- mad-user - MARS Mailing Lists

24bit + MX300 + Wrapper
by Geoff 05 Sep '01

05 Sep '01

I downloaded the aformentioned wrapper, and lo and behold, it works. I'm using the latest build of drivers for Vortex 2s which you can get from www.vortexofsound.com, and the wrapper program only works with waveout module of Winamp.

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RE: [mad-user] 24bit Output in WinXP on a MX300
by Jens Rex Bræmhøj 04 Sep '01

04 Sep '01

Hello. Below is an e-mail i recieved from Terratec Support. I was complaining that the 24 bit output on the EWX 24/96 was not working under Windows 2000 with the WDM drivers. ------------------Begin Quote------------------------- From: Stacey Curtis [ttsupport(a)terratec.de] ------------------------------------------------------ Hello, The WDM driver architecture does not support 24bit resolution. This is a limitation of the driver architecture itself. For ACPI you must use WDM - but for 24bit you need to use VxD drivers and disable ACPI under W2K. There is no other solution until the WDM architecture is expanded. Mfg/best regards, Stacey Curtis Email: ttsupport(a)terratec.de *****[snip snip snip]***** error: With the newest drivers I am unable to utilize the 24 bit output capability. 16 bit works fine however. With the older drivers (Version 4-something i think) 24 bits worked fine. But i want to use the version 5 driver because it allows me to use ACPI. The problem have persisted through a complete reinstall of Windows 2000. comment: THANKS for fixing the bug where the Digital Out Source was not remembered in the control panel. I was becoming *SO* annoyed by it. -------------------End Quote-------------------------- ------------------------------------------------------ -- Your momma is so dumb, she set her bufferlength to a minimal octal instead of purging the cache. > -----Original Message----- > From: Hitesh Patel [mailto:hpatel99@email.com] > Sent: Tuesday, September 04, 2001 7:18 PM > To: Geoff > Cc: mad-user(a)lists.mars.org > Subject: Re: [mad-user] 24bit Output in WinXP on a MX300 > > > On Mon, 3 Sep 2001 21:57:35 +1000, you wrote: > > >Has anybody gotten this combo to work? > > > >I ask because 24bit output works for MX300 (and I assume all > Aureal cards), > >under WinME. > > > >Any help would be great > >Thanks, > > > >Geoff > > I know it doesn't work with XP with the AudioPhile 2496. > > Which leads me to a question. What exactly is it that is preventing > me from having 24bit output? Is it the Winamp output plugin, or is it > the OS (XP now and W2K before that), or is it the M-Audio drivers? > > I have tried just about everything, including: > > Disabling ACPI so that Win2K would let me assign IRQs manually, > > which led to > > Reformatting and installing WIN2K > > As a result the 2496 has its own IRQ of 9. No more shared IRQs. > > I tried the Win NT drivers for the 2496. I set the max blocksize to > 16k and the min blocksize to 4k. This results in 24 bit output working > but with a horrible static/buzz that drowns out the actual music, no > matter what settings I try. > > What else should I do? I would love ANY suggestions. > > Regards, > > Hitesh Patel. > > >

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RE: [mad-user] Audio quality vs Nullsoft decoder
by Yura Kloubakov 04 Sep '01

04 Sep '01

I'm using Windows 2000 and Winamp 2.76 and I've noticed a "click" every time Winamp starts playing a new track when using mad plug-in. There is no such problem with Winamp's default decoder. I think it's somehow related to the caching algorithm. May be the default decoder reads the beginning of the next track while finishing playing the current one. Anyway even with the default decoder there are problems with playing the track and using Windows Explorer under Windows 2000. Yuri. -----Original Message----- From: mad-user-admin(a)lists.mars.org [mailto:mad-user-admin@lists.mars.org]On Behalf Of Geoff Sent: September 4, 2001 03:13 To: mad-user(a)lists.mars.org Subject: Re: [mad-user] Audio quality vs Nullsoft decoder It's fantastic. You don't even have to think to hear the diffrence - particularly with mp3s encoded at a lower bit rate - they sound much much better. Also if you can your Vortex 2 to accept 24bit it's even better. Geoff ----- Original Message ----- From: "George Pantazopoulos" <themaxx(a)mediaone.net> To: <mad-user(a)lists.mars.org> Sent: Tuesday, September 04, 2001 12:56 AM Subject: [mad-user] Audio quality vs Nullsoft decoder > > Have you guys done many comparisons between MAD and the defualt mp3 > decoder? What were your results and how much of a diffrence is there? What > kind of music did you use, etc? Im in the process of testing it myself as > well. I cant wait till my Hoontech DSP24 VE comes in to try this baby out :) > > > Thanks, > George Pantazopoulos > >

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Audio quality vs Nullsoft decoder
by George Pantazopoulos 04 Sep '01

04 Sep '01

Have you guys done many comparisons between MAD and the defualt mp3 decoder? What were your results and how much of a diffrence is there? What kind of music did you use, etc? Im in the process of testing it myself as well. I cant wait till my Hoontech DSP24 VE comes in to try this baby out :) Thanks, George Pantazopoulos

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24bit output soundcards - Vortex 2s
by Geoff 03 Sep '01

03 Sep '01

Well I found my answer here: http://www.tbrowne.demon.co.uk/comp/projects/dacs.htm In short, the MX300 doesn't acutally have 24 bit output but it does a very high quality downsample to 48kHz via an onboard SRC. Apparently the SBLives are supposed to have this although the quality isn't that good. Anyway, the reason this popped up is because I noticed a drop in playback quality from WinME (24bit, downsampled to 16bit on the card), to WinXP (16bit). Geoff

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MAD in CoolPlayer
by Rob Leslie 31 Aug '01

31 Aug '01

For CoolPlayer enthusiasts: I've just committed a working version of MAD into the newly GPL'd CoolPlayer project at SourceForge: http://sourceforge.net/projects/coolplayer/ There's not an official executable release yet, but I expect you'll see one before long. If you're developer-inclined, you can get the latest code from CVS and try it out. Cheers, -- Rob Leslie rob(a)mars.org

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Dithering Algorithm
by Marcus Kwok 22 Aug '01

22 Aug '01

On the r3mix forum (http://66.96.216.160/cgi-bin/YaBB.pl?board=general&action=display&num=99616…) they mention something about the error variable is uninitialized between songs, resulting in different decodes of the same song. They also mention that perhaps it might be better (speed or quality, I'm not sure) to use trunc() or round() instead of floor() in the algorithm. -- Marcus B. Kwok | 338178 Georgia Tech Station | AOL Instant Messenger: SLuGHEAd78 Atlanta, GA 30332 | ICQ: 20039767 http://gtf.org/ricecake/ | ricecake(a)gtf.org

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madplay in cirrus logic EP7312
by 陆如枫 22 Aug '01

22 Aug '01

Hi: all Did any one have ever tried to run madplay in Cirrus-logic EP7312 with Bluecat linux? The problem I meet is: when using the "audio_empeg" module and "mono" selection ,I can get normal decoded sound output, when selecting stereo,the output sound is noised. And the wave-format is OK. Can anyone tell me what is wrong and how can I get nomal stereo sound.

