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
Register FAQ Calendar Today's Posts Search

Reply
 
Thread Tools Search this Thread Display Modes
Old 8th October 2005, 19:26   #741  |  Link
Caroliano
Registered User
 
Join Date: Feb 2005
Location: Săo Paulo, Brazil
Posts: 392
I'm using the 0.22 beta 8 version and I have some complains:
  • The "qualidade" (quality) word of portuguese translation don't fit in it's space. This seen to be the only word in all languages that has this problem. You can fix it in someway? At least in MP3 tab for example, where space is not an issue, but it is cuted in some way.
  • In ogg, the nominal bitrate is not corrected when I change the Sampling rate.
  • In Mp3 tab, would be good if it show the exact bitrate of a certain point, in the same way as in winamp' AAC encoder "tab". And why I can't set 120kbps for example in Mp3's CBR mode? Is CBR mode more limited than ABR mode?
  • In Mp2 tab, in the right side of "encoding mode", the box seens to be a little cuted.

And some feature requests/sugestions:
  • When you click in the slide bar, outside the "pointer" (I don't know I call it), would be good if the pointer advances more, to the next defaut bitrate for example (96,112,128,160, and so on). This would economize a lot of time for me.
  • Make the queue space resizeable. That way I don't need to use the scroll bar when I'm in 800x600 resolution mode. Also, it can remember the last place and size it was when closed, because it is annoying have to drag it from the right boton of the screen every time I open it.
  • Make it able to receive correctly the "open with" calls. When I select an .wav file to be opened always by Belight when I double click it for example, I want to Belight open with this .wav file already in the queue, but it open w/o nothing.

Besides that, incredible app Kurtnoise!!!
Caroliano is offline   Reply With Quote
Old 8th October 2005, 19:47   #742  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
Quote:
Originally Posted by Caroliano
The "qualidade" (quality) word of portuguese translation don't fit in it's space. This seen to be the only word in all languages that has this problem. You can fix it in someway?
I see...yes for sure.

Quote:
Originally Posted by Caroliano
In ogg, the nominal bitrate is not corrected when I change the Sampling rate.
First, Ogg IS a container. Second, nominal bitrate is completely different of the Sampling-Rate. There is no interaction between these 2 settings.

Quote:
Originally Posted by Caroliano
In Mp3 tab, would be good if it show the exact bitrate of a certain point, in the same way as in winamp' AAC encoder "tab".
maybe yes, maybe no...

Quote:
Originally Posted by Caroliano
And why I can't set 120kbps for example in Mp3's CBR mode? Is CBR mode more limited than ABR mode?
BeSweet restriction if I remember correctly...



Quote:
Originally Posted by Caroliano
When you click in the slide bar, outside the "pointer" (I don't know I call it), would be good if the pointer advances more, to the next defaut bitrate for example (96,112,128,160, and so on). This would economize a lot of time for me.
For what kind of output format ?

Quote:
Originally Posted by Caroliano
Make the queue space resizeable. That way I don't need to use the scroll bar when I'm in 800x600 resolution mode. Also, it can remember the last place and size it was when closed, because it is annoying have to drag it from the right boton of the screen every time I open it.
it was already in my to-do list.

Quote:
Originally Posted by Caroliano
Make it able to receive correctly the "open with" calls. When I select an .wav file to be opened always by Belight when I double click it for example, I want to Belight open with this .wav file already in the queue, but it open w/o nothing.
ok...good idea.
Kurtnoise is offline   Reply With Quote
Old 8th October 2005, 20:18   #743  |  Link
Caroliano
Registered User
 
Join Date: Feb 2005
Location: Săo Paulo, Brazil
Posts: 392
Quote:
First, Ogg IS a container. Second, nominal bitrate is completely different of the Sampling-Rate. There is no interaction between these 2 settings.
Oops, I meant Vorbis. And at least in case of Vorbis quality setings, the sampling rate counts. At 48KHz the average bitrate of q-2 is 32kbps, at 8KHz the average bitrate is 10kbps, a big diference. See OggDropXpd for precise nominal bitrates for each sample rate.
Quote:
For what kind of output format ?
I only see it in Mp2 and AC3 tabs. So, for all others, excluding WAV of course.

