Welcome to Doom9's Forum, THE in-place to be for everyone interested in DVD conversion.

Before you start posting please read the forum rules. By posting to this forum you agree to abide by the rules.

 

Go Back   Doom9's Forum > General > Audio encoding

Closed Thread
 
Thread Tools Search this Thread Display Modes
Old 8th March 2013, 16:34   #12161  |  Link
Anakunda
Registered User
 
Join Date: Jan 2010
Posts: 330
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?

Last edited by Anakunda; 8th March 2013 at 16:51.
Anakunda is offline  
Old 8th March 2013, 17:38   #12162  |  Link
rapscallion
NY Frame of Mind
 
rapscallion's Avatar
 
Join Date: Dec 2005
Location: L.I.,NY
Posts: 586
Quote:
Originally Posted by Anakunda View Post
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.
__________________
"Talk to me Goose"

Last edited by rapscallion; 9th March 2013 at 15:42.
rapscallion is offline  
Old 8th March 2013, 17:45   #12163  |  Link
Anakunda
Registered User
 
Join Date: Jan 2010
Posts: 330
Quote:
Originally Posted by rapscallion View Post
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?

Quote:
Originally Posted by rapscallion View Post
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?

Last edited by Anakunda; 8th March 2013 at 17:48.
Anakunda is offline  
Old 8th March 2013, 17:57   #12164  |  Link
rapscallion
NY Frame of Mind
 
rapscallion's Avatar
 
Join Date: Dec 2005
Location: L.I.,NY
Posts: 586
Quote:
Originally Posted by Anakunda View Post
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
__________________
"Talk to me Goose"
rapscallion is offline  
Old 8th March 2013, 20:29   #12165  |  Link
rapscallion
NY Frame of Mind
 
rapscallion's Avatar
 
Join Date: Dec 2005
Location: L.I.,NY
Posts: 586
I just realized you were decoding to wav/pcm, so 8gb would not be unusual.

What are you trying to accomplish ?
__________________
"Talk to me Goose"

Last edited by rapscallion; 9th March 2013 at 15:42.
rapscallion is offline  
Old 9th March 2013, 06:26   #12166  |  Link
Sparktank
47.952fps@71.928Hz
 
Sparktank's Avatar
 
Join Date: Mar 2011
Posts: 940
Quote:
Originally Posted by Anakunda View Post
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
__________________
Win10 (x64) build 19041
NVIDIA GeForce GTX 1060 3GB (GP106) 3071MB/GDDR5 | (r435_95-4)
NTSC | DVD: R1 | BD: A
AMD Ryzen 5 2600 @3.4GHz (6c/12th, I'm on AVX2 now!)
Sparktank is offline  
Old 9th March 2013, 12:56   #12167  |  Link
DarkSpace
Registered User
 
Join Date: Oct 2011
Posts: 204
Quote:
Originally Posted by Anakunda View Post
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.
DarkSpace is offline  
Old 9th March 2013, 13:11   #12168  |  Link
Anakunda
Registered User
 
Join Date: Jan 2010
Posts: 330
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
Code:
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???
Anakunda is offline  
Old 9th March 2013, 15:47   #12169  |  Link
rapscallion
NY Frame of Mind
 
rapscallion's Avatar
 
Join Date: Dec 2005
Location: L.I.,NY
Posts: 586
As I asked above, just what is it that you're trying to accomplish????

@Sparktank, he has v1.1.0.0 25/04/2008
__________________
"Talk to me Goose"

Last edited by rapscallion; 9th March 2013 at 15:54.
rapscallion is offline  
Old 9th March 2013, 15:50   #12170  |  Link
Anakunda
Registered User
 
Join Date: Jan 2010
Posts: 330
I want to convert dts hd 8 channels track to aac 8 channels track
Anakunda is offline  
Old 9th March 2013, 16:07   #12171  |  Link
rapscallion
NY Frame of Mind
 
rapscallion's Avatar
 
Join Date: Dec 2005
Location: L.I.,NY
Posts: 586
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.
__________________
"Talk to me Goose"
rapscallion is offline  
Old 9th March 2013, 16:10   #12172  |  Link
Anakunda
Registered User
 
Join Date: Jan 2010
Posts: 330
Oh yes I dont have apple device .
Maybe someone else can explain why the 8channels aac sound so distorted.
Anakunda is offline  
Old 9th March 2013, 16:14   #12173  |  Link
Boulder
Pig on the wing
 
Boulder's Avatar
 
Join Date: Mar 2002
Location: Finland
Posts: 5,718
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.
__________________
And if the band you're in starts playing different tunes
I'll see you on the dark side of the Moon...
Boulder is offline  
Old 9th March 2013, 16:52   #12174  |  Link
DarkSpace
Registered User
 