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Now that CoolPlayer is Open Source, how about making it a showcase for MAD?
by Roger Ramsey 19 Jul '01

19 Jul '01

The author of CoolPlayer has now made it Open Source. Would Rob or some other worthy consider removing the XAudio decoding engine from it and making it the showcase for MAD? Roj ICQ: 12258585 "Faith manages." jms

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M-Audio Audiophile 24/96
by Scott 15 Jul '01

15 Jul '01

I am new to the list so please excuse me if this topic has already been discussed. I have searched the internet trying to find out why my M-Audio Audiophile 24/96 will not play back in 24bit mode. I am running Windows XP RC1 Beta and I am using M-Audio's latest Windows 2000 WDM drivers. I have verified with M-Audio that this driver version does support 24bit output. They told me: "The latest drivers are 4.12.1.20 are 24 bit but limited to your application rather then the drivers". When I try to play an mp3 file via winamp with the MAD plugin enabled with 24bit selected, I get an error that the format is not supported. I currently have this installed in my old Intel BX chipset based motherboard with a Intel Celeron 400mhz processor with 384 megs of PC100 SDRAM.

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(no subject)
by BN Dattaguru 14 Jun '01

14 Jun '01

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(no subject)
by BN Dattaguru 14 Jun '01

14 Jun '01

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RE: [mad-user] mad winamp plugin and terratec ewx 24/96
by Jens Rex Bræmhøj 01 Jun '01

01 Jun '01

I have the very same soundcard, and it's playing Massive Attack in 24 bits of sweet High Fidelity audio as i type these words. Have you made sure to set Max Blocksize to 16 and Min Blocksize to 4 in Winamps Wave_Out configuration dialog? (as specified on the Mad homepage) -- http://www.webstandards.org/upgrade/pr.html http://www.microsoft.com/windows/ie/download/ie55sp1.htm -----Original Message----- From: Carlos Gonzalez [mailto:carlos.gp@corevia.com] Sent: Friday, June 01, 2001 7:14 PM To: mad-user(a)lists.mars.org Subject: [mad-user] mad winamp plugin and terratec ewx 24/96 I've just installed my new terratec, but I'm having problems. How can I make mad to work at 24 bit under winamp? I can't control volume using windows volume slider. Is that normal? -- Carlos Gonzalez

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mad winamp plugin and terratec ewx 24/96
by Carlos Gonzalez 01 Jun '01

01 Jun '01

I've just installed my new terratec, but I'm having problems. How can I make mad to work at 24 bit under winamp? I can't control volume using windows volume slider. Is that normal? -- Carlos Gonzalez

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console win32 madplay.exe
by Dmitry 25 May '01

25 May '01

Hello mad-user-request, Friday, May 25, 2001, 4:26:23 PM, you wrote: murlmo> 1. Can anyone send me a Win98 Compile? :)? (Kurt Penca) http://dkutsanov.chat.ru/mad-0.13.0b.zip http://jthz.com/~lame/mad-0.13.0b.zip sorry, for i didn't do this early, forgot about this 8((( btw 2Rob: is it ok, or i must change something in package ;) Best regards, Dmitry mail to: kuts(a)atrus.ru http://dkutsanov.chat.ru/

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Listening test of auto-attenuation
by ff123 25 May '01

25 May '01

I believe the feature works as advertised. http://fastforward.iwarp.com/clipping.html ff123

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WAVEFPR<AT+EXTENSIBLE?
by greg.tiller＠ns.sympatico.ca 22 May '01

22 May '01

I assume the problem with win2k not supporting 24bit audio for Midiman cards lies here.. my question is does this lie with the wave_out not using this format, or is it on mad's end? --- * apps not supporting WAVEFORMAT_EXTENSIBLE (this is a new wave format spec released by Microsoft 2 years ago) will be limited to 16 bit stereo output. However, in the last few month more and more apps (like SoundForge 5, Acid 3...) do support this, so they're not limited in output bitdepth.

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Low level programming interface question
by Lionel Ulmer 08 May '01

08 May '01

Hi all, Let's hope there are still people reading this mailing list :-) I am in the process of writing a small AVI player using OpenDivX and MAD (I need MAD because the ultimate goal is for it to run on my Linux-powered iPAQ and I need fixed point math :-) ). If I put aside the licensing issues ( :-) ), my biggest problem is to really understand how the low-level MAD interface is working. After looking at the WinAMP plug-in source, I somewhat understood how it works (except for the timers, where I am completely lost). My understanding for now is : - read some data from the stream - call mad_stream_buffer - call mad_frame_decode - call mad_synth_frame - send the data to the sound device - copy the data remaining in the buffer (ie the 'next_frame' pointer) to the start of my input buffer For a stand-alone player, this works wonderfully. The problem comes when I try to incorporate this in my AVI player. Indeed, when I added some printfs to my small player, I saw that MAD always : - output the same amount of data (0.006 seconds of sound) - always 'progressed' of the same amount of data in my stream On the other hand, in my AVI stream, for each video frame, the amount of MP3 data is not AT ALL constant.... Moreover, it is sometimes smaller than the amount of data read by MAD => I get (as soon as the first video frame) a decode error from MAD (I did not check exactly what error it was). I could do 'buffering' (ie concatenate some sound AVI chunks) but then I would completely lose my synchro (and I do not want to rely on timers or to retrieve the synchro, it would start to get bad with long films). So anyone could light my lantern as to how MAD can be used to do what I need ? Thanks, Lionel PS: I hope this is the right mailing list, I did not know if I should send this to the devel list or the user list :-) -- Lionel Ulmer - http://www.bbrox.org/

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Winamp Plugin File Info Box suggestion
by Marcus Kwok 21 Apr '01

21 Apr '01

I have some large MP3s (greater than 100 MB/100 Minutes). I have noticed that every time I view the File Info box, the length of the song takes several minutes to calculate. I think it would be useful if the plugin cached this info, so that any subsequent visits to the box would have this information already available. -- Marcus B. Kwok | 338178 Georgia Tech Station | http://angband.org/~ricecake/ Atlanta, GA 30332 | AOL Instant Messenger: SLuGHEAd78 ricecake(a)angband.org | ICQ: 20039767

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Hoontech dsp 24
by Henri Chapelle 20 Apr '01

20 Apr '01

Hi, does the Hoontech dsp 24 value card work in 24 bit with the mad plugin in W2K ? Thanks, Henri Chapelle gandalf_hc(a)yahoo.com __________________________________________________ Do You Yahoo!? Yahoo! Auctions - buy the things you want at great prices http://auctions.yahoo.com/

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Audiophile 2496 and Mad Plugin
by Jeff Wawryk 18 Apr '01

18 Apr '01

When playing Mp3's in 24bit mode, Winamp will just stop playin and just the progress bar in winamp will just sit there, if I drag it forward to continue a just head a bunch a digital type sounds, kinda like a buzzing or constant glitch sound. This will happen just from Surfing the Web. how can I stop this from happening, I have to restart about half the songs from the beggining cause of this. Maybe a setting in the Wave ouput plugin. Thanks Jeff ICQ 32714714

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MAD 0.13.0b available
by Rob Leslie 11 Apr '01

11 Apr '01

MAD version 0.13.0b is now available. Highlights of this release include: - complete file decode, including last frame - support for reading Xing VBR headers in the plug-in for Winamp - fixed visuals display in the plug-in for Winamp for 24- and 32-bit output - support for Layer III ancillary data extraction - fixed MPEG-2 Layer III intensity stereo decoding - new `--downsample' option for madplay (reduced decoding overhead) - fixed PIC-related crashing bug under ARM - libmad should now be thread-safe More details can be found in the `CHANGES' file in the release. The source code is here: ftp://ftp.mars.org/pub/mpeg/ The plug-in for Winamp is here: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/ Cheers, -- Rob Leslie rob(a)mars.org

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MAD plug-in for Winamp 0.13.0b-pre1 and a wish
by D D 09 Apr '01

09 Apr '01

I must thank Dean-Ryan Stone for pointing me to this SqrSoft plugin... I had never encountered it before and it looks quite elaborated and versatile, definitely a pro product. Now let's see how it performs in conjunction with MAD :)