Regards.
Caroliano is offline   Reply With Quote
Old 9th October 2005, 00:00   #744  |  Link
ricknks2000
Registered User
 
Join Date: Mar 2005
Posts: 1
I am new to belight and want to use o.21 for i need to go from vob-wav. i have been thru this forum and i have seen questions like mine but no answer so please forgive a novice. when i open belight the screen is too large and the bottom third is cut off from my use. what can i do please,and thank you for what i hope to be a great tool. rick
ricknks2000 is offline   Reply With Quote
Old 9th October 2005, 19:30   #745  |  Link
wdmalik
Struggling for Perfection
 
wdmalik's Avatar
 
Join Date: Aug 2005
Posts: 108
can you get the parameters info displayed ..
like follwoing .... it can be optional

Code:
[00:00:00:000] |  Floating-Point Process: Yes
[00:00:00:000] |  PostGain normalize to : 0.97
[00:00:00:000] +-------- AZID -------                              
[00:00:00:000] |  Input Channels Mode: 3/2, Bitrate: 384kbps
[00:00:00:000] |  Output Stereo mode: Dolby surround 2 compatible
[00:00:00:000] |  Total Gain: 10.000dB, Compression:  Normal 
[00:00:00:000] |  LFE levels: To LR -3.0dB, To LFE 0.0dB
[00:00:00:000] |  Center   mix level: BSI
[00:00:00:000] |  Surround mix level: BSI
[00:00:00:000] |  Dialog normalization: -4dB
[00:00:00:000] |  Rear channels filtering: Yes
[00:00:00:000] |  Source Sample-Rate: 48.0KHz
[00:00:00:000] +-------- BOOST ------
[00:00:00:000] |  Algorithm by : Dg
[00:00:00:000] |  Boost Factor : 4.0
[00:00:00:000] |  Limit Factor : 0.95
[00:00:00:000] +-------- LAME -------
[00:00:00:000] |  Bitrate method  : VBR (OLD)   
[00:00:00:000] |  VBR Quality     :   2
[00:00:00:000] |  MP3 Min bitrate : 128
[00:00:00:000] |  MP3 Max bitrate : 320
[00:00:00:000] |  Channels Mode   : Joint Stereo
[00:00:00:000] |  Error Protection: No
[00:00:00:000] +---------------------
wdmalik is offline   Reply With Quote
Old 9th October 2005, 19:40   #746  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
Hi,

Quote:
Originally Posted by ricknks2000
when i open belight the screen is too large and the bottom third is cut off from my use. what can i do please,and thank you for what i hope to be a great tool. rick
This issue has been resolved in the 0.22 branch which don't have vob support yet... So, the best thing for you could be to extract ac3 streams and use BeLight to transcode them or try with an other tool.
Kurtnoise is offline   Reply With Quote
Old 9th October 2005, 19:41   #747  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
Quote:
Originally Posted by wdmalik
can you get the parameters info displayed ..
like follwoing .... it can be optional
by choosing the output logfile option, you can have these infos.
Kurtnoise is offline   Reply With Quote
Old 9th October 2005, 19:51   #748  |  Link
wdmalik
Struggling for Perfection
 
wdmalik's Avatar
 
Join Date: Aug 2005
Posts: 108
not logfile, i mean while the processing is been done ..
wdmalik is offline   Reply With Quote
Old 10th October 2005, 02:09   #749  |  Link
Caroliano
Registered User
 
Join Date: Feb 2005
Location: Săo Paulo, Brazil
Posts: 392
I had an error when tried to transcode an 6ch 48KHz AC3 stream to an 6ch Vorbis. I even tried to use the "output sampling rate" option, but in the log it continue reporting the same error.

Code:
BeSweet v1.5b31 by DSPguru.
--------------------------
Using azid.dll v1.9 (b922) by Midas  (midas@egon.gyaloglo.hu).
Using libVorbis.dll v1.0 ( Jul  9 2005 ) by John33 (www.inf.ufpr.br/~rja00).
Manual Dynamic-Compression algorithm by LigH (author of WaveBooster).

Logging start : 10/09/05 , 21:54:59.

