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 20th July 2011, 19:02   #11141  |  Link
tebasuna51
Moderator
 
tebasuna51's Avatar
 
Join Date: Feb 2005
Location: Spain
Posts: 6,890
Nope (use the search, this question was answered already in this thread).
__________________
BeHappy, AviSynth audio transcoder.
tebasuna51 is offline  
Old 21st July 2011, 03:51   #11142  |  Link
lostclusters
Registered User
 
Join Date: Apr 2009
Posts: 19
I put in a search of "6.1" and it told there are no results. Searching is fun with a crippled search function. It said the same thing when attempting to search on DTS-MA 6.1.

Last edited by lostclusters; 21st July 2011 at 03:55.
lostclusters is offline  
Old 21st July 2011, 05:28   #11143  |  Link
Sparktank
47.952fps@71.928Hz
 
Sparktank's Avatar
 
Join Date: Mar 2011
Posts: 940
Use Google with search functions

Quote:
dts-ma 6.1 site:forum.doom9.org
Most search engines in forums don't allow you to search things less than 4 characters.
"quotes" are usually required.

Google has a better chance, since it's diverse.
dts-ma will show you dts-hd ma, dtsma, dts ma, etc with Google.
With the forum search, it looks for exactly what you type in.
__________________
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 22nd July 2011, 03:10   #11144  |  Link
lostclusters
Registered User
 
Join Date: Apr 2009
Posts: 19
That was a very good tip, TYVM!
lostclusters is offline  
Old 26th July 2011, 03:55   #11145  |  Link
nibus
Telewhining
 
Join Date: Mar 2010
Posts: 272
quick question:

If I transcode DTS-MA to FLAC, and later decide to transcode that FLAC back to DTS-MA with Surcode, will the end file be basically identical to the first, or will the -core or other information be lost forever?
nibus is offline  
Old 26th July 2011, 20:43   #11146  |  Link
mastrboy
Registered User
 
Join Date: Sep 2008
Posts: 365
Is it possible to disable that annoying "Pling" sound when done with a file?
And is it possible to skip the 1-pass wav file and encode directly to in example a m4a aac file or is that wav file needed?
mastrboy is offline  
Old 26th July 2011, 20:55   #11147  |  Link
nurbs
Registered User
 
Join Date: Dec 2005
Posts: 1,460
Quote:
Originally Posted by mastrboy View Post
Is it possible to disable that annoying "Pling" sound when done with a file?
Rename or delete the file "success.wav" in your eac3to directory.

Quote:
Originally Posted by mastrboy View Post
And is it possible to skip the 1-pass wav file and encode directly to in example a m4a aac file or is that wav file needed?
I haven't tried, but adding "-no2ndpass" to your commandline might do it. The wav is written so that if a second pass is necessary, which isn't always known before the first pass is complete, you don't need to decode the audio again. You might have audiable problems if you use that option.
nurbs is offline  
Old 2nd August 2011, 11:03   #11148  |  Link
asarian
Registered User
 
Join Date: May 2005
Posts: 1,462
I'm having trouble converting the following stream to LPCM:

DTS-HD Master Audio English 1064 kbps 1.0 / 48 kHz

The resultant LPCM file is playing way too fast. I take it because there's no official LPCM 1.0? In any case, how do I convert this to a normal, playable, LPCM track?

Thanks.
__________________
Gorgeous, delicious, deculture!
asarian is offline  
Old 2nd August 2011, 22:13   #11149  |  Link
iSeries
Registered User
 
Join Date: Jan 2009
Posts: 625
About the eac3to 6.1 DTS-HD downmixing 'bug', if I output to individual WAV files could someone let me know how to mix the back center channel with the surround left/right channels manually/correctly please?

Could I do it with eac3to, eg: eac3to rightsurround.wav+backcenter.wav output.wav

Last edited by iSeries; 2nd August 2011 at 22:22.
iSeries is offline  
Old 3rd August 2011, 09:56   #11150  |  Link
tebasuna51
Moderator
 
tebasuna51's Avatar
 
Join Date: Feb 2005
Location: Spain
Posts: 6,890
Nope, you can use an audio editor (like Audacity or others), a command line tool like sox, or Avisynth (with BeHappy, wavi, ...) to do a mix like:

BL = 0.5858 x SL + 0.4142 x BC
BR = 0.5858 x SR + 0.4142 x BC

The recommended .avs' can be something like:

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.4142)
Normalize()

To obtain BL channel, and for BR channel:

sr = WavSource("D:\...\SideRigth.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sr, bc, 0.5858, 0.4142)
Normalize()
__________________
BeHappy, AviSynth audio transcoder.
tebasuna51 is offline  
Old 3rd August 2011, 17:06   #11151  |  Link
iSeries
Registered User
 
Join Date: Jan 2009
Posts: 625
tebasuna51, thanks! The resulting .wav files seem to be 32 bit float though (from BeHappy), what do I need to add to the scripts to get 24 bit?