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Inserting the gaps - solution
by Vlad Indiukov 08 Apr '01

08 Apr '01

Hello. It was taken from instructon to Dudsoft Gapless Output Plugin V1.4 (c)1999 By Nathan Ford: ----- The delay is directly proportional to the buffer size of the Wave-out Device. For instance, if the buffer is set at 2000ms, then it will take 2000ms (2 seconds) for the change to occur. ----- So maybe you should experiment with your standard WaveOut plugin and set the biggest buffer size to increase the length of pauses between tracks? Try 10000 ms :) It's terrible :) -- Best regards, Vlad mailto:voffka2@chat.ru

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MAD plug-in for Winamp 0.13.0b-pre1 and a wish
by D D 08 Apr '01

08 Apr '01

This was a most welcome development, Rob, thanks a lot! Since I encode everything using Lame VBR, this fix took care of a nuisance. MAD is as musical and stable as it ever was - I saw no bugs in this pre1 build. I also noticed the DLL is some 40 KB shorter too - I wonder what more improvements were made to the code :) I take the opportunity to focus on one of my most disturbing problems regarding mp3 playing, which is gapless playing across mp3 tracks. I have tried all the solutions available for Winamp and I was not satisfied at all with any of these. Has anybody come across an optimal solution specifically for use with MAD ? I suspect that existing gapless plugins are written for the default winamp decoder so they'd in principle fail to operate properly with MAD. If this is the case can we hope in the future for some solution for gapless playing through MAD? Thank you for all your efforts and have a happy and musical Easter :)

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MAD plug-in for Winamp 0.13.0b-pre1
by Rob Leslie 03 Apr '01

03 Apr '01

Here is a pre-release version of the MAD plug-in for Winamp: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/in_mad-0.13.0b-pre1.zip This version should properly read Xing VBR headers to quickly and accurately calculate the length of VBR files. It should also properly play the last complete audio frame of every file. Happy April Fool's Day, -- Rob Leslie rob(a)mars.org

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File length detection
by Mark Powell 31 Mar '01

31 Mar '01

Hi, Couldn't see this question asked in the archive so... Why is MAD so slow at detecting the correct length and average bitrate of a track? Surely just reading the Xing VBR tag gets all this info in just the first few hundred bytes of an mp3? It strikes me as quite odd that it takes so long to detect this. WinAmps in-mp3.dll seems to have none of these problems. Cheers. Mark Powell - UNIX System Administrator - The University of Salford Academic Information Services, Clifford Whitworth Building, Salford University, Manchester, M5 4WT, UK. Tel: +44 161 295 5936 Fax: +44 161 295 5888 www.pgp.com for PGP key

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(no subject)
by chf＠sunplus.com.tw 24 Mar '01

24 Mar '01

Dear Rob, ALL : I download MAD plusg-in 0.12.5b.exe from www.mars.org And Decoding Result of in_mad.dll is different from in_mp3.dll Using in_mad.dll: vol of right channel is larger than left. but if using in_mp3.dll, left and right have same vol. If Possible, please tell me which one is correct ? Best Regards, Judy 23.March, 2001 (See attached file: 8.mp3) (See attached file: 8.mp3)

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problem about MAD plug-in for winamp
by chf＠sunplus.com.tw 23 Mar '01

23 Mar '01

Dear Rob, ALL : I download MAD plusg-in 0.12.5b.exe from www.mars.org And Decoding Result of in_mad.dll is different from in_mp3.dll Using in_mad.dll: vol of right channel is larger than left. but if using in_mp3.dll, left and right have same vol. If Possible, please tell me which one is correct ? Best Regards, Judy 23.March, 2001 (See attached file: 8.mp3)

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problem about MAD plug-in for winamp
by chf＠sunplus.com.tw 23 Mar '01

23 Mar '01

Dear Rob, and All: I download MAD plusg-in 0.12.5b.exe from www.mars.org And Decoding Result of in_mad.dll is different from in_mp3.dll Using in_mad.dll: vol of right channel is larger than left. but if using in_mp3.dll, left and right have same vol. If Possible, please tell me which one is correct ? Best Regards, Judy 20.March, 2001

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Help with error msg!
by Jeff Wawryk 15 Mar '01

15 Mar '01

Just installed Winamp 2.72 and the newest mad plugin, Whenever I change to 24 or 32 bit I get a "Cannot Set Ouput " Im using an Audiophile 2496 on Win98se. Can anybody help? Ive tried changing buffer size and a few other setting with no luck. Thanks Jeff ICQ# 32714714

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playlist with mad?
by Jonah Michaud 15 Mar '01

15 Mar '01

Hello, I was wondering what people are using to do playlists with mad, maybe xhippo? I'm using Linux/KDE. And I love MAD, the sound is noticeably "cleaner". Not sure how else to describe it. Thanks! __________________________________________________ Do You Yahoo!? Get email at your own domain with Yahoo! Mail. http://personal.mail.yahoo.com/

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Re: [mad-user] playlist with mad?
by Jonah Michaud 12 Mar '01

12 Mar '01

> Jonah Michaud wrote: > > Hello, I was wondering what people are using to do playlists > > with mad, maybe xhippo? I'm using Linux/KDE. > > I use the filesystem as a playlist, with symlinks for various > cross-categories. I then usually use madplay's shuffle mode to mix things up, > or sometimes I just play through a single album using the filenames sorted > into track order. > > I like GUIs, but as you might guess I also often prefer alternatives. :-) > I hope this is the right way to reply- I'm just copying the subject line by hand. I didn't notice there was a shuffle mode, I'll try that for myself. Then maybe compile xhippo for my parents since they're still kind of new to Linux. > There has been one proposal to add some kind of option similar to mpg123's -@ > playlist option to read a list of filenames from another file. Does anyone > else also like this idea? > That sounds useful. Now that I know I can play a series of files on the command line though :) I'll just use that. It doesn't matter to me personally whether the list is on the command line or in a file. Thanks again for such a great free player. > Cheers, > -rob __________________________________________________ Do You Yahoo!? Yahoo! Auctions - Buy the things you want at great prices. http://auctions.yahoo.com/

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Slightly off topic - recording limiter
by D D 08 Mar '01

08 Mar '01

I am to ask in this list since over the years I have asked a lot of people in several www boards with no luck. I am looking helplessly for a solution to dynamic level limiting during recording via soundcard. It seems MAD's dynamic level limiting is unique worldwide (Rob, you should definitely patent this). I just cannot understand why nobody, repeat nobody, has ever thought about doing the same thing for recording. Why cannot a recording software (or audio editor plugin) implement a scheme for automatic level decreasing when recording in the same manner as MAD ? Is it technically impossible ? I'd appreciate very much any tips (or other places to ask this question) you may have, since I spend a lot of time to adjust the optimal level during recording. Thank you in advance :)

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Slightly off topic - recording limiter
by D D 08 Mar '01

08 Mar '01

Judging from the post so far and a private response I got, I realise that I was not very clear in my first post. I am not sure whether the input level sliders work on the analog or the digital part of soundcards, i.e. before or after the ADC, but having as a working hypothesis that the mixer works on the analog circuit, what I do is: a) match my preamplifier console output level and the soundcard input (mixer slider) so they give a little above 0db b) lower gradually *with the mouse* the mixer slider until no clipping occurs. In these terms what I desire is a software solution to lowering the slider i.e. a program to do what my hand does when lowering the slider, in the very same sense that MAD does it - when it sees clipping it reduces the "amplification". In the hope that I am more clear now, I wait for any further ideas :)

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MAD 0.12.5b
by Magnus Holmgren 04 Mar '01

04 Mar '01

Hi, It seems that the 0.12.5b version of the MAD WinAmp plugin is a bit unstable. Since installing it, I've had WinAmp (2.72) crash several times on me (generally trying to read memory it isn't allowed to read). I think this happens at the end of a file. Once or twice it has even crashed Windows 2000 on me (as luck would have it, once it happened when I'd just started burning a CD)... Have anyone else had problems with this version? -- Magnus Holmgren

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MAD w/ 4.1 speakers
by RSWAVE2000＠aol.com 25 Feb '01

25 Feb '01

Is MAD made only for 2 speakers and it also for 4.1 speakers. MAD sounds good on my front speakers but not my rear speakers. I only hear sound from my front speakers sitting down but if I go up to my rear speakers it is sound coming out. ???

