Space@pe
9th October 2002, 11:59
Hello all you audiophiles :D . ok, now iīve double checked the questions I had a few weeks ago and I hope these are more to the point...the link:
http://forum.doom9.org/showthread.php?s=&threadid=34953
(all questions Is for BeSweet)!
In question 2 In the above link I asked "How does the -ota( -G max ) differ from the -ota( -g max ) and which one of these give the best quality?...I got that cleared up by reading about the OTA Switches in BeSweet's CommandLine's Reference
http://besweet.notrace.dk/cli.html
there stand and I quote
"-g GAIN
-------
Default: 1.0 (or 0db)
This option controls the overall track PreGain. The value can be given
in db's (by specifying "db" after the argument) or a positive numerical
value. Examples: -g -3db, -g 5.3, -g 6db
-g max, is a special switch for finding maximum gain. (Normalize)
-norm VALUE
-----------
Default: not used.
This option is an alternative to "-g max", and can be used to set gain of a track to a lower value than its maximum.
"-norm 90%", for example, will normalize a track to 90% of its maximal gain.
"-norm 100%" is equivalent to "-g max".
-G GAIN
-------
Default: 1.0 (or 0db)
This option controls the overall track PostGain of encoded MP3/OGG. The value can be given
in db's (by specifying "db" after the argument) or a positive numerical
value. Examples: -G -3db, -G 5.3, -G 6db
-G max, is a special switch for finding maximum gain. (Normalize)
Important :
------------
"-G" switch differs from "-g" in the fact that "-G" will assert gain on a MP3/OGG after encoding prcoess was finished, while "-g" will assert the gain before encoding.
that's why "-g max" will take almost 50% time longer than "-G max".
about the quality difference between the two methods, here's a quote of 'tangent' :
[quote]
"I believe that the benefits of postgain are more than worthwhile. Quality wise, it would not be the same as pregain, and one cannot absolutely say if it would be worse, better or similar. I can say that for most people out there the difference is inaudible, and in my opinion the quality should be better. This, plus the fact that postgain method is faster is enough for me to choose the postgain method.
One last final advantage of post-gain is that you can guarantee that with post-gain applied, you can absolutely guarantee that the decoded playback will not clip and will have the highest volume possible (within 1.5dB). With pre-gain, there is no such guarantee, and you cannot be sure if the decode playback will clip or have the highest volume possible."
[end of quote]
In question 4 I wanted to know what would be the best mode to use wenn making a 192kb MP3 soundtrack (stereo or joint stereo)...and I found that Joint-stereo is the default mode for stereo files with VBR when -v is more than 4 or fixed bitrates of 160kbs or less (+with fixed bitrates of 128kb or less). At higher fixed bitrates or higher VBR settings, the default is stereo.
then I had this question "what Is cool about using VBR wenn your quality goes wildly from one bitrate to another (example--->192-224)?wouldnt that effect the soundquality or Is there something about the whole thing that Iīve missed?"?"
It was obvious that there was something i didnīt understand and Itīs still a little confusing, but at least now I know that
LAME has two types of variable bitrate: ABR and VBR.
ABR is the type of variable bitrate encoding usually found in other
MP3 encoders, Vorbis and AAC. The number of bits is determined by
some metric (like perceptual entropy, or just the number of bits
needed for a certain set of encoding tables), and it is not based on
computing the actual encoding/quantization error. ABR is still in
beta, but it should always give results equal or better than CBR:
VBR is a true variable bitrate mode which bases the number of bits for
each frame on the measured quantization error relative to the
estimated allowed masking. VBR is currently under heavy development.
Right now it can often result in too much compression. I do not
recommend using VBR. But if you must, you should at least use a
minimum bitrate of 112 kbps. This will let LAME increase the bitrate
for difficult-to-encode frames, but prevent LAME from being too
aggressive for simple frames.
and:
The use of -B is NOT RECOMMENDED. A 128kbs CBR bitstream, because of the bit reservoir, can actually have frames which use as many bits as a 320kbs frame. VBR modes minimize the use of the bit reservoir, and thus need to allow 320kbs frames to get the same flexibility as CBR streams.
Then I asked In question 9 "Scale Input (Multiply PCM Data)..what Is It?And why Is It sometimes sat to 1 by default?
I found that It was a little like Dynamic Compression only better...hmm but otherwise very little info on that :(
I also think that the "Use Dialog Normalization"/"Error Protection" option Is ok to use!?...:D maybe..I dont think that about the "Force Minimum allowed Bitrate" option.
things that still are unclear to me!!
1. What Is the difference between Azidīs Gain from the "overall output configuration" option and the Boost options from the "BoostCli.exe"?
5. what about "Present/Alt Present...what are they good for, does they change the bitrate..or dont they?
So wenn I use VBR one thing pussles me. why can I only choose the quality/Noise Sharping of the MP3 compression wenn using VBR?
10. Wenn you set the "Gain" to 10db inside Azid I find that the "Auto Gain" Inside OTA still Is marked!...whatīs the deal here?And which of these options get encoded first and how does that effect the soundtrack?
and the CLI again :)
"" -core( -input "" -output "C:\@#Ītest.mp3" -logfile "C:\Whatever\BeSweet1.4\BeSweet.log" ) -azid( -n1 -s surround2 -d 2/2 -c normal -L -3db ) -ota( -G max ) -boost( /b2=5 ) -lame( -m s -q 0 -v --vbr-new -V 0 -b 192 -B 192 ) -profile( spaceape )
THX :D
http://forum.doom9.org/showthread.php?s=&threadid=34953
(all questions Is for BeSweet)!
