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Sv: Sv: [mad-user] Attenuation, clipping
by Peter Olufsen 06 Dec '00

06 Dec '00

From: Anthony Airon Oetzmann <Anthony.Oetzmann(a)epost.de> To: Peter Olufsen <po(a)dsinet.dk> Dato: 6. december 2000 01:49 Emne: Re: Sv: [mad-user] Attenuation, clipping >>1) Is it correct that the clipping appears in the decoder even though the file was encoded "correct" without clipped samples ? > >You're thinking lossless compression here, 'cause you won't get a signal >different from the original with lossless compression such as Monkeyaudio. >However, MP3 reconstructs an audio signal from less than the original signal >plus noise. Add for example one frequency x and another frequency 2*x, then >you'll have some areas that are boosted, and some that are attenuated. When 2*x >is filtered out again(or gets discarded in some places, because the encoding >algorythem thought you won't hear it) then some places will get attenuated that >were peaks and some will be boosted that weren't peaks before. And sometimes in >complex signals that are mastered to use the maximum of the 16 bit resolution on >a CD and are mastered to extreme loudness(single releases are mangled this way a >LOT), the signal's reconstruction overshoots, because the stuff that got >filtered out in the encoding process meant that the tightly controlled >frequencies of the signal will have some things missing. You get some peaks and >attenuations you previously didn't have. You can check this easily, by taking a >ripped track and using an EQ in a sample editor to EQ 100Hz by -6 dB for >example. You almost always get clipped samples, because of the great energy that >lower frequencies take and thus command more amplitude and heavier changes to >the signal. Ok, i get it ! >>3) I think it would be nice to have some form of auto-attenuation (and also auto normalize-to-peak or even amping/compressing), but to make it perfect wouldn't you have to scan all your files first, or could you make some setup for that ? >>I have testet some files and they typically have clipping about 1db, but i found one file with 6db clipping in one sample (?), and if you auto-attenuate that the result is not good, so you should have some form of setup to prevent that. >>I have also tested some auto-amp-plugin and the result wasen't good because you could hear that the sound-level was adjusted when the songs changed in level. > >I propose to set a minimum attenuation level of -2 dB. That seems reasonable for >most material. Automatic attenuation would set in with a look-ahead algorhythem, >and perhaps a good limiter could be used in cases of less than five clipped >samples. I agree. >>4) Would it be possible to use MAD's 24/32-bit output with some DSP-plugin like DFX ? >>Or do you loose the advantage from MAD when you use DSP ? > >MAD needs to do this itself. DSP plugins get the decoded signal. >The signal path is DECODER PLUGIN -> (optional DPS Plugins->) OUTPUT PLUGIN But it would seem like a good idea that MAD delivered its 32-bit output to DFX (or some other DSP-plugin) instead of MAD going 16->32->16 and DFX doing the same thing, or am i wrong here ? Peter

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Attenuation, clipping
by Rob Leslie 05 Dec '00

05 Dec '00

I just want to provide a little background for anyone confused by all this clipping and attenuation nonsense. :-) During the decoding process, PCM samples are reconstructed from MPEG audio frames. The nominal output range of the PCM samples can be considered to be a fraction number between -1.0 and +1.0. This is eventually scaled to a number with certain resolution, for example 0..255 for 8-bit output, or -32768..32767 for 16-bit output, etc. Because of the nature of the decoding process, it sometimes happens that PCM samples are calculated to have a value less than -1.0 or greater than +1.0. This isn't really an error, it's just an artifact of the lossy compression and depends a lot on the encoder and the strength of the original signal. Most decoders have little choice but to clip the output signal to the maximum representable values. This adds distortion to the signal. By "attenuating" the signal, the entire signal can be weakened proportionally so that the peaks fall exactly on the -1.0 or +1.0 full scale limits rather than beyond them. It is a digital volume adjustment to prevent clipping. Internally, MAD represents signals with about 169 dB of dynamic range and enough headroom to calculate samples 18 dB above full scale. It's therefore possible to further manipulate the signal without losing quality before it is finally scaled and dithered to the output resolution. The "Attenuate" button in the pre-release activates an attenuation filter to weaken the output signal based on the amount of clipping experienced. An automatic version of this would do the same thing as soon as clipping is detected. Cheers, -rob

