View Full Version : AAC vs. DTS for 5.1 Audio
Seraphic-
3rd August 2008, 01:12
Hi,
Was interested to know, how AAC (640kbps) compare to DTS (1.536MB/s) for 5.1 audio? Looked around Google and found lots of AC3vsDTS, but not really anything for AACvsDTS.
Thanks
lexor
3rd August 2008, 02:03
With bitrates that high, I'd be very impressed if you will be able to tell the difference. So it all depends on your objective/situation. In fact you will need very good ears, and a sound setup of which I would be very envious, to tell either one of them from lossless source.
If you have a lossless source (DTS-HD or DD TrueHD) and you want to pick which format to convert to, then compatibility with your playback setup should be your first concern (i.e. if you want it to play in stand alone, and it doesn't support AAC, you can't really go AAC rout).
If you are thinking of saving some space by converting DTS to AAC, aside from usual warning that transcoding between lossy formats will always reduce quality (though again at your bitrate, you are unlikely to notice), I think you should be safe transcoding.
Seraphic-
3rd August 2008, 02:41
With bitrates that high, I'd be very impressed if you will be able to tell the difference. So it all depends on your objective/situation. In fact you will need very good ears, and a sound setup of which I would be very envious, to tell either one of them from lossless source.
If you have a lossless source (DTS-HD or DD TrueHD) and you want to pick which format to convert to, then compatibility with your playback setup should be your first concern (i.e. if you want it to play in stand alone, and it doesn't support AAC, you can't really go AAC rout).
If you are thinking of saving some space by converting DTS to AAC, aside from usual warning that transcoding between lossy formats will always reduce quality (though again at your bitrate, you are unlikely to notice), I think you should be safe transcoding.
In this case, the raw format is 24Bit-48kHz uncompressed WAV files. So it would be from lossless to AAC or DTS. That is why I was interested in how they match up vs. each other at max bit-rates.
When I tried a test encode the AAC was peaked out/much loade/worse then the DTS. But I think this is because DTS has audio/volume normalizing from AC3Filter and the AAC does not.
lexor
4th August 2008, 01:09
Yeah, the playback setup does come into play when comparing different formats. Also make sure you are using a fairly modern encoder for AAC, not old faad/faac builds. Nero and Apple provide the current top of the line encoders for AAC these days.
Seraphic-
9th August 2008, 02:38
Thanks for the help. Going to go with DTS at 1.536MB/s after running some comparison tests. (AC3 at 640kpbs sounded good too)
Searched the audio forum here and after going back several months, came up with a few guides that give a really damn good sounding up-mix of Stereo to 5.1 multichannel.
Sample is below if anyone wants take a listen. :devil:
Download (http://www.temp.seraphicgate.com/51Test3.dts)
shon3i
10th August 2008, 12:55
DTS is obsolete, but is usefull if you goning to force compatability, aslo AAC is superior, LC-AAC @ 640kbps must make full transparency, even on 448kbps where show more quality than classic AC3@448
Seraphic-
10th August 2008, 14:29
DTS is obsolete, but is usefull if you goning to force compatability, aslo AAC is superior, LC-AAC @ 640kbps must make full transparency, even on 448kbps where show more quality than classic AC3@448
Thing is with AAC, I'm not sure how to combine the six mono WAV files (with correct channel layout) into one six channel multichannel WAV for the MeGUI Nero AAC encoder. Do you know how to do that?
With AC3/DTS, it is easier because it asks for the six mono WAV files for each channel.
tebasuna51
10th August 2008, 16:28
Thing is with AAC, I'm not sure how to combine the six mono WAV files (with correct channel layout) into one six channel multichannel WAV for the MeGUI Nero AAC encoder. Do you know how to do that?
With AC3/DTS, it is easier because it asks for the six mono WAV files for each channel.
The channel map order for a wav 6 channel is always:
FL-FR-FC-LF-BL-BR
You can use WaveWizard to make the phisical wav file or you can use any AviSynth method (BeHappy, MeGUI, Wavi, Bepipe, SoundOut,...) to merge the channels with this merge.avs file:
fl = WavSource("G:\channelFL.wav")
fr = WavSource("G:\channelFR.wav")
fc = WavSource("G:\channelC.wav")
lf = WavSource("G:\channelLFE.wav")
sl = WavSource("G:\channelSL.wav")
sr = WavSource("G:\channelSR.wav")
MergeChannels(fl, fr, fc, lf, sl, sr)
Example:
wavi merge.avs - | neroaacenc -ignorelength -q 0.5 -if - -of output.m4a
Seraphic-
10th August 2008, 19:18
Thanks tebasuna51, that worked using Nero in MeGUI.
I had problems with playback of the .m4a version though, so encoded again with .mp4 and it played.
Also, how does Variable Bitrate compare to Adaptive Bitrate?
shon3i
10th August 2008, 19:37
.m4a and .mp4 extensions are same, just rename.
VBR produce same effect like CRF in x264, if you don't care about final size, let's encoder to use much bitrate how he think is need.
Seraphic-
10th August 2008, 19:50
.m4a and .mp4 extensions are same, just rename.
VBR produce same effect like CRF in x264, if you don't care about final size, let's encoder to use much bitrate how he think is need.