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Re: Clipping - Some naive (?) questions
by D D 15 Feb '01

15 Feb '01

Dear all, Thank you for your lengthy responses, things seem to be much clearer in my head now. By reading the replies and ff123's article I think realise three points: a) Clipping is an inherent feature of the mp3 decoding-encoding process as related to the necessary time domain (integers) <--> frequency domain (floating point) transformations. b) Clipping during the reproduction of digital audio is also inevitable, due to sampling effects, because of players (domestic or computers) trying to restore the waveform parts contained within the samples. c) If I understood well, these two effects are interrelated. (a) implies that any wav file close to 100% risks of being clipped, and (b) implies that the extent of clipping depends on the type of music program encoded. (b) also for me implies that whenever there is high frequency content present in the wav (e.g. cymbals), the decoding process has few samples to process and moreover the respective analog waveform has acute peaks, so repeated short-term clippings are inevitable. On the other hand, low frequencies (violins, voice) have smooth peaks so when when clipping occurs, we have a long-term clipping i.e. a flattening of the waveform. I am convinced that both of these phenomena, detected by the peak detector of Mad, are quite audible - perhaps the low-frequency cases are a bit easier to detect because of higher ear and brain sensitivity to them. I wonder whether modern recording engineers are fully aware of these effects. The "Remastered" stickers on all those new generation CD's make me think ...

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Conditional setting in MAD plugin
by Helmut Hübers 14 Feb '01

14 Feb '01

How does this conditional stuff for the tag work? I would like MAD to show me the trackname if an ID3 tag is not present, not only this "? - ?". Can someone give me an example? In the MAD plugin configuration there is mentioned a conditional option %?x<IF|ELSE>, but I don't understand how it should be used. Thanks, HelH

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flaws in 12.2 fixed ?
by Samuel Tan 14 Feb '01

14 Feb '01

Hear there's a problem with the dithering in 12.2b from decoder test page. http://privatewww.essex.ac.uk/~djmrob/mp3decoders/decoders_winampmad.html Just wondering, has it been fixed yet ? And what about the clipping of the end of files ? _________________________________________________________________________ Get Your Private, Free E-mail from MSN Hotmail at http://www.hotmail.com.

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end of files
by ff123 14 Feb '01

14 Feb '01

I don't know if it's a general problem or not, but I notice that MAD produces glitches in the output at the end of files. For instance, take a short .wav file and cut off the end with a sound editor so that there's still music playing at the end (not silence). Encode to mp3 and play with MAD. Then try the same thing with the decoder in Winamp 2.7x. I tried this test with MAD versions .12.3, .12.4, and .12.5. Maybe there's something about my PC or soundcard, I don't know. Also, I haven't checked recently, but at the time I was doing a MAD listening test on Usenet, I noticed that the MAD decoder (version .12.2) cut off about 30 milliseconds at the end compared with the Fraunhofer decoder in Winamp 2.666. Maybe the two problems are related. ff123

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Clipping - Some naive (?) questions
by D D 14 Feb '01

14 Feb '01

Greetings to all, I'm a new user to MAD and it's immediately become my standard for mp3 listening and decoding. The first thing that shook me when using Mad, was something that I never had encountered before, the mp3 clipping indicator. Low-bit distortion is one thing, but high-bit problems is surely a worse one, if not the worst! And when I witnessed Mad reports clipping even with files produced by the best encoders like Lame, this rang a red alert for me: are we constantly exposed to clipping distortion when using winamp's decoder or other decoding apps? Research in the web showed that there are people who claim that clipping is a well-known artifact of lossy mp3 compression. For me this is still a mystery, since I had the misconception (or is it true ?) that a well-established or even commercial encoder cannot possibly produce an mp3 that exceeds the 100% level. I am not familiar with the binary pcm structure of wav files, but I know it uses 16-bit numbers, so clipping is by the definition either the presence of a 17th bit (impossible) or the miscoding of a music level of 99% as 100% (which means that everything above 99% is clipped cause it's represented as a constant 100%). On the other hand, I understand that an encoded mp3 level of 99% is possible to be interpreted by a decoder as a level of 100%, which would mean that clipping is indeed possible to occur, but *only* when the file is badly decoded. For me two possibilities exist: a) The offending mp3 does not contain clipped samples but Mad misreports them (rather improbable, I do trust Mad's author programming skills) b) The offending mp3 does contain clipped samples which were produced by bad encoding (therefore Mad is correct) c) The offending mp3 does not contain clipped samples, but a decoder (either because of bad programming or because of the nature of mp3 lossy compression) *could* misinterpret them, so a decoder should take corrective action (i.e. attenuation like Mad does). I cannot easily find a way to produce convincing evidence as to which of the above is the case. At first sight one could propose: "just take an mp3 and decode it with a standard decoder and see if it contains clipped samples". However this is not good enough for me, for the following reason: If i use 10 standard decoders I'll get 10 wavs which will almost certainly differ. The next step would be to *visually* inspect the wavs for clipping, cause I do not think that any wav editor is capable of detecting and reporting (and possibly take action like Mad's attenuation) levels above 100% when opening a wav or importing an mp3 (am I wrong here?). In conclusion, to rephrase my (naive ?) questions: how is it in principle possible that Mad detects clipping, how does Mad do it and where exaclty is Mad detecting it: a) is clipping stored inside the mp3 file content? b) is it an artifact of decoding programs ? In addition, can you suggest another reliable means of verifying that an mp3 contain or not clipped content ? Excuse me for the lenghty post, but I think the issue I bring forward is an important one. For one, I would hate the idea of being forced to first normalise all my wav's to a safe e.g. 95% before encoding it, to just prevent the eventual clipping upon decoding.... Moreover if my worries are justified, we should definitely propose "response to mp3 clipping" as a very important criterium for the evaluation of mp3 decoders (and encoders ?) in that famous www page that does "objective" mp3 software comparisons. Best regards, Mits

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MAD 0.12.5b available
by Rob Leslie 13 Feb '01

13 Feb '01

Hot on the heels of 0.12.4b, MAD 0.12.5b is now available with even more performance improvements. This version incorporates the long-awaited Huffman decoding optimization that performs requantization in the same step. Many thanks to Andre McCurdy for his help that led to this release. There are still a few more improvements to be made, but this probably represents the bulk of what can be optimized for significant gains. The source code release: ftp://ftp.mars.org/pub/mpeg/ The MAD plug-in for Winamp: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/ The MAD home page: http://www.mars.org/home/rob/proj/mpeg/ Cheers, -- Rob Leslie rob(a)mars.org

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ID3v2 support?
by Vlad Indiukov 12 Feb '01

12 Feb '01

What about this stuff for winamp plugin?

