An idea of audio encode algorithm, based on maximum allowed volume of , WavPack hybrid mode test included Options

[softrunner]

Full topic title: "An idea of audio encode algorithm, based on maximum allowed volume of signals difference"

Recently I have discovered for myself, that the difference of the source and encoded audio can be easily obtained by inverting source audio and mixing it with the encoded one. Then the idea of encode algorithm came into my head: just try to keep the signals difference at the same level (or less), defined by user. Thus, the audio quality is simply measured by volume of the difference of the signals, and this difference is nothing but distortions, produced by encoder.
The whole algorithm looks like this:
1. Take maximum allowed volume of signals difference from user.
1. Make a copy of source audio and invert it.
2. Split both source and inverted audio on frames of the same size.
3. Encode first frame of source audio, mix the result with first frame of inverted audio and calculate the volume of obtained difference.
4. If the volume of the difference is higher, than allowed by user, add some bitrate and repeat from item no. 3.
5. If the volume of the difference is not higher, than allowed by user, add first encoded frame to the final output.
6. Repeat items 3-5 with second, third, etc... frames, until the end of the source file.

Of cause, this algorithm is much slower then just direct encode, but definately if should not be slower, than video encoding (and people are ready to wait for many hours while their videos are being encoded).

I tried to reproduce this algorithm manually by test using WavPack hybrid mode as an encoder (source audio sample was splitted on 11 parts of 1 second), and it showed, that 23.4 % of space/bitrate could be saved. Another important thing is that the user is guaranteed, that he will not get distortions with volume level, higher then he expects, so he can safely encode many files simultaneously without looking at the content. User gets freed both from unnecessary waste of bitrate and uncontrolled distortions.

The only thing is needed is that some audio developers get interested in this idea and implement it as a computer program.

The whole set of files of the WavPack test I've made is here.

This post has been edited by softrunner: Mar 6 2013, 00:20