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8th October 2005, 19:26 | #741 | Link |
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I'm using the 0.22 beta 8 version and I have some complains:
And some feature requests/sugestions:
Besides that, incredible app Kurtnoise!!! |
8th October 2005, 19:47 | #742 | Link | |||||||
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8th October 2005, 20:18 | #743 | Link | ||
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Regards. |
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9th October 2005, 00:00 | #744 | Link |
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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
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9th October 2005, 19:30 | #745 | Link |
Struggling for Perfection
Join Date: Aug 2005
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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] +--------------------- |
9th October 2005, 19:40 | #746 | Link | |
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Hi,
Quote:
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9th October 2005, 19:41 | #747 | Link | |
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10th October 2005, 02:09 | #749 | Link |
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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. |
10th October 2005, 07:00 | #750 | Link | |
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Quote:
@Caroliano : check out the BeSweet FAQ. |
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12th October 2005, 07:52 | #755 | Link |
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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. |
13th October 2005, 08:35 | #756 | Link | |
The smoker
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Quote:
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" |
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13th October 2005, 08:51 | #757 | Link |
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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. |
17th October 2005, 08:44 | #758 | Link | |
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Lame ACM vs Fraunhofer ACM
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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. |
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17th October 2005, 18:15 | #759 | Link | |
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Quote:
http://corecodec.org/frs/?group_id=4...se_id=175#r175 |
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19th October 2005, 12:39 | #760 | Link | |
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Quote:
And bear in mind that nominal bitrate and average bitrate are 2 concepts completely different. |
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