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Old 16th December 2001, 19:07   #1  |  Link
phynix
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AZID - Cracks in the output by using dynamic compression

hi!
i have a little problem with the dynamic compression-method of azid. when it's activated, the encoded files contain cracks - not many, ~5 in the whole file i would say, but when i encode it without this option, the output is just perfect. i have "auto find maximum gain" activated, too. what do i do wrong?? if you can help me, i would be pleased to hear from you!

p.s.: excuse my bad english
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Old 16th December 2001, 19:44   #2  |  Link
MaTTeR
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Hmm...I've always used Azids Dynamic Compression and never heard any pops. We really need more info from you before we can help.

What GUI are you using for the encode process? DD's GUI? Commandline?

Your encoding to MP3? If so, what version of LAME? What LAME settings are you using?

Need more info! :-)
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Old 17th December 2001, 14:06   #3  |  Link
phynix
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ok - here is the full story:

first i rip the vob-files with smartripper and extract the ac3-stream with vstrip -> with only the demux-function activated and only one ac3-substream selected. then i start dds azid/lame GUI [v 0.3] and encode the ac3-stream into wav-format [azid v1.7.1]. the only activated functions are: LFE to LR channels: -3db; dynamic compression: normal; auto find maximum gain; print current settings/Bitstream information/Downmix matrix; Quit on Bitstream Error. i later encode the file to mp3, but the errors occure in the wav-file too, so it can't be lame that causes the problem, also it can't be vstrip, because when i encode the ac3-file without the use of the dynamic compression, there aren't any errors in the output.

i hope this info is enough, if you need more, plz ask me again !
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Old 17th December 2001, 14:29   #4  |  Link
DSPguru
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Re:

Quote:
Originally posted by phynix
auto find maximum gain
try setting a manual gain value.
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Old 17th December 2001, 16:02   #5  |  Link
DarkAvenger
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This bug has already been reported to Midas some weeks ago. Unfortunately he didn't fix it yet.
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Old 17th December 2001, 17:41   #6  |  Link
ChristianHJW
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I never had probs like this, maybe because i output into a 24 bit format from Azid ??? Or maybe i'm already deaf ....

BTW : Is Azid's 24 Bit output floating point ? Can LAME read 24 bit floating point ( it certainly can read Azid/SSRC 24 bit output ) ?

Really looking forward to DSP Gurus 32 bit floating point implementation from Azid to Lame. I just hope he doesnt forget about SSRC, for obvious reasons ....

Not that i would believe 32 bits resolution are necessary at all .... but if you dont have to deal with intermediate WAV files, thanks to BeSweet, you dont have to care about file size at all .... so i fully agree with DSPGuru's point of view :

Why not using 32 bit ??? Quality freak as i am, using 32 bit handling for resampling and/or normalizing gives me this undescribable feeling that the absolute optimum was achieved ..... how wonderful this is .... and pls. LigH, dont be cruel now and remind me about the brutal things that are done in most studios ...
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Old 17th December 2001, 17:42   #7  |  Link
phynix
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i tried to set a gain-level manually, but there were exactly the same errors as with auto find-function. i have varied almost all settings in the used programms, but the result didn't get better. hopefully midas will fix this bug soon...

thx for your help!
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Old 17th December 2001, 17:47   #8  |  Link
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hmm... when i select 24 or 32 bit output, wmp can't read the file and when i encode it to mp3, the file contains very loud murmur.
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Old 17th December 2001, 18:20   #9  |  Link
MaTTeR
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Re:

Quote:
Originally posted by phynix
hmm... when i select 24 or 32 bit output, wmp can't read the file and when i encode it to mp3, the file contains very loud murmur.
I use 24bit output with 2 passes to find the gain. I then encode it with Lame without any issues. Don't use WiMP, use another player for your audio/video needs. I'm pretty sure LAME can't except 32bit WAV files so stick with 24bit if you have the space :-)
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Old 17th December 2001, 19:59   #10  |  Link
phynix
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Do you mean, you encode it with azid to a 24bit-floating-point-wav an then perform the 2-pass processing with SSRC?? i didn't use SSRC by now, i will try that!

by the way, what is WaveBooster for?? doesn't it do something very similar to dynamic compression in azid??
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Old 17th December 2001, 21:21   #11  |  Link
MaTTeR
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Re:

Quote:
Originally posted by phynix
Do you mean, you encode it with azid to a 24bit-floating-point-wav an then perform the 2-pass processing with SSRC?? i didn't use SSRC by now, i will try that!

by the way, what is WaveBooster for?? doesn't it do something very similar to dynamic compression in azid??
Thats correct. Use DD's GUI, think the latest is beta 4. You might also try HeadAC3he 0.15. I've been using it more freqently as the speed and output quality is great.

