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View Full Version : DTS to AAC - How?


Eretria-chan
27th May 2007, 13:16
I would like to ask if anyone knows a way to transcode DTS 5.1 to AAC 5.1? Everything I've tried has failed...
I have tried VLC, Belight (0.22, 0.21), dts2wav...
Any help would be appreciated!
:thanks:

tebasuna51
27th May 2007, 13:27
Try Foobar2000 or BeHappy-AviSynth to direct conversion.

Tranzcode and NeroAacEnc with intermediate wav file.

Eretria-chan
27th May 2007, 13:30
Forgot to mention I tried Foobar2000, but it doesn't seem to support DTS anymore...
But I'll try Behappy! Thanks!

tebasuna51
27th May 2007, 14:46
Forgot to mention I tried Foobar2000, but it doesn't seem to support DTS anymore...
Why not?:
foo_input_dts (http://www.saunalahti.fi/~cse/foobar2000/foo_input_dts.zip)

Eretria-chan
27th May 2007, 17:11
Well, maybe I was missing a plugin, to which I only found a broken link - but it doesn't matter...
BeHappy works like a charm!
100 x :thanks:

EDIT:
Actually, I seem to have another problem now...
The audio quality is worse than the original DTS track - but it isn't the bitrate! I maxed it up to test but it seemed to have no effect. I have no idea what it is or what the effect is called, so I uploaded two samples: http://www.eluni.net/temp

The DTS is the original and the AAC is the encoded. Both are 5s clips that clearly illustrates my problem...
If anyone could tell me what's wrong and how to fix it...?

tebasuna51
27th May 2007, 19:05
Actually, I seem to have another problem now...
The audio quality is worse than the original DTS track - but it isn't the bitrate! I maxed it up to test but it seemed to have no effect. I have no idea what it is or what the effect is called, so I uploaded two samples: http://www.eluni.net/temp

The DTS is the original and the AAC is the encoded. Both are 5s clips that clearly illustrates my problem...
If anyone could tell me what's wrong and how to fix it...?

I think the original is the dts in matroska container:
Sample2_AAC.mka 559 KB (754 Kb/s)
And the encoded is:
Sample1_DTS.m4a 61 KB (96 Kb/s)

Really your encoded sample1 have a really good quality ... for a 6 channel audio encoded at 96 Kb/s.
I don't know what is the problem or the effect you listen.

Eretria-chan
27th May 2007, 19:15
Oops, I messed up the names... Now it's right, though.
96 kbps for 6 channels... the wonder of AAC!
If one listens closely to the "It's been too long," you'll hear a difference (especially on the word "long"). Listen closely... I really can't describe what it is that's wrong, but I can clearly hear it. Like it's some noise or some other thing that's destroying the audio. And it isn't the bitrate, as I explained...

tebasuna51
27th May 2007, 19:40
If one listens closely to the "It's been too long," you'll hear a difference (especially on the word "long"). Listen closely... I really can't describe what it is that's wrong, but I can clearly hear it. Like it's some noise or some other thing that's destroying the audio. And it isn't the bitrate, as I explained...
Maybe, there are a strange change in volume at that point.
What is your process?
- Decoder
- DSP functions
- Encoder

Eretria-chan
27th May 2007, 19:47
DirectShowSource --> CT aacPlus @ 96 kbps / Joint Stereo. No DSP functions. Delay: 2000 ms

Bigmango
27th May 2007, 21:01
According to my tests, the best quality is achieved with the nero encoder. Go for quality based, but if you really want to reach a certain bitrate then use abr with 2 pass encoding (-br -2pass). Nero is the only aac encoder than does 2 pass. I get awesome results at lower bitrates.

For 2 pass you will need to save your audio to a temporary wav file; nero can only use 2 pass encoding on wav files.

There is a nice gui for the nero encoder:
http://forum.doom9.org/showthread.php?t=110970

tebasuna51
27th May 2007, 21:44
DirectShowSource --> CT aacPlus @ 96 kbps / Joint Stereo. No DSP functions. Delay: 2000 ms
We need know the DirectShow filter decoder and settings.
For instance ffdshow audio decoder can include many DSP functions to change the sound, and is obvious the sound is, at least, amplified.

I recommend you use NicDtsSource instead ffdshow to decode the dts extracted from the matroska container and apply a Normalize before encode.

I can't reproduce the volume differences detected using NicDtsSource - Normalize - CT aacPlus HE 96 Kb/s

Eretria-chan
29th May 2007, 11:35
I think that did the trick...
One less obstacle to go!
:thanks:

shon3i
29th May 2007, 12:49
According to my tests, the best quality is achieved with the nero encoder. Go for quality based, but if you really want to reach a certain bitrate then use abr with 2 pass encoding (-br -2pass). Nero is the only aac encoder than does 2 pass. I get awesome results at lower bitrates.IMHO, CT does better job for multichannel encodings, better bitrate distribution beetwen channels especialy for SL/SR/LFE, where nero don't care too much on lower rate.

96kbps HE-AAC CT is good option for long moves and 1cdrips. Very good even very great quality for that bitrate with 15khz of bandwith.