BeSweet.exe -core( -input E:\Testes\Azumanga\japa.ac3 -output E:\Testes\Azumanga\japa.ogg -logfile E:\Testes\Azumanga\japa.log ) -azid( -c normal ) -ogg( -q -0.150 -6ch 1 ) -boost( /b=3 /l=0.95 ) 

[00:00:00:000] +------- BeSweet -----                          
[00:00:00:000] |  Input : E:\Testes\Azumanga\japa.ac3
[00:00:00:000] |  Output: E:\Testes\Azumanga\japa.ogg
[00:00:00:000] |  Floating-Point Process: Yes
Error 68: 6chogg only works with 48khz streams.
          Gain & Dynamic Compression should be set against azid.

Quiting...
[00:00:00:000] Conversion Completed !                                    

Logging ends : 10/09/05 , 21:54:59.
Caroliano is offline   Reply With Quote
Old 10th October 2005, 07:00   #750  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
Quote:
Originally Posted by wdmalik
not logfile, i mean while the processing is been done ..
It's possible of course but in this case we can't have/save logfile...


@Caroliano : check out the BeSweet FAQ.
Kurtnoise is offline   Reply With Quote
Old 10th October 2005, 15:04   #751  |  Link
drob
Registered User
 
Join Date: May 2004
Posts: 255
Kurt still problems with batch processing, when you throw a couple of files, BeLight marks all the job as done but actually process only the last file.
drob is offline   Reply With Quote
Old 10th October 2005, 16:12   #752  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
I know and I'm busy with my real life now. I hope to fix this this week-end.
Kurtnoise is offline   Reply With Quote
Old 10th October 2005, 19:33   #753  |  Link
drob
Registered User
 
Join Date: May 2004
Posts: 255
real life always comes first, thanks for help.
drob is offline   Reply With Quote
Old 12th October 2005, 01:09   #754  |  Link
suspiciousBob
Registered User
 
Join Date: Apr 2003
Location: London Town
Posts: 38
Will the mp3 area of BeLite be changed for LAME 3.97 and the new preferred switches as documented here?
BTW great job on this project fella.
suspiciousBob is offline   Reply With Quote
Old 12th October 2005, 07:52   #755  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
Hi,

BeLight uses already lame_enc.dll from 3.97 beta version if you use the whole package you know... Concerning the settings, I already managed these recommendations. If you look the command line, you can see that BeLight uses the recommended settings most of the time. However, there are some addition in Quality Mode caused by BeSweet restriction.

The only point which must be checked today is the MP3 presets for BeLight. Of course, I'll look into this.
Kurtnoise is offline   Reply With Quote
Old 13th October 2005, 08:35   #756  |  Link
Immi
The smoker
 
Immi's Avatar
 
Join Date: Sep 2005
Location: Kiskunfélegyháza, Hungary
Posts: 3
Quote:
Originally Posted by Kurtnoise13
Hi,

BeLight uses already lame_enc.dll from 3.97 beta version if you use the whole package you know...
Hi Kurt,

together with WMP10 (only Win XP users) everybody get the Fraunhofer MPEG Layer-3 Audio Codec Professional (it is ACM codec located at %windir%\sytem32\l3codecp.acm after installing WMP10).
Maybe ( I am not an programming expert) using of this codec also can be implemented into Besweet/Belight.
BR, Immi

Off topic PS:
Tip: Using of prof codec in any encoding programs (ie. EAC, Vdub etc.) apply the following registry modification

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows NT\CurrentVersion\Drivers32]
"msacm.l3acm"="C:\\WINDOWS\\system32\\l3codecp.acm"
Immi is offline   Reply With Quote
Old 13th October 2005, 08:51   #757  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
mmmh, it's possible I think. There is a tool which works with acm filters. But I'm pretty sure that Lame is probably more interesting than the Fhg codec. Do you perform some tests with this codec ?


Edit : here is the tool.