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Re: [mad-user] predecoding volume scaling
by Rob Leslie 02 Dec '00

02 Dec '00

Anthony Airon Oetzmann <airon(a)gmx.net> wrote: > A fairly serious quality question here. I'd like the encoder to scale back > the volume of the output before it gets decoded. The reason is that too many > encoded files clip when decoded. I'm assuming you mean you'd like the *decoder* to scale back the volume? As soon as you have an MP3 file the encoder is out of the picture. > I'd like to suggest a way to do this without mucking about in the actual > decoding code. It's the way the Waves Renaissance EQ handles it. It actualy > calculates with a headroom and at a higher bitresolution. It is able to > accuratly tell me how many dB I'm overshooting my output. MAD already does something similar. Output samples are actually calculated internally with 29 bits of precision, with 3 extra bits of "headroom" for decoded samples that exceed full scale. > What I'd like the MAD plugin to do is have a control setting that tells it > to pull x dB out of the final signal, before it gets CUT OFF at the 0dB Full > Scale, and sent to the output plugin. I'd suggest a default setting of -2 > dB. Perhaps even a clip indicator, if this is possible to display in some > way in the Winamp window or another window(on/off switch in prefs). This might be possible. Scaling the output is easy to do, and wouldn't lose any quality because the internal samples are 32-bit values. A clip indicator might also be easy to add. The trick will be for the user to find an appropriate setting. > Some of you may not hear this revolting distortion. Well I do, and it don't > sound good folks. Some songs are mastered to already CLIP on the CD, though > they keep that clipped signal below 0dB, as not to make the CD production > plant mad(no more than 5 clipping samples on an entire CD is a RED BOOK > standard provision where clipped -> 0dB I guess). Do you hear clipping even with MAD? I ask because MAD doesn't clip the same way other decoders do. > The problem with almost any damn plugin out there is that the volume in fact > can only be changed AFTER decoding. Calculating perhaps with 24 + 2 bits > would perhaps be advisable, buy I'll leave that to you guys. I think what you mean is that the volume can typically only be changed after samples have been clipped and scaled to the output resolution (e.g. 16 bits); "decoding" in my mind doesn't include this step. The output of decoding in MAD is 32-bit samples that can be further manipulated without loss of quality before clipping, scaling, and dithering to the output resolution. If there is consensus among other MAD users, I will consider adding something like this to monitor the amount of clipping and perhaps allow the user to attenuate the signal before output samples are produced. What do people think? -- Rob Leslie rob(a)mars.org

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If anyone's interested, my review of the M Audio Audiophile 2496 card
by David Shin 02 Dec '00

02 Dec '00

With a nod to the MAD decoder. :) http://headwize.com/ubb/Forum4/HTML/000165.html

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predecoding volume scaling
by Anthony Airon Oetzmann 01 Dec '00