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resample problem..
by J.Saravanan 12 Feb '01

12 Feb '01

Hi All, I've compiled madplay for my arm target ( Its a CLPS7500 evaluation board). When I try to run the application I get the following error message... output: sample frequency 44100 Hz not availlable;closest 41666 Hz output: cannot resample 44100 Hz to 41666 Hz And the program quits. Can sombody tell me how to fix this problem. Thanks, _Saran_

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MAD 0.12.4b available
by Rob Leslie 12 Feb '01

12 Feb '01

MAD 0.12.4b is now available. This version incorporates a number of performance improvements for all platforms. It also includes (untested) native fixed-point math support for the PowerPC platform, contributed by David Blythe. Since I've made some significant changes to the way the fixed-point math routines are maintained, I'd appreciate feedback from anyone using MAD on a PPC to make sure I haven't broken anything. Finally, this version changes the way mixed short blocks are handled in Layer III with respect to alias reduction, correcting another apparent ambiguity in ISO/IEC 11172-3. Since mixed blocks are rare (many encoders don't support them) this has until now gone unnoticed. Thanks to H.O. for reporting this. The source code release is here: ftp://ftp.mars.org/pub/mpeg/ Winamp users should notice improved performance in the MAD plug-in: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/ The MAD home page is, of course: http://www.mars.org/home/rob/proj/mpeg/ Cheers, -- Rob Leslie rob(a)mars.org

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New mailing list software
by Rob Leslie 01 Feb '01

01 Feb '01

This message serves both as a test and as an announcement of new mailing list software that is managing the mad-dev and mad-user mailing lists. If all went well, the transition should be pretty seamless. You can now manage your subscription from one of these pages: (mad-dev) http://www.mars.org/bin/mailman/listinfo/mad-dev (mad-user) http://www.mars.org/bin/mailman/listinfo/mad-user To make changes, first click the button to have your password mailed to you. As a bonus, both lists now have an online archive of past messages: (mad-dev) http://www.mars.org/mailman/public/mad-dev/ (mad-user) http://www.mars.org/mailman/public/mad-user/ Please let me know if you experience any problems. Cheers, -- Rob Leslie rob(a)mars.org

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Comparing with old decoders
by Vlad Indiukov 30 Jan '01

30 Jan '01

Does anyone compare MAD quality with NAD and Appolo decoders? I mean objective comparing with frequency analysis etc, not sayings like "smooth and deep sound of decoder"

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Why the strange drop-off in discussion?
by David Shin 30 Jan '01

30 Jan '01

At least 4 or 5 a day and then, boom, nothing. Am I still on the right mailing list?

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l3dec 24-bit convert to wav
by ff123 28 Jan '01

28 Jan '01

I have written a short program to convert the 24-bit ASCII output of l3dec to .wav format, which can be read in by Cool Edit Pro. For fun only -- I don't see a useful purpose for it. http://fastforward.iwarp.com/convert.html ff123

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Re: [mad-announce] MAD 0.12.3b available
by Anthony Airon Oetzmann 24 Jan '01

24 Jan '01

On Sat, 20 Jan 2001 23:05:54 -0500, Rob Leslie wrote: >MAD version 0.12.3b is now available. > >Highlights of this release include: > > - MPEG 2.5 support Way to go. > - better mono/stereo/left/right channel selection > - attenuation option Yeah, how'd you implement those levels of sensitivity ? Any limiter code in there ? Take care and keep it up Tony

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Binary of Madplay
by engdev 24 Jan '01

24 Jan '01

Do any of you nice fellows out there have a binary of Madplay that runs on windows (or ms dos) operating systems. I've looked extensively but have only managed to find source code, and, I think it was a Linux binary. I know, doze is really bad, but it has to work with other programs written for doze. I don't have the programming skills to compile my own. Thanks, Owen.

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Playlist track time abnormalities
by David Shin 28 Dec '00

28 Dec '00

This has been reported before, but I just thought I'd bring it up again. When you have Winamp set to Read Titles on Play (Winamp Preferences->Options), MAD will not show the track time for the first file that's played. Not the first track on the playlist, but the first track that you try to play (which doesn't have a track time listed already). Also, VBR track time handling needs to be better (until VBR headers is supported). I realize MAD can't help reading wrong track times when Fast Playing Time Calculation is enabled, but it should show the correct times when the files are actually playing. Every other decoder changes the track time correctly (even after MAD has loaded the wrong track time into the playlist), but MAD will not. For example, if you enable MAD and load a bunch of VBR files, the wrong track times will show in the playlist. Now disable MAD, enable the default decoder, and play each of the files. The correct track times will now show. Why can't MAD do the same thing? It should send the correct track time when the file is actually playing. Even if it takes up to 5 or 6 seconds, it should still send the correct time. Is this somehow related to the previous bug where it won't show any track time at all when you first play a file in Read Titles on Play mode?

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Re: [mad-user] re wishlist
by kaiwei 21 Dec '00

21 Dec '00

(2) the DSP plug-in can be used with other input sources, not just MPEG audio Just for the above reason alone, I think we should go for the 2nd method;-) Imagine, CD audio playback at 24bit!

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New to the list
by Darryl Miyaguchi 20 Dec '00

20 Dec '00

Hi all, I just joined the list, and am interested in finding out if there is an archive of messages I can browse through. I'm intrigued by the idea behind MAD, and have been trying to compile a list of tests which can definitively (in the form of multiple blind listening trials) show that the differences between MAD and plain-vanilla ISO-compliant layer 3 decoders are audible. This is my web page: http://fastforward.iwarp.com/mad.html ff123 (aka Darryl Miyaguchi)

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Re: [mad-user] re wishlist
by Rob Leslie 19 Dec '00

19 Dec '00

"Anthony Airon Oetzmann" <airon(a)gmx.net> wrote: > Sounds complicated to me :). I didn't figure the Winamp specs were that lax. The API is actually somewhat reasonable, it's just that not all plug-ins may support all sample sizes. > There's supposed to be a v3 of Winamp soon. I wonder how well MAD will fit > in there, given that it's supposed to be that much mroe flexible. One > distinct advantage of a single plugin-doing-it-all approach, might be the > portability to XMMS and so on. I'm not shure how compatible a seperate DSP > module will be, when realy you're using the same amount of memory and > processing time. On the contrary, portability is usually increased when you limit the number of dependencies. The DSP stuff can always be folded back in if there is not an equivalent place to put it in a port to another platform. > Is it possible to control a DSP plugin from inside an input plugin ? That's > what I'd gathered from your idea of the "High-resolution DSP support" switch > that's greyed right now. It's possible to control the DSP plug-ins so far as to decide whether to invoke them at all, what sample size to feed them, and of course the samples themselves. > Remember that the DSP will rake in the dithering code, since all else would > destroy the profit made from having 24 or 32 bit output in the first > place. It's not great to dither audio twice either if you're going 16 bit > out and DSP'ing it to 16 bit again after processing it perhaps in 24 or 32 > bit with the DSP plugin. Eh? I'm not suggesting to dither twice. Here's the current picture for 16-bit output: +-----+ +-----+ +----------------+/ | MAD | --32--> dither --16--> | DSP | --16--> | Output Plug-in |- audio +-----+ +-----+ +----------------+\ Here's what I'm proposing with the high-res DSP option: +-----+ +-----+ +----------------+/ | MAD | --32--> | DSP | --32--> dither --16--> | Output Plug-in |- audio +-----+ +-----+ +----------------+\ In all cases "--32-->" means MAD's internal 24-bit + 3-bit headroom sample representation. > I'm not shure a modular design has that many advantages, but to provide two > seperate setting controls. While I think this is indeed probably an advantage, the other advantages are (1) I can eventually call the MAD input plug-in "finished" and concentrate on the DSP stuff independently, (2) the DSP plug-in can be used with other input sources, not just MPEG audio, (3) maybe others would be interested in contributing to the DSP plug-in, being less complex than a full-blown decoder, (4) a bug in one plug-in can be isolated from the other, (5) etc. Cheers, -rob