Last edited by Kurtnoise; 13th October 2005 at 09:01.
Kurtnoise is offline   Reply With Quote
Old 17th October 2005, 08:44   #758  |  Link
Immi
The smoker
 
Immi's Avatar
 
Join Date: Sep 2005
Location: Kiskunfélegyháza, Hungary
Posts: 3
Lame ACM vs Fraunhofer ACM

Quote:
Originally Posted by Kurtnoise13
mmmh, it's possible I think. There is a tool which works with acm filters. But I'm pretty sure that Lame is probably more interesting than the Fhg codec. Do you perform some tests with this codec ?


Edit : here is the tool.
Hi,

I look the net for this info and generally it is said that the FHF codec quite old and outdated.

At the weekend I try to check this.

The method:
So I am unable to hear any difference by my ears, consequently I had to compare the encodings sample by sample.
Source: ripped (EAC) typical music (with vocal+drums+hiss) PCMwav
Encodings: EAC wav encoding funciton using
- FraunhoferACM CBR 80 mono
- LameACM CBR 80 mono
- FraunhoferACM CBR 128
- LameACM CBR 128
- FraunhoferACM CBR 320
- LameACM CBR 320
Additionally Besweet encoding using
- Lame.dll CBR 80 mono
- Lame.dll CBR 128
- Lame.dll CBR 320

To get the differences between the original and the encoded files inverted mixing was applied (mix the encoded file into the original wav inverted).
This mixing was checked, because when the original wav is invert mixed into itself I get pure flatline(silence) as result.

Now the tricky part. To get proper difference file, the sample accurate timing is needed. Problem is that the encoding process insert silence in the begining of the audio file. So I "draw" a timer signal in the original then always cut out the not necessary silences to get the sample accurate match of original and encoded file.

For the attachments please download
http://www.mytempdir.com/211968

Attachment: the original timer signal and the timer after encoding
original - 080lame compare.jpg
Attachment: Timer signal section after application of inverted-mixing
original - 080lame difference.jpg

Results:
Codec - Encoding time - Encoded audio file size
difference file in the attachment

80kbs mono:
FHF ACM 0:19 2654892 bytes
difference original-080fhf.jpg
Lame ACM 0:12 2655574 bytes
difference original-080lame.jpg
Lame dll 0:10 2657043 bytes
difference original-080lamedll.jpg

128kbs:
FHF ACM 0:34 4241724 bytes
difference original-128fhf.jpg
Lame ACM 0:23 4249260 bytes
difference original-128lame.jpg
Lame dll 0:21 4251190 bytes
difference original-128lamedll.jpg

320 kbs:
FHF ACM 0:15 10618524 bytes
difference original-320fhf.jpg
Lame ACM 0:19 10615392 bytes
difference original-320lame.jpg
Lame dll 0:19 10627785 bytes
difference original-320lamedll.jpg

"Additional test" VBR-insane:
lame dll 0:19 8246932 bytes
difference original-vbr-insane-lamedll.jpg

Now you can see the the lost info-data is more in the Fraunhofer encoded files. Furthermore Fraunhofer only support CBR encodings.

So you are right: Lame is probably more interesting than the Fhg codec

BR, Imre

PS.: It was inresting to listen the difference file. You can clearly hear the lost audio data. It is mainly the high freqency and slighly the mid freqency part of the audio.

Last edited by Immi; 17th October 2005 at 08:56.
Immi is offline   Reply With Quote
Old 17th October 2005, 18:15   #759  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
Quote:
Originally Posted by drob
Kurt still problems with batch processing, when you throw a couple of files, BeLight marks all the job as done but actually process only the last file.
fixed...

http://corecodec.org/frs/?group_id=4...se_id=175#r175
Kurtnoise is offline   Reply With Quote
Old 19th October 2005, 12:39   #760  |  Link
Kurtnoise
Swallowed in the Sea
 
Kurtnoise's Avatar
 
Join Date: Oct 2002
Location: Aix-en-Provence, France
Posts: 5,191
Quote:
Originally Posted by Caroliano
At 48KHz the average bitrate of q-2 is 32kbps, at 8KHz the average bitrate is 10kbps, a big diference. See OggDropXpd for precise nominal bitrates for each sample rate.
Which version did you have ? coz I don't see any changes when I choose the Sampling Rate...

And bear in mind that nominal bitrate and average bitrate are 2 concepts completely different.
Kurtnoise is offline   Reply With Quote
Reply


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 18:26.


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