How to decode the full information of DD+ audio track?

[ddeell72]

I want to decode 1536kbps DD+ into wav and I tried eac3to. The process was simple, using libav or Nero filter.

But there is definitely something wrong with the quality of such wavs. Opening these wavs in Adobe Audition, I can't believe they were converted from 1536kbps DD+! Just the same like the old 448kbps AC3!

Some one say that libav and Nero filter just decode the basic information of DD+ track, which means equals to old AC3.

So, is there any way to decode full information of DD+ on PC??? And how to convert it into wav? eac3to seems only apply libav, Nero and sonic filter but sonic dosen't work with DD+







In the attachment is a frequency analyzing by Adobe Audition. Very obviously high frequency over 20KHz is almost empty. I really can’t believe this wav is converted from a 1536kbps DD+ file!!!

[ddeell72]

Here is another frequency analyzing, a wav file converted from a normal 1536kbps dts track. Complete high frequency up to 22KHz!

[nautilus7]

Libav and Nero decoders eac3to use work as they should. Whatever is the difference you see is format dependent i think.

[EPiPH0NE]

Yeah, I think nautilus is right. Using eac3to as you already have will give you what you want:

eac3to input.eac3 output.wav/pcm

Acutally the best results fro DD+ I have found are:

DD+ 768k/1536k -> DTS 768k/1536k


DD+ is not all it's cracked it up to be.

[ddeell72]

Then .... DD+ is meaningless to me ....

[nautilus7]

It's just an improvement over AC3 with more channels and higher bitrate.

[ddeell72]

It's just an improvement over AC3 with more channels and higher bitrate.

Higher bitrate brought nothing useful ....... :mad:

[Blue_MiSfit]

And it's hideously expensive to encode with. Trust me, we have a license and a dedicated mac (sorry no windows encoder), and it is HORRIBLE.

No CLI (you have to manually start every single job), low speed, you name it. It's a total POS. We're dumping it in favor of Aften :)

DD+ FTL.

[madshi]

I don't think that the loss of high frequency information is caused by the codec. Probably the master didn't contain that much information to begin with, or maybe the compressionist dumbed the audio track down before encoding it. I'd suggest trying some more DD+ audio tracks before doing a final judgement...

[G_M_C]

I also think it is just the way the codec is build/intended. Frequencies above 20khz are outside the the hearing-frequencies of human beings anyway, so why keep 'em ? It could also have been the idea in the studio, they might have filtered out those frequencys during (pre)mastering. Do all DD+ tracks have a similar frequency-graph ?

Anyway: I have this question for you; If you can't hear those frequencies anyway, why are you so bothered with the lack 'em (and dont give a "hifi-pwnage-ftw" like answer plz).

It's just an improvement over AC3 with more channels and higher bitrate.

Wasn't there also a different compressor used (MLP) ?

Higher bitrate brought nothing useful ....... :mad:

Maybe not frequency-wise, but it produces more accurate reproduction and it has a higher dynamic range. Those two things should mean something usefull ...

[nautilus7]

MLP is used in Dolby TrueHD, not E-AC3.

[G_M_C]

MLP is used in Dolby TrueHD, not E-AC3.

Oh yeah, that was it ...

Thx, i've corrected my previous post (removing the "lossless" statement).

[ddeell72]

Er ....... Horrible .....

[ddeell72]

20KHz higher frequency is the most basic feature of a high quality audio file, pay attention, BASIC feature; and of course a basic requirement!

We all know that LAME contributes greatly to improve MP3 quality, mainly by increasing high frequency information and performance! SONY ATRAC3 also choose the same way, even better than LAME.

Maybe there exist other tricks, but I don’t care. I just unable to accept such terrible 20KHz+ performance. The original E-AC3 file is 1536kbps!!!!!!!!!

Watch this, a LAME encoded MP3 at 320kbps:

[ddeell72]

DOLBY does neglect the importance of 20KZ+ frequency??? No, no, no, absolutely impossible!…….

So, I still believe that problems occur during the decoding ……

[tebasuna51]

I don't think that the loss of high frequency information is caused by the codec. Probably the master didn't contain that much information to begin with, or maybe the compressionist dumbed the audio track down before encoding it. I'd suggest trying some more DD+ audio tracks before doing a final judgement...

I agree with madshi.

In my test encoding with the Aften free encoder using stereo input and bitrate 320 I obtain until 23.6 KHz without problems.
(160 Kb/s per channel, 1536/6 = 256 Kb/s)

The decoder is NicAudio (same than libav). For me the problem isn't in encoder/decoder soft.