The Wavelet codec performs a wavelet transform on the original data. It then achieves its compression by thresholding the transformed data, converting all values below a given magnitude to zero, and then run-coding the resulting data. The run-coding provides automatic variable compression depending on the parameters chosen.

As is, this codec achieves reasonable reproduction at 10:1 compression, although the quality from the GSMCodec is definitely better. I feel that the quality would be comparable if uLaw scaling were introduced prior to thresholding.

The nice thing about using wavelets is there are numerous factors to play with for better performance:
nLevels - the "order" of the transform performed
alpha and beta - these specify the wavelet shape (some are better for speech)
the actual threshold used
By simply changing these parameters, one can easily vary the compression achieved from 5:1 to 50:1, and listen to the quality at each step.

The specific format for an encoded buffer is as follows:
4 bytes: frameCount.
4 bytes: samplesPerFrame.
4 bytes: nLevels.
4 bytes: alpha asIEEE32BitWord.
4 bytes: beta asIEEE32BitWord.
frameCount occurrences of...
2 bytes: frameSize in bytes, not including these 2
may be = 0 for complete silence, meaning no scale even.
4 bytes: scale asIEEE32BitWord.
A series of 1- or 2-byte values encoded as follows:
0-111: a run of N+1 consecutive 0's;
112-127: a run of (N-112)*256 + nextByte + 1 consecutive 0's;
128-255: a 15-bit signed value = (N*256 + nextByte) - 32768 - 16384.

	subclass responsibilities	top
	bytesPerEncodedFrame Answer the number of bytes required to hold one frame of compressed sound data. Answer zero if this codec produces encoded frames of variable size.
	decodeFrames:from:at:into:at: Decode the given number of monophonic frames starting at the given index in the given ByteArray of compressed sound data and storing the decoded samples into the given SoundBuffer starting at the given destination index. Answer a pair containing the number of bytes of compressed data consumed and the number of decompressed samples produced.
	encodeFrames:from:at:into:at: Encode the given number of frames starting at the given index in the given monophonic SoundBuffer and storing the encoded sound data into the given ByteArray starting at the given destination index. Encode only as many complete frames as will fit into the destination. Answer a pair containing the number of samples consumed and the number of bytes of compressed data produced.
	frameCount: Compute the frame count for this byteArray. This default computation will have to be overridden by codecs with variable frame sizes.
	samplesPerFrame Answer the number of sound samples per compression frame.