Join Date: Oct 2011
Posts: 204
Quote:
Originally Posted by Anakunda View Post
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).
Code:
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).
DarkSpace is offline  
Old 9th March 2013, 18:12   #12175  |  Link
rapscallion
NY Frame of Mind
 
rapscallion's Avatar
 
Join Date: Dec 2005
Location: L.I.,NY
Posts: 586
Quote:
Originally Posted by Anakunda View Post
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?
__________________
"Talk to me Goose"
rapscallion is offline  
Old 9th March 2013, 19:16   #12176  |  Link
Chumbo
Registered User
 
Chumbo's Avatar
 
Join Date: Feb 2005
Posts: 585
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.
__________________
Chumbo
Chumbo is offline  
Old 11th March 2013, 01:12   #12177  |  Link
Furiousflea
Registered User
 
Join Date: Aug 2006
Posts: 696
Quote:
Originally Posted by Chumbo View Post
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)
Furiousflea is offline  
Old 11th March 2013, 03:27   #12178  |  Link
Chumbo
Registered User
 
Chumbo's Avatar
 
Join Date: Feb 2005
Posts: 585
Quote:
Originally Posted by Furiousflea View Post
Yea, Arcsoft -> Sonic -> Nero

(Best avoid nero altogether for any DTS-HD)
Thanks, that's what I thought. Thankfully, not a dts-hd track.
__________________
Chumbo
Chumbo is offline  
Old 11th March 2013, 12:54   #12179  |  Link
Anakunda
Registered User
 
Join Date: Jan 2010
Posts: 330
Quote:
Originally Posted by DarkSpace View Post
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).
I did so now and got strange results.
The dtshd track reported length by MI is none. In foobar2000 it's reported only 1h02m which is even not a half of true length.

ediainfo for dtshd:
Code:
Audio
Format                                   : DTS
Format/Info                              : Digital Theater Systems
Format profile                           : MA / Core
Mode                                     : 16
Format settings, Endianness              : Big
Bit rate mode                            : Variable
Bit rate                                 : Unknown / 1 509 Kbps
Channel(s)                               : 8 channels / 6 channels
Channel positions                        : Front: L C R, Side: L R, Back: L R, LFE / Front: L C R, Side: L R, LFE
Sampling rate                            : 48.0 KHz
Bit depth                                : 24 bits
Compression mode                         : Lossless / Lossy
Now I decoded it to wav using eac3to source.dtshd intermediate.wav -normalize -full

and got this 22Gigs big WAV file. If I open it in foobar2000 it seems to play all channels properly but is reported length only 10m29s

Code:
Audio
Format                                   : PCM
Format profile                           : Float
Codec ID                                 : 00001000-0000-0300-8000-00AA00389B71
Codec ID/Hint                            : IEEE 
Duration                                 : 2h 6mn
Bit rate mode                            : Constant
Bit rate                                 : 24.6 Mbps
Channel(s)                               : 8 channels
Channel positions                        : Front: L C R, Side: L R, Back: L R, LFE
Sampling rate                            : 48.0 KHz
Bit depth                                : 64 bits
Stream size                              : 21.8 GiB (100%)
Also trying to convert this to aac results in only 10m29s track
Anakunda is offline  
Old 11th March 2013, 15:09   #12180  |  Link
DarkSpace
Registered User
 
Join Date: Oct 2011
Posts: 204
Quote:
Originally Posted by Anakunda View Post
The dtshd track reported length by MI is none. In foobar2000 it's reported only 1h02m which is even not a half of true length.
I have no idea exactly what foobar2000 is doing, but I have read several times already that guessing the duration of an Elementary Stream with variable bitrate can be tricky (there is no container to tell you the true duration, after all).

Quote:
Originally Posted by Anakunda View Post
Now I decoded it to wav using eac3to source.dtshd intermediate.wav -normalize -full and got this 22Gigs big WAV file. If I open it in foobar2000 it seems to play all channels properly but is reported length only 10m29s
Some players have problems with WAV files larger than 2 GB. If your player doesn't play the whole file, try using the .w64 format for conversion instead of .wav (no worries, both formats are lossless). If your player plays the whole file but displays a wrong duration, you don't have to convert to .w64, although of course it shouldn't harm things.

Quote:
Originally Posted by Anakunda View Post
Also trying to convert this to aac results in only 10m29s track
Same as above, WAV files greater than 2 GB may be problematic. However, before I mention that you may need to look for a switch to convert the whole file to AAC, make sure the --ignorelength parameter is present. Alternatively, you can also use the .w64 approach mentiond above, if qaac can handle .w64 files.

Last edited by DarkSpace; 11th March 2013 at 21:18.
DarkSpace is offline  
Closed Thread

Tags
eac3to

Thread Tools Search this Thread
Search this Thread:

Advanced Search
Display Modes

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off

Forum Jump


All times are GMT +1. The time now is 16:45.


Powered by vBulletin® Version 3.8.11
Copyright ©2000 - 2024, vBulletin Solutions Inc.