01 Dec '00

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 A fairly serious quality question here. I'd like the encoder to scale back the volume of the output before it gets decoded. The reason is that too many encoded files clip when decoded. I'd like to suggest a way to do this without mucking about in the actual decoding code. It's the way the Waves Renaissance EQ handles it. It actualy calculates with a headroom and at a higher bitresolution. It is able to accuratly tell me how many dB I'm overshooting my output. What I'd like the MAD plugin to do is have a control setting that tells it to pull x dB out of the final signal, before it gets CUT OFF at the 0dB Full Scale, and sent to the output plugin. I'd suggest a default setting of -2 dB. Perhaps even a clip indicator, if this is possible to display in some way in the Winamp window or another window(on/off switch in prefs). Some of you may not hear this revolting distortion. Well I do, and it don't sound good folks. Some songs are mastered to already CLIP on the CD, though they keep that clipped signal below 0dB, as not to make the CD production plant mad(no more than 5 clipping samples on an entire CD is a RED BOOK standard provision where clipped -> 0dB I guess). The problem with almost any damn plugin out there is that the volume in fact can only be changed AFTER decoding. Calculating perhaps with 24 + 2 bits would perhaps be advisable, buy I'll leave that to you guys. What do you think ? Tony Sound Engineer and Designer - - Homepage - http://members.tripod.de/airon -----BEGIN PGP SIGNATURE----- Version: PGPsdk version 1.7.1 (C) 1997-1999 Network Associates, Inc. und Tochtergesellschaften. iQA/AwUBOieqe2nRM1szI4hMEQJLswCgojZAMhMByJB81PgRiZjKkgAx2GIAoO5s p2ux2Y3nqDb3MaKKzkPczAjn =sxlQ -----END PGP SIGNATURE-----

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CoolPlayer to use the MAD decoder?
by David Shin 01 Dec '00

01 Dec '00

Has anyone heard the rumor that Niek Albers might put the MAD decoder into a future build of CoolPlayer?

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Using MAD with a 24-bit sound card
by David Shin 01 Dec '00

01 Dec '00

I've been using MAD with an M Audio Audiophile 2496 card for a week now. Here's what I've found. The oscilliscope and VU meter is useless in 24 or 32 bit resolution mode; it just constantly squiggles, even during silent passages. With my particular card, 24 bit resolution only works with a waveOut buffer of 9000 ms or more. If I use a lower buffer, the MP3 files will start "screeching", making nothing but a loud static noise if I scan through the song. However, 32 bit resolution works fine. MAD's 32-bit output is padded 24-bit, correct Rob? Well, I'm using it with my 24-bit sound card, and it works perfectly with the default waveOut settings. And like I even need to say this, MAD at 24-bit resolution blows Fraunhofer or audioEnlightenment out of the water. I'm using a Creek OBH-11 headphone amplifier mated to Sony MDR-V6 Studio Monitor headphones, and the difference is pretty stark and obvious. The MAD output sounds more natural, has better instrument decay, has more bass power, and the bass is smoother and the treble clearer. Fraunhofer sounds positively fake in comparison. Regards to you all, David

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Re: [mad-user] Some suggestions for the Winamp plug-in
by Wim 23 Nov '00

23 Nov '00

I think the ID3 tag editor is fine just the way it is with the tabs.

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Some suggestions for the Winamp plug-in
by David Shin 23 Nov '00

23 Nov '00

-quick way to strip all ID3 tags -ability to edit title formatting for MP3 files without ID3 tags (like Winamp 2.7) -show MPEG Audio info and ID3 tag info on one screen -someone with artistic talent, please make a cool MAD logo!

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MAD plug-in 0.12.2b CPU utilization
by David Shin 21 Nov '00

21 Nov '00

Hey you MAD, MAD crowd, this is just Nawhead giving you a little lowdown into the new release's speed improvement results. :) I have a Celeron 400 with Windows 98 (original). I used Another Task Manager to isolate the decoder thread and get the results. I did three test runs on different bitrate MP3's, getting the average after at least a minute of playback. [please use a fixed width font to view the following table] | 0.12.2 beta | 0.12.1 beta --------------------------------------------------- 192 Stereo | 8.67/8.85/8.58 | 9.10/9.13/9.14 160 Stereo | 8.85/8.86/8.70 | 9.32/9.16/9.26 128 Joint Stereo | 9.06/8.91/8.88 | 9.14/9.20/9.25 You can get ATM at http://web.tiscalinet.it/TbEnrHP/higheng.html and try this test yourself. Thanks for the release Rob!

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