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re wishlist
by Anthony Airon Oetzmann 19 Dec '00

19 Dec '00

On Mon, 18 Dec 2000 21:26:03 -0500, Rob Leslie wrote: >If I write the DSP plug-in myself, I can support 24-bit samples with 3-bit >headroom without any problem. >Actually many DSP plug-ins can probably handle 24- or 32-bit samples, so this >probably isn't a big issue. >What's needed is a way to send high-resolution samples to the DSP plug-ins and >then dither to 16 bits (or whatever resolution) afterwards. That's what I plan >the "High-resolution DSP support" configuration option to be for. Sounds complicated to me :). I didn't figure the Winamp specs were that lax. There's supposed to be a v3 of Winamp soon. I wonder how well MAD will fit in there, given that it's supposed to be that much mroe flexible. One distinct advantage of a single plugin-doing-it-all approach, might be the portability to XMMS and so on. I'm not shure how compatible a seperate DSP module will be, when realy you're using the same amount of memory and processing time. Is it possible to control a DSP plugin from inside an input plugin ? That's what I'd gathered from your idea of the "High-resolution DSP support" switch that's greyed right now. Remember that the DSP will rake in the dithering code, since all else would destroy the profit made from having 24 or 32 bit output in the first place. It's not great to dither audio twice either if you're going 16 bit out and DSP'ing it to 16 bit again after processing it perhaps in 24 or 32 bit with the DSP plugin. I'm not shure a modular design has that many advantages, but to provide two seperate setting controls. Tony

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Re: [mad-user] wishlist
by Rob Leslie 19 Dec '00

19 Dec '00

"Anthony Airon Oetzmann" <airon(a)gmx.net> wrote: >> I wonder though if it wouldn't be better to implement this in a special DSP >> plug-in rather than further complicate the input plug-in... > > When would Winamp ever support 24 bit output of an input plugin? Trouble is, > you'll degrade the output if the plugin is used on 16 bit output up > front. Use it on 24 bit output however, and then dither that and you'll keep > your high resolution that was the point of MAD in the first place. If I write the DSP plug-in myself, I can support 24-bit samples with 3-bit headroom without any problem. Actually many DSP plug-ins can probably handle 24- or 32-bit samples, so this probably isn't a big issue. What's needed is a way to send high-resolution samples to the DSP plug-ins and then dither to 16 bits (or whatever resolution) afterwards. That's what I plan the "High-resolution DSP support" configuration option to be for. Cheers, -rob

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Re: [mad-user] wishlist
by Rob Leslie 19 Dec '00

19 Dec '00

"Anthony Airon Oetzmann" <airon(a)gmx.net> wrote: >>> I am considering an option to set a permanent attenuation setting when >>> auto attenuation is disabled. >> >> Unless your entire MP3 collection has the same dynamic characteristics, >> I don't see how this would be very useful. > > I'd favor a manualy settable attenuation for each song either way. I think I will eventually implement: - manually settable global (default) attenuation - manually settable and remembered per-file attenuation When auto attenuation is enabled, the global/default attenuation is 0 dB but per-file attenuation will be whatever level the auto attenuation ends at. > Btw Rob, how's that code ? Workable ? (Tony sent me some code that implements a compressor.) Yes, it's enough to understand how to do it; thanks. I wonder though if it wouldn't be better to implement this in a special DSP plug-in rather than further complicate the input plug-in... Cheers, -rob

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Winamp plug-in pre-release 0.12.3b-pre4
by Rob Leslie 18 Dec '00

18 Dec '00

The fourth pre-release: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/in_mad-0.12.3b-pre4.zip Not much has changed for auto attenuation, but this version does have a few extra bonuses: - output channel selection now works (that's for you, Roj) - by request, the title format can now include conditionals What are title conditionals? If you want the title to include the ID3 Title if present but the filename otherwise, you can now enter: %?2<%2|%7> These can nest, so something more complicated might be: %?1<%1%?3< (%?4<%4/>%3)>: >%?0<%0. >%?2<%2|%7>%?6< [%6]> This results in titles like: Filename Title Artist: Filename Artist: Title Artist (Album): Title Artist (Year/Album): Title Artist (Year/Album): 01. Title Artist (Year/Album): 01. Title [Genre] ...all depending on which ID3 tag values are present in the file. The default title is now: %?1<%1 - >%?2<%2|%7> Cheers, -rob

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wishlist
by Gabriel Bouvigne 18 Dec '00

18 Dec '00

You already made my 2 first wishes before I had to ask (priority setting and ability to disable the plugin). My other wishes are (no particular order): Output: Ability to set a permanent attenuation level. This attenuation would be done first, then the auto attenuation would be done after. This would allow to reduce clipping without the level changes caused by auto attenuation Decoding: Ability to enable/disable dithering Decoding: mpeg 2.5 Information: information about the encoder used to produce the file Vbr header: use and do not decode it. Visual information: Ability display M/D/S/MS/IS/IMS in the winamp area used to display the bitrate Plugin info: reseting the info box on file change Decoding quality: an even better decoding quality due to post processing (oversampling) Error correction: ability to correct the kind of errors present in the file I sent you, and use interpolation when a frame can't be decoded. CPU usage: use only 0.00005% of the cpu mpeg 2: decoding of multichannel files (ok, we'll have to made an encoder first) I'll stop here, otherwise it seems that I could give you work for the full next year. Regards, -- Gabriel Bouvigne - France bouvigne(a)mp3-tech.org mobile phone: gsm(a)mp3-tech.org icq: 12138873 MP3' Tech: www.mp3-tech.org personal page: gabriel.mp3-tech.org

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Re: [mad-user] Re: Winamp plug-in pre-release 0.12.3b-pre3
by David Shin 18 Dec '00

18 Dec '00

LAME VBR Roger Ramsey wrote: > > I'm catching up on my reading and by question to you is: > > Why on God's Green Earth would you want absolutely **ANYTHING** to do with > that retroactive abortion known as Xing **ESPECIALLY** in that pure > excrement known as its VBR mode? > > Just asking... > > Roj > > ----- Original Message ----- > From: "David Shin" <naw(a)SoftHome.net> > To: "MAD User Mailing List" <mad-user(a)lists.mars.org> > Sent: Friday, December 15, 2000 6:27 AM > Subject: Re: [mad-user] Re: Winamp plug-in pre-release 0.12.3b-pre3 > > > Rob Leslie wrote: > > > > > > David Shin <naw(a)SoftHome.net> wrote: > > > > I have a question though, how does the gradual attenuation work? > > > > > > I thought I explained it succinctly, but I'll try again. :-) > > > > > > Whenever clipping occurs, the peak clipped value is examined and a new > > > attenuation setting is calculated. Previously (or with the "Most" > sensitivity > > > setting) the new attenuation is one that would have avoided the clipping > > > entirely. This can cause sharp drops in the output signal level that may > not > > > be desirable, especially if it was only a few clipped samples. > > > > > > With a less sensitive setting, the new attenuation is somewhere between > the > > > previous setting and one that would avoid the clipping entirely. This > way, if > > > clipping continues, the attenuation will gradually still be increased to > the > > > point that clipping no longer occurs. > > > > > > Once increased, the attenuation is never decreased unless the "Reset" > button > > > is pressed in the Statistics tab. > > > > Ah, that clarified it. I just wanted to know where that "between" point > > was for settings less than Most. This question sort of went along with > > my "do you need all those gradations?" I think a Most and then a Least > > and Moderate setting is all you'd really need. But I think 8 gradations > > is overkill, and doesn't offer any real benefit. Yet I think a > > coninuously changing attenuation would be even better. It doesn't even > > need to look ahead, like the limiter that Anthony is talking about. > > > > > > Is all this pre-releasing building up to a Christmas day present for > all > > > > the devoted good little MAD users out there? If so, reading Xing VBR > > > > headers is on top of my Santa list! > > > > > > Santa is making a list and checking it twice. ;-) > > > > I'll be sure to leave out my virtual milk and cookies! > > > > ---------- > > | | <------Oreo > > ----------

