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mahsah
30th January 2007, 16:49
Hello,

I want to transcode a 448 kbps ac3 file to a lower bitrate HE-AAC2. How would I do this?

Also, would there be much artifacting if I did this, or should I just keep the AAC stream

Lastly, what bitrate would be good, and will FFDshow correctly decode the HE-AAC? Also, will I be able to downmix on playback with FFDshow?

Thanks.

shon3i
30th January 2007, 20:05
1. HE-AAC (aka HE-AACv1) and LC-AAC is only for 5.1, HE-AACv2 is only for stereo

2. If you use medium bitrates 128-256 for HE-AAC and high bitrates 256+ for LC-AAC, should not to be artifact's

3. Keeping original AC3 depends on you ( you should do some test's), but if want to be close original then LC-AAC must be used.

4. ffdshow will decode correctly any of aac audio, and can downmix using mixer presets.

Skelsgard
30th January 2007, 20:14
1) You can load the file into SNG (Simple NeroAACEnc GUI) as AC3 file source or Avisynth file source and encode to AAC-HEv2.
Or you can load the file into BeSweet (Belight) or BeHappy and encode with either NeroAACEnc or CT's encoder.

2) Of course, info will have to be dropped in order to reach the target size. If the target size is to low, much info will be lost, but with SBR and PS, the file can still sound good, without too much noticeable artifacts for standard ears.

3) The bitrate determines the filters used, no backwards. If you use a bitrate below a given number SBR will be applied, giving you HE-AAC. If you go even lower, PS will be added, giving you HEv2-AAC.
NeroAACEnc allows you to override the preset relationship between bitrate and filters, but they donīt recommend to do so as they state the encoder itīs tunned to work at its best.

4) ffdshow will decode the file correctly and custom and preset downmixing is possible within ffdshow.

Cheers.

foxyshadis
7th February 2007, 11:24
1. HE-AAC (aka HE-AACv1) and LC-AAC is only for 5.1, HE-AACv2 is only for stereo

There was actually some discussion, I think on HA, that it could be done by encoding the main 4 channels as 2x HEv2, center as HE, and LFE as LC (if there's anything in it at all), and I think one of the aac developers was looking into just such a thing. You could further optimize it by determining if the center channel is a dumb mix of L&R, but I don't know if AAC has that capability. (Though the mixer should.)

I guess that's not terribly relevant if it can't be done now though.

shon3i
7th February 2007, 12:42
There was actually some discussion, I think on HA, that it could be done by encoding the main 4 channels as 2x HEv2, center as HE, and LFE as LC (if there's anything in it at all), and I think one of the aac developers was looking into just such a thing. You could further optimize it by determining if the center channel is a dumb mix of L&R, but I don't know if AAC has that capability. (Though the mixer should.)

I guess that's not terribly relevant if it can't be done now though.
Well, i've relised that is possible (but is not in standard and that said one of nero audio devs on HA.org forum), but 5.1 HE-AAC @ 96kbps already have up to 15khz, which means, to have very accetable quality. This maybe will be usefull for 64kbps 5.1, and only way to make 5.1 audio @64kbps until MPEG surround not came in public.