Ah, I see. But what was odd though is that 0.5 was larger in file size and sounded better then 1.0.
Isn't the further right you move the drag bar, the higher quality?
Also, I wonder how many people with 5.1 speakers have support for 5.1 ACC vs. 5.1 AC3/DTS.
~bT~
11th August 2008, 02:39
^ not in neroaacenc afaik.
raquete
11th August 2008, 03:10
Sample is below if anyone wants take a listen. :devil:
Seraphic,the sound is too loud,clipping and with distortions. :(
Seraphic-
11th August 2008, 06:39
^ not in neroaacenc afaik.
Where you talking about the VBR setting? 0.5 - 1.0?
Seraphic,the sound is too loud,clipping and with distortions. :(
What are you using for playback? What channels are too loud? Distortions from clipping?
If by clipping you mean, the "when the signal is restricted by the range of a chosen representation", I recall it was well within the audio range limit when exporting after upmixing.
http://upload.wikimedia.org/wikipedia/commons/thumb/4/4e/Clipping.svg/800px-Clipping.svg.png
madshi
11th August 2008, 08:30
Ah, I see. But what was odd though is that 0.5 was larger in file size and sounded better then 1.0.
"1.0" is not really supported by the Nero encoder. It's seen as "0.0", I believe. I think the Nero encoder doesn't expect a "1.something". So use "0.99" if you want max quality.
raquete
12th August 2008, 07:30
What are you using for playback? What channels are too loud? Distortions from clipping?
If by clipping you mean, the "when the signal is restricted by the range of a chosen representation", I recall it was well within the audio range limit when exporting after upmixing.
http://upload.wikimedia.org/wikipedia/commons/thumb/4/4e/Clipping.svg/800px-Clipping.svg.png
Seraphic-
happens the same using a.audition or media player classic.
you can use a.audition and load your sample file in edit view,select one by one and play.
or in multitrack view,click "insert"-audio" and load your sample.
now you have 6 tracks in the multitrack view: L,C,R,SL,SR and LFE.
play and use "S" for "solo" to hear each individual track.
Channels C and LFE are ok.
L,R,LS and RS are too loud and clipping,out of range(~39 seconds after start)
you told as that your file is raw format is 24Bit-48kHz uncompressed WAV files.
as the idea is to help,can you host(rapidshare) this WAV that was used to do the sample? (round 01:27.125 time)
Seraphic-
12th August 2008, 19:29
Seraphic-
happens the same using a.audition or media player classic.
you can use a.audition and load your sample file in edit view,select one by one and play.
or in multitrack view,click "insert"-audio" and load your sample.
now you have 6 tracks in the multitrack view: L,C,R,SL,SR and LFE.
play and use "S" for "solo" to hear each individual track.
Channels C and LFE are ok.
L,R,LS and RS are too loud and clipping,out of range(~39 seconds after start)
you told as that your file is raw format is 24Bit-48kHz uncompressed WAV files.
as the idea is to help,can you host(rapidshare) this WAV that was used to do the sample? (round 01:27.125 time)
Used Audition and the V.I STEREO TO 5.1 CONVERTER for the upmix process.
Though I don't recall the audio levels peaking/clipping out in track view.
However, it was a few days ago that I did that upmix, so I don't remember for sure.
Here is another WAV version of that track.
Opened in Audition and the audio levels are within range as seen below.
Download WAV (http://www.temp.seraphicgate.com/upmix.rar)
http://www.temp.seraphicgate.com/upmix.jpg
raquete
14th August 2008, 16:09
Seraphic-,
i have 3 explanations:
1- when i wrote that the sound was too loud,clipping and with distortions was used the sample 5.1(51Test3) that you posted.
this happens because your source wav. have distortions and somewhere when you convert and extract channels,the levels encrease too much and "clip" before you encode the final sample.
2- i open the the wav(upmix.wav) that you send in audition and is exactly as your screenshot.
the sound don't clip and it's not too loud but is full of distortions(no quality).seems that was too high and was normalized or amplified to -1.6dB but still have distortions.
3- knowing that the source is not so high and is not clipping but have distortions(no quality) i download the wav. sample(upmix.rar) from your last post and did the job.
result using your stereo source:
the sound stay with distortions (no quality) as the source but don't clip because is not so high as your 5.1(51Test3).
have different kind of separations between channels with better definition in each....i mean...
please,read all again because is hard for me explain it all,english is not my mother language. :p
http://rapidshare.com/files/137184045/03_Seraphic.dts.html
:thanks:
lexor
14th August 2008, 16:48
That or maybe you are using an old ffdshow or current MPC built in DTS decoder, which both clip like crazy. The dts posted in #5 (while loud) doesn't clip at all for me.
raquete
14th August 2008, 17:30
That or maybe you are using an old ffdshow or current MPC built in DTS decoder, which both clip like crazy. The dts posted in #5 (while loud) doesn't clip at all for me.
read the posts.
happens the same using a.audition or media player classic..
see how to load the file in audition here:
http://forum.doom9.org/showpost.php?p=1169527&postcount=16
did you download the file that i posted? please do that!
load my sample and the Seraphic- sample in audition and see how clip!!!
:thanks:
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