I've never used WaveBooster so I can't really help you there.
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Old 18th December 2001, 11:01   #12  |  Link
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i don't understand it, a bitrate higher than 16 just don't work ! it is not only, that wmp can't play it, but the output is at very bad quality - as i said - very loud murmur and the original sound of the input is hardly recognizable, like recorded in a snowstorm. i have resampled the file to 16bit, but the quality didn't become better... this question is probably stupid, but do i need a special codec for floating-point wavs??
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Old 18th December 2001, 11:44   #13  |  Link
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... what OS are you using ?? Its in fact possible that your OS doesnt support 24 bit floating point WAVs to be stored on the HDD.

Try latest BeSweet and BeSweet GUI, according to DSPGuru it will process sound internally in 32 bit mode ( highest possible quality, even a bit exagerated maybe, but why not ) and doesnt create intermediate WAV files, the blocks are fed from Azid directly to Lame or SSRC/Lame ...

@Midas :

If you're still around Mate, we know you didnt have the time to look into the problem of overflow errors for the auto find max gain feature recently, but wouldn it be possible to simply reduce the amplification factor for 2nd pass by lets say 3 dB ( factor found in 1st pass divided by 1.414 ) if 24 bits output was chosen in Azid ?? This would remove the problems instantaneously and with 24 bit output we're still on the safe side in terms of SNR ....

@DSPGuru :

Do you have access to Azid code ??
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Old 18th December 2001, 12:05   #14  |  Link
phynix
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im using winXP pro. the methode you described sounds interresting - i'll try it.

thx for the hint!
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Old 18th December 2001, 12:57   #15  |  Link
phynix
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the result is the same - when i open the mp3 with winamp, it tries to write a wav-file to c:, the lenght of it is ~3x as long as the input-file, there isn't any sound and the file is "played" fast forward. i tried to encode the ac3 to a 32 bit wav [with dyn compression] resampled it to a 16 bit wav [2-pass] and let lame encode it to a vbr mp3.

i don't know what's still wrong - perhaps winXp is such a os that doesn't support it...?
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Old 18th December 2001, 15:26   #16  |  Link
MaTTeR
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Re:

Quote:
Originally posted by phynix
the result is the same - when i open the mp3 with winamp, it tries to write a wav-file to c:, the lenght of it is ~3x as long as the input-file, there isn't any sound and the file is "played" fast forward. i tried to encode the ac3 to a 32 bit wav [with dyn compression] resampled it to a 16 bit wav [2-pass] and let lame encode it to a vbr mp3.

i don't know what's still wrong - perhaps winXp is such a os that doesn't support it...?
I use XP just like others on here and it works fine. Your WinAmp settings are screwed up. Somehow you have it setup to decode an MP3 to a WAV file on your hard disk. Go to the WinAmp prefs and select the output and then make sure that DiskWriter isn't highlighted. You should have it set to Wave Out.
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Old 18th December 2001, 17:33   #17  |  Link
phynix
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thx, i think i got it finally!
your right, it was winamp that caused that problem with the >16 bit wav-playback.

just one more question, then i stop bugging you: could you tell me, if these setting are allright??

DDs BeSweet GUI [0.4 b]:

Azid [1.7.1]
++++++++++++

LFE 2 LR: -3 db
dynamic compression: normal
auto find max. gain
file type: 32 bits floating-point wav

SSRC [1.28]
+++++++++++

Sampling rate to: 44100hz
Perform 2 pass processing

Lame [3.89 b]
+++++++++++++

Mode: Joint Stereo
VBR: Old Routine; qual.:8; disable writing xing tag; minimum allowed br: 0; max. allowed br: 190
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Old 18th December 2001, 17:40   #18  |  Link
MaTTeR
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Your welcome. Glad your up and running now.

Your settings look good for BeSweet but..I'm not so sure about the LAME/MP3 encode settings your using. Maybe someone else can advise you on these but they look low quality to me. Do a listening test and see what you think.
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Old 18th December 2001, 18:00   #19  |  Link
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... matter is correct, V8 in VBR will result in very low bitrate files ( about 100 kbps ), R3MIX experts recommend standard ABR with default ATH instead of VBR for these low bitrates ...
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Old 18th December 2001, 18:09   #20  |  Link
DSPguru
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Re:

Quote:
Originally posted by ChristianHJW
@DSPGuru :

Do you have access to Azid code ??
i wish i had..
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