Fr4nz
24th August 2003, 16:28
Hi Danni.
Well the problem is this: when I want to make the audio from VOB to MP3 via LST file, b76 makes the job perfectly.
BUT after beta 76 -azid commandline is misteriously ignored, so, for example, if I want to encode the track in Surround 2 or I want to lower the LFE level of 3dB simply I can't do it because Besweet is ignoring azid commandline!
These are the screenshots:
Beta 76 (correct way):
http://users.libero.it/i3ltt/Cazzate%20varie/be1.JPG
Log of the right VOB->MP3 way:
BeSweet v1.5b20 by DSPguru.
--------------------------
Using VOBInput.dll v1.3 by DVD2SVCD (http://www.dvd2svcd.org)
Using hip.dll v1.19 by Myers Carpenter <myers@users.sf.net>
Using Ogg Vorbis v1.0 dlls (http://www.vorbis.com).
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using Shibatch.dll v0.24 by Naoki Shibata & DSPguru (shibatch.sourceforge.net).
Using lame_enc.dll v1.32 (8/8/2003), Engine 3.90 <http://www.mp3dev.org/>.
Logging start : 08/24/03 , 17:27:34.
E:\dvd\azid\BeSweet.exe -core( -input e:\list.lst -output e:\list.mp3 -substream 0x80 -logfilea E:\dvd\azid\BeSweet.log ) -azid( -s surround -d 2/0 -L -3db -f1 ) -ota( -hybridgain ) -ssrc( --rate 44100 ) -lame( --alt-preset standard ) -profile( fr4nz )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : e:\list.lst
[00:00:00:000] | Output: e:\list.mp3
[00:00:00:000] | Substream ID: 0x80
[00:00:00:000] | Floating-Point Process: Yes
[00:00:00:000] | PostGain normalize to : 0.97
[00:00:00:032] | A/V Delay found : 0msec
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 10.000dB, Compression: None
[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: Yes
[00:00:00:000] | Rear channels filtering: Yes
[00:00:00:000] +------ Shibatch -----
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Dest. Sample-Rate: 44.1KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +-------- LAME -------
[00:00:00:000] | 'alt standard' preset is used
[00:00:00:000] +---------------------
etc.
betas after 76 (wrong way):
http://users.libero.it/i3ltt/Cazzate%20varie/be2.JPG
Log of wrong way:
BeSweet v1.5b20 by DSPguru.
--------------------------
Using VOBInput.dll v1.3 by DVD2SVCD (http://www.dvd2svcd.org)
Using hip.dll v1.19 by Myers Carpenter <myers@users.sf.net>
Using Ogg Vorbis v1.0 dlls (http://www.vorbis.com).
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using Shibatch.dll v0.24 by Naoki Shibata & DSPguru (shibatch.sourceforge.net).
Using lame_enc.dll v1.32 (8/8/2003), Engine 3.90 <http://www.mp3dev.org/>.
Logging start : 08/24/03 , 17:17:10.
E:\dvd\azid\BeSweet.exe -core( -input e:\list.lst -output e:\list.mp3 -substream 0x80 -logfilea E:\dvd\azid\BeSweet.log ) -ota( -hybridgain ) -ssrc( --rate 44100 ) -lame( --alt-preset standard )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : e:\list.lst
[00:00:00:000] | Output: e:\list.mp3
[00:00:00:000] | Substream ID: 0x80
[00:00:00:000] | Floating-Point Process: Yes
[00:00:00:000] | PostGain normalize to : 0.97
[00:00:00:032] | A/V Delay found : 0msec
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 10.000dB, Compression: None
[00:00:00:000] | LFE levels: To LR -INF, To LFE 0.0dB <-- **ARGH!**
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: Yes
[00:00:00:000] | Rear channels filtering: No <-- **argh!!**
[00:00:00:000] +------ Shibatch -----
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Dest. Sample-Rate: 44.1KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +-------- LAME -------
[00:00:00:000] | 'alt standard' preset is used
[00:00:00:000] +---------------------
[00:00:54.458] W7: Downmix overflow (1: +0dB)
[00:00:54.656] W7: Downmix overflow (1: +0.6dB)
etc...
As you can see again from the logfile azid commandline isn't added and so I don't have any control over the downmixing! I hope this will be fixed in next betas!
Luckily direct AC3->MP3 transcoding isn't affected by this bug!
