper

Internet low bitrate codec. This is a swc that wraps the base functionality of the iLBC codec now maintained by google as part of the WebRTC project.

To use this with a flash microphone sampledata/bytearray:

  • mic.rate = 8 // 8 kHz
  • Convert from big to little endian for iLBC (just readFloat the byte array into a little endian array.

default iLBC encode input and decode output

PCM 16 bit signed, little endian, 8 kHz

Flash Microphone audio

PCM 32 bit float, Big Endian, mic.rate kHz Flash Microphone Doc. This means that to use this with flash audio you will need to set mic.rate=8 and convert from float to short.

ILB wrapper input/output

Encoder takes little endian PCM 32 bit float, 8 kHz audio and decoder outputs little endian PCM 32 bit float 44 kHz. Optionally it can use base64 for encoder output and decoder input with the special base64 swc. The input is the default audio that will come out of a flash microphone, if mic.rate = 8. If you want to have contiguous audio even if there are silent patches, I recommend setting mic.setSilenceLevel(0) from it's default of 10.

Compile iLBC.swc (optional)

Download Adobe Alchemy

Download flex 4.6

Getting Started/Install Adobe Alchemy

Webrtc version contains the most changes including upsampling audio, short to float conversion, and base64 encoding, although there is a reference implementation that you can play around with build off of the rfc. I suggest using the webrtc version is it's still being developed by google. Confirmed working build of webrtc with revision 1842 of webrtc trunk.

$ cd lib/webRTC
$ alc-on
$ make extract
$ make compileswcb64

Optionally: make compileswc - which gives for input/output deals directly with raw audio instead of base64 data.

Example

Full Usage of bin/iLBC_webrtc.swc at src/org/ilbc/codec/ILBCCodec.as

AS3 Code

Methods encode and decode are asyncronous and take as their first params a callback for completed. yieldTicks represents the number of ticks the function loops on a encode/decode before it yields to the main thread. It should be noted that you don't want to reset the encoder or decoder in between chunks of contiguous audio, but only once the audio has been completely processed.

import cmodule.iLBC_webrtc.CLibInit;
import org.ilbc.event.ILBCEvent;

Initialization of lib

private var ilbcCodec:Object;
ilbcCodec = (new cmodule.iLBC_webrtc.CLibInit).init();

Encode

encodedData = new ByteArray();
encodedData.endian = Endian.LITTLE_ENDIAN;
//noiseSupression 1=>Mild, 2=>Medium, 3=>Aggressive
ilbcCodec.encode(encodingCompleteHandler, encodingProgressHandler, rawPCMByteArray, encodedData, decodedData.length, yieldTicks, noiseSupression);

Decode

decodedData = new ByteArray();
decodedData.endian = Endian.LITTLE_ENDIAN;
ilbcCodec.decode(decodingCompleteHandler, decodingProgressHandler, encodedData, decodedData, encodedData.length, yieldTicks);

Progress Handler

function progressHandler(progress:int):void;

Decode Complete Handler

function decodingCompleteHandler(event:Event):void {
	ilbcCodec.decoderReset();
}

Encode Complete Handler

function encodingCompleteHandler(event:Event):void {
	ilbcCodec.encoderReset();
}