Thanks again!

Last edited by iSeries; 3rd August 2011 at 17:28.
iSeries is offline  
Old 3rd August 2011, 19:09   #11152  |  Link
b66pak
Registered User
 
b66pak's Avatar
 
Join Date: Aug 2008
Location: The Land Of Dracula (Romania - EU)
Posts: 934
add:

Code:
ConvertAudioTo24bit()
_
__________________
if you ask a question and somebody give you the correct answer don't forget to leave a "thank you" note...
Visit The Land Of Dracula (Romania - EU)!
b66pak is offline  
Old 3rd August 2011, 20:03   #11153  |  Link
iSeries
Registered User
 
Join Date: Jan 2009
Posts: 625
Thanks guys!

...is it ok to reduce the bit depth of the new .wav's from 32 bit float back to 24 bit with eac3to? or is it more recommended to do this with BeHappy?

Last edited by iSeries; 3rd August 2011 at 20:16.
iSeries is offline  
Old 3rd August 2011, 21:55   #11154  |  Link
iSeries
Registered User
 
Join Date: Jan 2009
Posts: 625
Also, do I need to normailize ALL the wav's that will make up the 5.1, or just the two created with BeHappy?
iSeries is offline  
Old 3rd August 2011, 22:54   #11155  |  Link
tebasuna51
Moderator
 
tebasuna51's Avatar
 
Join Date: Feb 2005
Location: Spain
Posts: 6,890
Well, nothing is perfect. You can't obtain always the same global audio volume with 7 speakers than with 6.

If you want preserve the balance volume betwen front and back channels maybe you need make this mix:

BL = SL + 0.7071 x BC
BR = SR + 0.7071 x BC

preserve the 32 bit float and check if there are values greater than 0 dB, if yes you need use the normalize only over BL-BR and lose some back volume, if not you don't need the normalize and the balance volume is ok.
__________________
BeHappy, AviSynth audio transcoder.
tebasuna51 is offline  
Old 3rd August 2011, 23:05   #11156  |  Link
iSeries
Registered User
 
Join Date: Jan 2009
Posts: 625
Thanks, like this?

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.7071)
iSeries is offline  
Old 4th August 2011, 09:34   #11157  |  Link
kypec
User of free A/V tools
 
kypec's Avatar
 
Join Date: Jul 2006
Location: SK
Posts: 826
Quote:
Originally Posted by iSeries View Post
Thanks, like this?

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.7071)
Almost correct but the SL / SR channels must be factored by 1, not 0.5858.
Replace your last line with this:
Code:
MixAudio(sl, bc, 1.0, 0.7071)
kypec is offline  
Old 4th August 2011, 13:29   #11158  |  Link
iSeries
Registered User
 
Join Date: Jan 2009
Posts: 625
Thanks! Sorry one last question lol, how can I check if clipping has occured? Am I right in thinking that if I open the wav in Audacity and nothing goes beyond 1.0 or -1.0 then there is no clipping?

Also, I don't really understand any of this so a brief explanation on the difference between

MixAudio(sl, bc, 0.5858, 0.4142)

and

MixAudio(sl, bc, 1.0, 0.7071)

...would be greatly appreciated

Last edited by iSeries; 4th August 2011 at 14:11.
iSeries is offline  
Old 4th August 2011, 14:58   #11159  |  Link
tebasuna51
Moderator
 
tebasuna51's Avatar
 
Join Date: Feb 2005
Location: Spain
Posts: 6,890
Quote:
Originally Posted by iSeries View Post
..., how can I check if clipping has occured? Am I right in thinking that if I open the wav in Audacity and nothing goes beyond 1.0 or -1.0 then there is no clipping?
You can use Audacity -> View -> Show clipping

MixAudio(sl, bc, 0.5858, 0.4142)
make:
BL = 0.5858 x SL + 0.4142 x BC
the audio never can be greater than 1.0 (0 dB), without clip

and

MixAudio(sl, bc, 1.0, 0.7071)
make:
BL = 1.0 x SL + 0.7071 x BC
the audio can be greater than 1.0, then you can have clipping.
__________________
BeHappy, AviSynth audio transcoder.

Last edited by tebasuna51; 4th August 2011 at 15:01.
tebasuna51 is offline  
Old 4th August 2011, 16:26   #11160  |  Link
iSeries
Registered User
 
Join Date: Jan 2009
Posts: 625
Many thanks. I used MixAudio(sl, bc, 1.0, 0.7071) for LOTR Fellowship 1/2, and this produced no clipping. The Two Towers 1/2 and Return of the King 1/2 did produce clipping using that so for these I re-did with MixAudio(sl, bc, 0.5858, 0.4142). I'm guessing thats all ok (Audacity shows no clipping for all wavs)
iSeries 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 05:47.


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