Well the problem is this: when I want to make the audio from VOB to MP3 via LST file, b76 makes the job perfectly.
BUT after beta 76 -azid commandline is misteriously ignored, so, for example, if I want to encode the track in Surround 2 or I want to lower the LFE level of 3dB simply I can't do it because Besweet is ignoring azid commandline!
These are the screenshots:
Beta 76 (correct way):
http://users.libero.it/i3ltt/Cazzate%20varie/be1.JPG
Log of the right VOB->MP3 way:
BeSweet v1.5b20 by DSPguru.
--------------------------
Using VOBInput.dll v1.3 by DVD2SVCD (http://www.dvd2svcd.org)
Using hip.dll v1.19 by Myers Carpenter <myers@users.sf.net>
Using Ogg Vorbis v1.0 dlls (http://www.vorbis.com).
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using Shibatch.dll v0.24 by Naoki Shibata & DSPguru (shibatch.sourceforge.net).
Using lame_enc.dll v1.32 (8/8/2003), Engine 3.90 <http://www.mp3dev.org/>.
Logging start : 08/24/03 , 17:27:34.
E:\dvd\azid\BeSweet.exe -core( -input e:\list.lst -output e:\list.mp3 -substream 0x80 -logfilea E:\dvd\azid\BeSweet.log ) -azid( -s surround -d 2/0 -L -3db -f1 ) -ota( -hybridgain ) -ssrc( --rate 44100 ) -lame( --alt-preset standard ) -profile( fr4nz )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : e:\list.lst
[00:00:00:000] | Output: e:\list.mp3
[00:00:00:000] | Substream ID: 0x80
[00:00:00:000] | Floating-Point Process: Yes
[00:00:00:000] | PostGain normalize to : 0.97
[00:00:00:032] | A/V Delay found : 0msec
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 10.000dB, Compression: None
[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: Yes
[00:00:00:000] | Rear channels filtering: Yes
[00:00:00:000] +------ Shibatch -----
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Dest. Sample-Rate: 44.1KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +-------- LAME -------
[00:00:00:000] | 'alt standard' preset is used
[00:00:00:000] +---------------------
etc.
betas after 76 (wrong way):
http://users.libero.it/i3ltt/Cazzate%20varie/be2.JPG
Log of wrong way:
BeSweet v1.5b20 by DSPguru.
--------------------------
Using VOBInput.dll v1.3 by DVD2SVCD (http://www.dvd2svcd.org)
Using hip.dll v1.19 by Myers Carpenter <myers@users.sf.net>
Using Ogg Vorbis v1.0 dlls (http://www.vorbis.com).
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using Shibatch.dll v0.24 by Naoki Shibata & DSPguru (shibatch.sourceforge.net).
Using lame_enc.dll v1.32 (8/8/2003), Engine 3.90 <http://www.mp3dev.org/>.
Logging start : 08/24/03 , 17:17:10.
E:\dvd\azid\BeSweet.exe -core( -input e:\list.lst -output e:\list.mp3 -substream 0x80 -logfilea E:\dvd\azid\BeSweet.log ) -ota( -hybridgain ) -ssrc( --rate 44100 ) -lame( --alt-preset standard )
[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : e:\list.lst
[00:00:00:000] | Output: e:\list.mp3
[00:00:00:000] | Substream ID: 0x80
[00:00:00:000] | Floating-Point Process: Yes
[00:00:00:000] | PostGain normalize to : 0.97
[00:00:00:032] | A/V Delay found : 0msec
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 10.000dB, Compression: None
[00:00:00:000] | LFE levels: To LR -INF, To LFE 0.0dB <-- **ARGH!**
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: Yes
[00:00:00:000] | Rear channels filtering: No <-- **argh!!**
[00:00:00:000] +------ Shibatch -----
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Dest. Sample-Rate: 44.1KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +-------- LAME -------
[00:00:00:000] | 'alt standard' preset is used
[00:00:00:000] +---------------------
[00:00:54.458] W7: Downmix overflow (1: +0dB)
[00:00:54.656] W7: Downmix overflow (1: +0.6dB)
etc...
As you can see again from the logfile azid commandline isn't added and so I don't have any control over the downmixing! I hope this will be fixed in next betas!
Luckily direct AC3->MP3 transcoding isn't affected by this bug!