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Re: [mad-user] clipping
by David Shin 16 Dec '00

16 Dec '00

"Roberts, Wes" wrote: > > Wow, I always wondered why I saw the winamp graphics equalizer show the > dancing waveform as level and at the top of the range of loudness. > > But these songs always sounded the best. > > Before I read your email I even wanted to know how to make my own songs like > that. > > Should I still do it, (not to point of clipping). It's hard to do it right in software. You can use tools like Cool Edit Pro, which sounds the best, but that's tedious. Most people just use what's available in their ripping software to merely normalize tracks, which isn't really the same as compression. A lot of these kind of encodes sound very boomy and distorted. I think the normalize function in AudioGrabber is flawed in some way as a matter of fact. Read the specifics of that somewhere, but the main problem is that it's using a crude and fast algorithm to merely get the job done and nothing more. But I take Anthony's view on this matter: "To me material that has been dynamically compressed too much sounds boring. The ear gets used to it, same as it gets used to too much treble boost and you yawn after a few minutes." I like to minimize the flaws in badly mastered discs, but for discs that are made well--meaning no dynamic compression or clipping--I like to keep it untouched (except for encoding into a 256kbps or VBR MP3 file of course).

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Re: [mad-user] re: .12.3b pre3
by Rob Leslie 15 Dec '00

15 Dec '00

"Anthony Airon Oetzmann" <airon(a)gmx.net> wrote: > David, you wrote: >> I have a question though, how does the gradual attenuation work? How >> about something that clips yet isn't continuous, like a cymbal crash? >> And are all those graduations for the setting necessary? I think >> instead of this, a self-adjusting attenuation would be even better. It >> should work at an aggressive level however, and step back or >> un-attenuate as soon as the clipping stops. What I'm thinking is >> something like the concept of variable bitrate MP3 encoding. When the >> decoder sees the first instance of a clipped sample, it automatically >> attenuates the full level, but after so many samples of no clipping, it >> steps back to no attenuation. And so on. What I fear is that a large >> spike could occur in the beginning or middle of a file, setting a high >> attenuation level, and then muffling the rest of the song which might >> not clip at all. > > What you're describing is a Limiter. It would indeed be neat to have this > function. > > Of course this should be a lookahead limiter. The Ultramaximizer L1 by Waves > looks ahead by up to 68 samples on a Protools TDM system(that's its > processing latency anyway). I don't think having a second latency i our case > would be all that bad. Problem is, who can write a Limiter like that ? > Perhaps one the authors of a free VST plugin would include his Limiter code > in here. For short peaks this would be the right thing. Are there any papers or web sites that discuss the pros/cons of limiters? I have always had the impression limiters are undesirable because of the continuous fluctuation of the output signal level. The way attenuation is currently implemented in the MAD plug-in, doing any kind of look-ahead is unfortunately promblematic. This could be changed, but the primary benefit of the current method is that attenuation changes are less audible than they would be otherwise. -rob

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Re: [mad-user] clipping
by Gabriel Bouvigne 15 Dec '00

15 Dec '00

There is something I don't understand. Clipping is sound above the maximum > >allowed sound. As the sound of a CD is stored on a CD, and the maximum > >amplitude sound that can be stored on a CD is obviously the maximum allowed > >intensity on a CD, how can there be clipping on a CD, ie the sound of the CD > >exceding the sound that can be stored on this CD. > >That seems strange to me. > > I'll repeat what I sent to Peter. > > You're thinking lossless compression here, 'cause you won't get a signal > different from the original with lossless compression such as Monkeyaudio. No, I'm speaking of the original CD, without any compression. Some users are telling that there is some clipping on the CD itself. I personnaly think that it's impossible, but wanted to know if I was missing something about it. Regards, -- Gabriel Bouvigne - France bouvigne(a)mp3-tech.org mobile phone: gsm(a)mp3-tech.org icq: 12138873 MP3' Tech: www.mp3-tech.org personal page: gabriel.mp3-tech.org

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Winamp plug-in pre-release 0.12.3b-pre3
by Rob Leslie 15 Dec '00

15 Dec '00

The third installment in the pre-release series... http://www.mars.org/home/rob/proj/mpeg/mad-plugin/in_mad-0.12.3b-pre3.zip This version adds configurability to the automatic attenuation, and also fixes the problem with the joint stereo pie graph. I decided it might be better to control the sensitivity of the auto attenuation rather than use a simple hard limit. The default is somewhat less aggressive than it was before (which would correspond to the "Most" setting.) The idea is to control how gradually the attenuation is increased. If only a few isolated samples clip, a less sensitive setting will attenuate less than a more sensitive setting would. Continuous clipping will continuously increase the attenuation until the signal no longer clips; the sensitivity controls how fast this occurs. Let me know what you think. Cheers, -rob

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Re: [mad-user] decoding to wav
by Rob Leslie 15 Dec '00

15 Dec '00

"Anthony Airon Oetzmann" <airon(a)gmx.net> wrote: >> Once a file stops playing the info dialog no longer updates itself. This is >> so the information stays relevant to the file for which the info was >> opened. >> >> If you close and then re-open the info after starting playback again, >> you'll see things have been reset. > > Shure, but I don't close it :). I write the WAV file twice. Once for getting > attenuation and the second for writing with the gathered attenuation setting. What I meant is, everything is reset even if you don't close the window. It's just that the window stops updating itself once decoding stops, so you don't see that things have actually reset until you open the window again. -rob

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Re: [mad-user] decoding to wav
by Rob Leslie 15 Dec '00

15 Dec '00

"Anthony Airon Oetzmann" <airon(a)gmx.net> wrote: >> Unfortunately, the setting is not remembered even when the file is played >> again immediately. Sorry -- this feature is yet to come. > > Ahaa! But it is if you keep the info window open AND have another way to > activate playback. > > If you have the Winamp main window and the Playlist window open, you can > activate the INFO on one and still START/STOP/FF/REWIND the playback in the > other window. > > This is easy to prove by playing a file that clips. Keep the info window > open and click on your other WInamp window(main or Playlist) and hit 'X' to > play. As you will see the attenuation section will not add any more samples > to the "Clipped Samples". The attenuation stays and you're cool to go :). Actually, no. :-( Once a file stops playing the info dialog no longer updates itself. This is so the information stays relevant to the file for which the info was opened. If you close and then re-open the info after starting playback again, you'll see things have been reset. Sorry, -rob

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Re: [mad-user] decoding to wav
by Rob Leslie 14 Dec '00

14 Dec '00

"Anthony Airon Oetzmann" <airon(a)gmx.net> wrote: > IN any case, for all of you who'd like to write mp3 to a WAV file, have the > file all by itself in the playlist of Winamp, decode it once to get the > perfect noclipping signal, then decode it again with that attenuation > setting. Only having one file in the playlist will ensure that that > attenuation setting is kept, since MAD cannot write that attenuation setting > to a file or registry yet(?). Unfortunately, the setting is not remembered even when the file is played again immediately. Sorry -- this feature is yet to come. -rob

