mad-user May 2001

mad-user@lists.mars.org

5 participants
4 discussions

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/

1 0

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

1 0

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.

2 1

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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mad-user May 2001