eac3to - audio conversion tool

[Boulder]

Have you tried updating eac3to for starters? You are using an old version, the latest one is v3.27.

[Anakunda]

No yet, I noticed there's new version just now. I'll made a rip with new e3t and will see..

[Anakunda]

Wov it works with 3.27 now why the temporary wav file for 2hour 8 channel audio track only has about 8GiB when the same 6 channel temporary wav from same track had almost 20gigs with ffmpeg? Does Arscoft decode lossy? This is what mediainfo writes about it:
Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Bit rate mode : Constant
Bit rate : 9 216 Kbps
Channel(s) : 8 channels
Channel positions : Front: L C R, Side: L R, Back: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits

And why the encoded track is so quiet? This in log:

eac3to v3.27
command line: D:\media\eac3to\eac3to.exe Source.dtshd stdout.wav -normalize
------------------------------------------------------------------------------
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48kHz)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
Decoding with ArcSoft DTS Decoder...
Applying RAW/PCM delay...
Writing WAV...
Creating file "stdout.pass1.wav"...
The original audio track has a constant bit depth of 24 bits.

Can this be fixed please?

[rapscallion]

Originally Posted by Anakunda

HI! I have finally managed to make working ArcSoft DTS decoder with eac3to but to my surprise the result is unusable.

You don't say which version of AS decoder you're using.

Those files, from Ver v1.1.0.0, work perfectly. Anything newer, not so much.

Plus, 20gb for just an audio track. That's not right, considering the complete file, with video, is 27gb !

Edit: is source.dtshd a result of encoding via DTS Master Audio Suite? If so, you have to remove the header hex by demuxing, via Tsmuxer , before running through eac3to.

[Anakunda]

Originally Posted by rapscallion

You don't say which version of AS decoder you're using.

Those files from Ver v1.1.0.0, work perfectly. Anything newer, not so much.

Plus, 20gb for just an audio track. That's not right !

Yes I got such a sizes with stadard dts decoder, don't know where's the difference. Maybe it was floating point while AS does 24 bit fixedpoint? I don't remember the std decoder output params already.

And where can I get the old AS decoder files which "work perfectly"? I extracted the mines from Totalmediatheatre 6.0.1.119 installation. Are these ok?

Originally Posted by rapscallion

Edit: is source.dtshd a result of encoding via DTS Master Audio Suite? If so, you have to remove the header hex by demuxing, via Tsmuxer , before running through eac3to.

No this was extracted right from bluray disc by eac3to. Do I need to remove the header anyway?

[rapscallion]

Originally Posted by Anakunda

No this was extracted right from bluray disc by eac3to. Do I need to remove the header anyway?

No, however, I would rename it to "dts"

PM sent

[rapscallion]

I just realized you were decoding to wav/pcm, so 8gb would not be unusual.

What are you trying to accomplish ?

[Sparktank]

Originally Posted by Anakunda

I extracted the mines from Totalmediatheatre 6.0.1.119 installation. Are these ok?

For ArcSoft 1.1.0.0, the "dtsdecoderdll.dll" has two different dates you should watch out for.

(From TMT 2.1.6.120)
Good: v1.1.0.0 25/04/2008
Bad: v1.1.0.0 21/04/2008

More info here:
http://forum.doom9.org/showthread.php?p=1266679#post1266679

[DarkSpace]

Originally Posted by Anakunda

now why the temporary wav file for 2hour 8 channel audio track only has about 8GiB when the same 6 channel temporary wav from same track had almost 20gigs with ffmpeg?

I just did some rough calculations, but (8 GB / 24 bit) * 64 bit equals roughly 21 GB, so I'd guess that ffmpeg decodes the lossy DTS core track at 64 bit floating point precision whereas ArcSoft decodes the lossless DTS-HD track at its specified precision of 24 bit. Comparing file sizes is not always a reliable way to find out which file is better, it only serves to show which file holds more (theoretical) information.

[Anakunda]

I guessed that libavcodec uses floating point too, this may explain the huge filesize difference. But I still get awfully sounding conversion on output with

eac3to Source.dtshd stdout.wav -normalize | qaac --ignorelength -o Destination.m4a -

whichever ArcSoft files I use (now testing ArcSoft package 1.1.0.0)
What I don't understand if I open the dtshd file in foobar2000 using eac3to as decoder wrapper, the file plays very good, no annoying hiss and not quiet even in 7.1 layout. Simply no artifacts. Why the conversion is so poor, I use the same command line???

[rapscallion]

As I asked above, just what is it that you're trying to accomplish????

@Sparktank, he has v1.1.0.0 25/04/2008

[Anakunda]

I want to convert dts hd 8 channels track to aac 8 channels track

[rapscallion]

So, I assume, that you can play the BD on an Apple device.

I use Arcsoft Media Coverter for that. One step process and works every time.

Any further with eac3to, maybe someone else can chime in.

[Anakunda]

Oh yes I dont have apple device .
Maybe someone else can explain why the 8channels aac sound so distorted.

[Boulder]

I have sometimes had problems with outputting stuff if ffdshow is set to output 32bit floating point audio, there's static and distortion. Have you checked that the problem is not in the decoder of the result instead of something else. Maybe you could put a sample somewhere so we can test it too.

[DarkSpace]

Originally Posted by Anakunda

Maybe someone else can explain why the 8channels aac sound so distorted.

As mentioned in the comment by Boulder, you need to make sure that isn't just your decoder making things sound ugly. Aside from that, I propose that you separate your command line into two steps and use an intermediate WAV file and play that back to verify that it's not eac3to that's making your audio sound ugly (remove the -full switch if you don't have enough free space).

eac3to "Source.dtshd" "Intermediate.wav" -normalize -full
qaac --ignorelength -o "Destination.m4a" "Intermediate.wav"

Also, I don't know how qaac works, so make sure that it's getting the input it expects (e.g. don't give it floating point data when it expects integer).

[rapscallion]

Originally Posted by Anakunda

Oh yes I dont have apple device .
Maybe someone else can explain why the 8channels aac sound so distorted.

OK, you've really got me curious, what's the point/purpose of doing this conversion?

[Chumbo]

I have a few dts files I'm converting to ac3 but the ArcSoft decoder is reporting an error. What are the best ways to get the file converted using an alternate method? For now, I'm basically checking if a failure occurs on the default setting, then use -sonic and if that fails, use -nero. Is this a good approach? Seems to be working fine. Am I using the recommended next-in-line to the ArcSoft decoder by going to sonic first and then nero? Thanks.

[Furiousflea]

Originally Posted by Chumbo

I have a few dts files I'm converting to ac3 but the ArcSoft decoder is reporting an error. What are the best ways to get the file converted using an alternate method? For now, I'm basically checking if a failure occurs on the default setting, then use -sonic and if that fails, use -nero. Is this a good approach? Seems to be working fine. Am I using the recommended next-in-line to the ArcSoft decoder by going to sonic first and then nero? Thanks.

Yea, Arcsoft -> Sonic -> Nero

(Best avoid nero altogether for any DTS-HD)

[Chumbo]

Originally Posted by Furiousflea

Yea, Arcsoft -> Sonic -> Nero

(Best avoid nero altogether for any DTS-HD)

Thanks, that's what I thought. Thankfully, not a dts-hd track.