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re: .12.3b pre3
by Anthony Airon Oetzmann 14 Dec '00

14 Dec '00

David, you wrote: >I have a question though, how does the gradual attenuation work? How >about something that clips yet isn't continuous, like a cymbal crash? >And are all those graduations for the setting necessary? I think >instead of this, a self-adjusting attenuation would be even better. It >should work at an aggressive level however, and step back or >un-attenuate as soon as the clipping stops. What I'm thinking is >something like the concept of variable bitrate MP3 encoding. When the >decoder sees the first instance of a clipped sample, it automatically >attenuates the full level, but after so many samples of no clipping, it >steps back to no attenuation. And so on. What I fear is that a large >spike could occur in the beginning or middle of a file, setting a high >attenuation level, and then muffling the rest of the song which might >not clip at all. What you're describing is a Limiter. It would indeed be neat to have this function. Of course this should be a lookahead limiter. The Ultramaximizer L1 by Waves looks ahead by up to 68 samples on a Protools TDM system(that's its processing latency anyway). I don't think having a second latency i our case would be all that bad. Problem is, who can write a Limiter like that ? Perhaps one the authors of a free VST plugin would include his Limiter code in here. For short peaks this would be the right thing. Tony

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decoding to wav
by Anthony Airon Oetzmann 14 Dec '00

14 Dec '00

IN any case, for all of you who'd like to write mp3 to a WAV file, have the file all by itself in the playlist of Winamp, decode it once to get the perfect noclipping signal, then decode it again with that attenuation setting. Only having one file in the playlist will ensure that that attenuation setting is kept, since MAD cannot write that attenuation setting to a file or registry yet(?). That's the way I do it with modules too. One decoding cycle for finding the attenuation setting, another to use that perfect setting. Tony

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Re: [mad-user] Winamp plug-in pre-release 0.12.3b-pre2
by Rob Leslie 13 Dec '00

13 Dec '00

"Gabriel Bouvigne" <bouvigne(a)mp3-tech.org> wrote: > Nice looking stats. > There is a little problem with the file I sent you. Play it and look a the > stats: suddendly there is a 100% use of IS. Actually, I discovered a bug: the joint stereo graph is completely wrong. It'll be fixed in the next release. Cheers, -rob

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Re: [mad-user] Auto Attenuation and decoding.
by Juan Covas 13 Dec '00

13 Dec '00

Greetings, Doh! :) heh sorry for the stupid question :| I meant to write about using the diskwriter plug-in to convert to a wave file, but I assume now that it does take place too. Heh, thanks. Best Regards, Juan _____________________________________________________________________________________ Get more from the Web. FREE MSN Explorer download : http://explorer.msn.com

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Auto Attenuation and decoding.
by Juan Covas 13 Dec '00

13 Dec '00

Greetings, Simple question... auto attenuation also takes action while a file is being decoded, correct? Best Regards, Juan _____________________________________________________________________________________ Get more from the Web. FREE MSN Explorer download : http://explorer.msn.com

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Winamp plug-in pre-release 0.12.3b-pre2
by Rob Leslie 13 Dec '00

13 Dec '00

It's not quite finished, but I thought some people might like another pre-release of the plug-in for Winamp: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/in_mad-0.12.3b-pre2.zip You can at least now turn on (but not configure) the auto attenuation, and I also added some joint stereo statistics. You'll notice some other config options pending too. This version also detects a special (rare) kind of bitstream corruption and discards the corrupt frames instead of trying to play them. Cheers, -rob

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MAD DLL
by FunkyMan 11 Dec '00

11 Dec '00

I wanted to use the MAD decoder in my MP3 Player, but as a matter of facts, i wasn't able to compile it under windows :-( has anyone build a DLL that can be used by a windows program (a kind of LIBMAD.DLL), or anyone made a VC++ project to compile MAD under windows ? thanx in advance. FunkyMan PS: my MP3 Player is mean't to be built with VC++ 6... so i someone could help me compiling the LIBMAD library ... ++ TBX Player by the TeufCorp (http://www.teufcorp.fr.st)

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Sv: Sv: [mad-user] Attenuation, clipping
by Peter Olufsen 11 Dec '00

11 Dec '00

>> 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 ? > >The expected mode of use would be that an appropriate setting would be learned >the first time a song is played, and recalled when the song is played again. > Ok, but where would you store the info ? And would it be fast enough with large song-collections ? If you do this it would perhaps be good to put the plugin in a "learn"-mode so you could get the info from all your songs fast and write it in a file for later use. >> 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. > >This is a good point, but also a tough decision... how much clipping is >acceptable? > >Are there any audio technicians on the list (Anthony? :) who could go into >details with me about sound energy levels, compression, normalization, etc.? I found another file with a few (160 ?) samples with 8db clipping so i think that some kind of setup would be nice! Thanks, Peter

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MAD and insane bitrates
by whee＠e247.com 08 Dec '00

08 Dec '00

Apparently MAD handles 550kbps mp3s just fine, and winamp's default decoder does not. Just one more reason to use MAD .D (the CVS LAME I got allows bitrates up to 550kbps) -whee Get your free e24/7 email at http://www.e247.com/eyelandreg.jsp Send & receive large files with 100MB of storage

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Attenuating pre-release
by Rob Leslie 07 Dec '00

07 Dec '00

Here's a mailing list exclusive: the following URL leads to a pre-release version of the MAD plug-in. http://www.mars.org/home/rob/proj/mpeg/mad-plugin/in_mad-0.12.3b-pre1.zip This version includes a special "Statistics" tab in the file info dialog for the currently playing song. Among other things, you'll find there an indication of how often and how much samples are being clipped. Furthermore, you can choose to attenuate the signal on-the-fly based on the amount of clipping. Attenuation is performed internally on high-precision pre-dithered samples, so it should not affect sound quality except to improve it by eliminating clipped samples. Further ideas I have in this direction include: - "Automatic clipping attenuation" configuration option - remembering the attenuation setting on a per-file basis Currently the attenuation setting is automatically reset at the beginning of each song. I'd like feedback on where this is going. Is this useful? If an automatic attenuation option is made available, would the manual buttons still be useful? Would there be any benefit in an even more manual configuration like, say, a slider? What about amplification rather than attenuation? Cheers, -rob

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

06 Dec '00

It was nice to have that information (and it's a great plug-in) ! 1) Is it correct that the clipping appears in the decoder even though the file was encoded "correct" without clipped samples ? 2) What is the "Other"-errors in the info-box, it seems that there is 1 og 2 in every file ? 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. 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 ? Thanks, Peter

2 1

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?

2 1

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!

3 2

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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winamp plugin speed
by Gabriel Bouvigne 20 Nov '00

20 Nov '00

Isn't mad supposed to be faster than mpg123? In this case, why is in_mad much slower than in_mpg123? Regards, -- Gabriel Bouvigne - France bouvigne(a)mp3-tech.org mobile phone: gsm(a)mp3-tech.org icq: 12138873 MP3' Tech: www.mp3-tech.org

2 1

[mad-announce] MAD 0.12.2b available
by Rob Leslie 20 Nov '00

20 Nov '00

I have released version 0.12.2b of MAD and the MAD plug-in: ftp://ftp.mars.org/pub/mpeg/ http://www.mars.org/home/rob/proj/mpeg/mad-plugin/ This version includes several performance improvements. Cheers, -- Rob Leslie rob(a)mars.org

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