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Old 2007-08-04, 22:41   Link #41
edogawaconan
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Quote:
Originally Posted by Zanejin View Post
Just in case you aren't already, go ahead and begin using BeLight (version 0.22 RC1 is safe to use), as it's meant to be a replacement for the original BeSweet GUI. Then, you can go to the "WAV/PCM" tab and select "16 Bits 5.1 Wave" under "Output Channels"—you probably used the equivalent of "16 Bits (Six) Mono Waves", or maybe something more bizarre.
I tried that before (with -6chwav) and the resulting file is no sound LFE channel and same sounds (which is from LFE channel) on SR dan SL.
And is there no way to make it just 4 channel? And from what I've read yesterday (on Hydrogenaudio, wiki, and doom9), multichannel support seems to be still in mess...
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Old 2007-08-05, 12:09   Link #42
ArchMageZeratuL
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Quote:
Originally Posted by edogawaconan View Post
Don't worry, I properly tagged my vorbis track in mkv with 'ogg vorbis'
That's still incorrect: when you multiplex an Ogg Vorbis file into a Matroska Video, you just have Vorbis, since it's no longer in an Ogg container. Calling it "Ogg Vorbis" when it's in mkv is similar to saying that the video in your Matroska is "AVI XviD" or "MP4 H.264".
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Old 2007-08-05, 13:57   Link #43
Zanejin
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Quote:
Originally Posted by edogawaconan View Post
I tried that before (with -6chwav) and the resulting file is no sound LFE channel and same sounds (which is from LFE channel) on SR dan SL.
And is there no way to make it just 4 channel? And from what I've read yesterday (on Hydrogenaudio, wiki, and doom9), multichannel support seems to be still in mess...
I don't know how to create a 4-channel PCM file, but 5.1 should be equivalent as long as the channels are correctly mapped and fed data. Regarding your problem with -6chwav, I think you might have enabled some unwanted Azid switches (enabled by default if unspecified), namely -d, which is likely misconfigured due to the input being 4-channel and the output being 5.1, or -s.

I'm reading all of this off of the command-line reference by Brother John.
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Old 2007-08-05, 14:50   Link #44
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Quote:
Originally Posted by xat View Post
... is this meant to be a troll?
...No?

I'm fine with .mkv but i prefer .avi or somenthing else i can play with winamp.
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Old 2007-08-05, 14:57   Link #45
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Quote:
Originally Posted by Cheezy View Post
I'm fine with .mkv but i prefer .avi or somenthing else i can play with winamp.


1) get out of this thread (which is about AUDIO FORMATS) and go troll playback help instead
2) read http://www.cccp-project.net/wiki/ind...title=Winamp_5 and get your "problem" solved
3) stop posting
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Old 2007-08-05, 22:26   Link #46
edogawaconan
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Quote:
Originally Posted by ArchMageZeratuL View Post
That's still incorrect: when you multiplex an Ogg Vorbis file into a Matroska Video, you just have Vorbis, since it's no longer in an Ogg container. Calling it "Ogg Vorbis" when it's in mkv is similar to saying that the video in your Matroska is "AVI XviD" or "MP4 H.264".
is that it? thanks for the correction

Quote:
Originally Posted by Zanejin View Post
I don't know how to create a 4-channel PCM file, but 5.1 should be equivalent as long as the channels are correctly mapped and fed data. Regarding your problem with -6chwav, I think you might have enabled some unwanted Azid switches (enabled by default if unspecified), namely -d, which is likely misconfigured due to the input being 4-channel and the output being 5.1, or -s.

I'm reading all of this off of the command-line reference by Brother John.
doesn't seem to work
Code:
I:\aaa>d:\encode\besweet\BeSweet -core( -input "aaa AC3 T01 3_1ch 448Kbps DELAY 0ms.ac3" -output "aaawavv2.wav" -6chwav ) -azid( -d 3/1 )
BeSweet v1.5b31 by DSPguru.
--------------------------
Using azid.dll v1.9 (b922) by Midas  (midas@egon.gyaloglo.hu).

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] |  Input : aaa AC3 T01 3_1ch 448Kbps DELAY 0ms.ac3
[00:00:00:000] |  Output: aaawavv2.wav
[00:00:00:000] |  Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] |  Input Channels Mode: 3/1, Bitrate: 448kbps
[00:00:00:000] |  Total Gain: 0.000dB, Compression:  None
[00:00:00:000] |  LFE levels: To LR -INF, 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: No
[00:00:00:000] |  Rear channels filtering: No
[00:00:00:000] |  Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[01:02:45:664] Conversion Completed !
[01:02:45:664] Actual Avg. Bitrate  : 1140kbps
[00:02:27:000] <-- Transcoding Duration


Visit DSPguru's Homepage at :
                                http://DSPguru.doom9.net/

I:\5cm>
same results with before. Or that is the correct result? weird though with number of channels increased from 4 to 5 (number of non-silent channels)

edit: after reading wiki some more, I find this:
- 3-channel stereo (Left, Center, Right)
- 3-channel stereo with mono surround (Left, Center, Right, Surround)
how can I know which format I get?
If it's the first then the fourth channel in ac3 file is LFE and if it's the later then the fourth channel is indeed SR/SL. *even more confused now *

disclaimer: my audio system here is 2.1 which means I totally rely on audacity and wave file specification in doing this all. And I don't know how's the normal form of LFE channel...

Last edited by edogawaconan; 2007-08-05 at 22:37.
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Old 2007-08-06, 01:34   Link #47
jpwong
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Hmm, which do I prefer? I'd probably say MP3, though I haven't had a lot of experiance with the other ones since most people use mp3.

Though I have to admit, I'm not too picky, just as long as the volume got normalized before or during encoding. I really detest it when I have to boost the audio level by over 500% to bring it up to something that you can listen to properly.
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Old 2007-08-06, 02:02   Link #48
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Personally, I'd prefer AC3, but that's usually only from DVD source anyway.

Otherwise, I'm not too fussy. MP3 is good if I'm putting it on a portable device, since it saves me from recoding and losing quality, else AAC.
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