Kostya's Boring Codec World

I’m working on creating psychoacoustic model from recommendations presented in 3GPP TS26.403. Implementation is very rough but at least it can produce the files with desired bitrate (not quite that bitrate but ~2kbps around it).

Now the tasks are to eliminate noise from encoded material and add block switching. Maybe window switching as well.
Oh, and commit that all to FFmpeg SVN.

Looks like Gabriel Bouvigne of mp3-tech.org (how many information I got from there!) and Lame fame took interest in AAC encoder. For now I’m following his advise and trying to implement psychoacoustic model after 3GPP TS 26.403 document. It should be simple yet effective enough.

In the other news: AAC decoder mutates to become fit for FFmpeg SVN inclusion. I hope that will happen soon. Keep going, Robert, and keep reviewing, Michael!

If you are interested in what happens with my encoder, here’s a piece of report.

Simple encoding works. That means you can encode files with it now and they can be played back and you’ll be able to recognize the sound. Also I’ve separated psychoacoustic model and encoder itself, so it calls model to ask what windowing to use and what scaling/coefficients to encode.
Can I say this concludes the task for this summer of code? Technically yes but there are few points I ought to finish.

Encoder side:

• MDCT for the cases different from simple 1024-point window (8 short windows sequence and two transition windows)
• correct bitstream writing for 8SS case
• probably multichannel encoding (it’s useless until we have defined multichannel audio API though)

Psychoacoustic model(s) side:

• good psychoacoustic model 🙂
• quantizer which allows rate control
• something else?

I can add some models after the work is complete too and probably tune it for my ears and music I like to listen to. Reading papers I got on psychoacoustic models should help.

Back to work then.

OK, now I have simple and not very correct AAC encoder. Because of quantization step missing (spectral coefficients should be downscaled by cube root from them) resulting AAC becomes louder and FAAD complain on quantisation value being too large. FFmpeg future AAC decoder just silently clips it. In any case, it produces sound close to original.

Since no psychoacoustics is employed for now, bitrate is too high (~400kbps per channel, no joint stereo savings too).

So, the plan is to:

• Fix and optimize bitstream writing (yes, bitstream packing is far from optimal too)
• Psychoacoustic model (I hope it will be easier than multichannel audio API in FFmpeg)
• Bitrate control

Back to work…

I’ve started working on AAC encoder for FFmpeg. I’ve bricked (=made a dead-tree brick copy) a bit of standard (it’s really big) and have written a bit of code too. Hopefully we will have fully working AAC encoder to the end of summer. It’s time to get rid of libfaac and libfaad dependencies!

The phrase chosen as title was coined by Robert Swain, who works on bringing GSoC-2006 AAC decoder to FFmpeg and adding SBR support to it.