In question 2 In the above link I asked "How does the -ota( -G max ) differ from the -ota( -g max ) and which one of these give the best quality?...I got that cleared up by reading about the OTA Switches in BeSweet's CommandLine's Reference
http://besweet.notrace.dk/cli.html
there stand and I quote
"-g GAIN
-------
Default: 1.0 (or 0db)
This option controls the overall track PreGain. The value can be given
in db's (by specifying "db" after the argument) or a positive numerical
value. Examples: -g -3db, -g 5.3, -g 6db
-g max, is a special switch for finding maximum gain. (Normalize)
-norm VALUE
-----------
Default: not used.
This option is an alternative to "-g max", and can be used to set gain of a track to a lower value than its maximum.
"-norm 90%", for example, will normalize a track to 90% of its maximal gain.
"-norm 100%" is equivalent to "-g max".
-G GAIN
-------
Default: 1.0 (or 0db)
This option controls the overall track PostGain of encoded MP3/OGG. The value can be given
in db's (by specifying "db" after the argument) or a positive numerical
value. Examples: -G -3db, -G 5.3, -G 6db
-G max, is a special switch for finding maximum gain. (Normalize)
Important :
------------
"-G" switch differs from "-g" in the fact that "-G" will assert gain on a MP3/OGG after encoding prcoess was finished, while "-g" will assert the gain before encoding.
that's why "-g max" will take almost 50% time longer than "-G max".
about the quality difference between the two methods, here's a quote of 'tangent' :
[quote]
"I believe that the benefits of postgain are more than worthwhile. Quality wise, it would not be the same as pregain, and one cannot absolutely say if it would be worse, better or similar. I can say that for most people out there the difference is inaudible, and in my opinion the quality should be better. This, plus the fact that postgain method is faster is enough for me to choose the postgain method.
One last final advantage of post-gain is that you can guarantee that with post-gain applied, you can absolutely guarantee that the decoded playback will not clip and will have the highest volume possible (within 1.5dB). With pre-gain, there is no such guarantee, and you cannot be sure if the decode playback will clip or have the highest volume possible."
[end of quote]
In question 4 I wanted to know what would be the best mode to use wenn making a 192kb MP3 soundtrack (stereo or joint stereo)...and I found that Joint-stereo is the default mode for stereo files with VBR when -v is more than 4 or fixed bitrates of 160kbs or less (+with fixed bitrates of 128kb or less). At higher fixed bitrates or higher VBR settings, the default is stereo.
then I had this question "what Is cool about using VBR wenn your quality goes wildly from one bitrate to another (example--->192-224)?wouldnt that effect the soundquality or Is there something about the whole thing that Iīve missed?"?"
It was obvious that there was something i didnīt understand and Itīs still a little confusing, but at least now I know that
LAME has two types of variable bitrate: ABR and VBR.
ABR is the type of variable bitrate encoding usually found in other
MP3 encoders, Vorbis and AAC. The number of bits is determined by
some metric (like perceptual entropy, or just the number of bits
needed for a certain set of encoding tables), and it is not based on
computing the actual encoding/quantization error. ABR is still in
beta, but it should always give results equal or better than CBR:
VBR is a true variable bitrate mode which bases the number of bits for
each frame on the measured quantization error relative to the
estimated allowed masking. VBR is currently under heavy development.
Right now it can often result in too much compression. I do not
recommend using VBR. But if you must, you should at least use a
minimum bitrate of 112 kbps. This will let LAME increase the bitrate
for difficult-to-encode frames, but prevent LAME from being too
aggressive for simple frames.
and:
The use of -B is NOT RECOMMENDED. A 128kbs CBR bitstream, because of the bit reservoir, can actually have frames which use as many bits as a 320kbs frame. VBR modes minimize the use of the bit reservoir, and thus need to allow 320kbs frames to get the same flexibility as CBR streams.
Then I asked In question 9 "Scale Input (Multiply PCM Data)..what Is It?And why Is It sometimes sat to 1 by default?
I found that It was a little like Dynamic Compression only better...hmm but otherwise very little info on that :(
I also think that the "Use Dialog Normalization"/"Error Protection" option Is ok to use!?...:D maybe..I dont think that about the "Force Minimum allowed Bitrate" option.
things that still are unclear to me!!
1. What Is the difference between Azidīs Gain from the "overall output configuration" option and the Boost options from the "BoostCli.exe"?
5. what about "Present/Alt Present...what are they good for, does they change the bitrate..or dont they?
So wenn I use VBR one thing pussles me. why can I only choose the quality/Noise Sharping of the MP3 compression wenn using VBR?
10. Wenn you set the "Gain" to 10db inside Azid I find that the "Auto Gain" Inside OTA still Is marked!...whatīs the deal here?And which of these options get encoded first and how does that effect the soundtrack?
and the CLI again :)
"" -core( -input "" -output "C:\@#Ītest.mp3" -logfile "C:\Whatever\BeSweet1.4\BeSweet.log" ) -azid( -n1 -s surround2 -d 2/2 -c normal -L -3db ) -ota( -G max ) -boost( /b2=5 ) -lame( -m s -q 0 -v --vbr-new -V 0 -b 192 -B 192 ) -profile( spaceape )
THX :D