Re: [MP3 ENCODER] lame exit problems
At 22:56 09/07/00 -0400, you wrote: Is there any way to force a DOS box close after lame has completed encoding? With the Windows binary, it exits properly but with the DOS one it just says Finished and the box stays open. So if I'm using the DOS version to encode with say audiograbber, I can never get to the next track without manually closing the box. Hello, Call a BAT file instead Lame.exe: 88 Echo off Cls Lame.exe %1 %2 %3 %4 %5 %6 %7 %8 %9 Cls Exit 88 This should work... I hope. -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] ACM (2)
At 23:10 22/06/00 +1200, you wrote: It was Acy Stapp not Mark T that had a look at doing it but apparently the Windows ACM doesn't support VBR and I think it was abandoned. I'm not sure where the problem is because I'm fairly sure that the Fraunhofer ACM does play VBR MP3 WAV files. Hello, The real problem is that ACM codecs only know how deal with CBR file formats, this is not a problem in playback (without seeking) because you simply go forward in the stream. If you perform a seek over a VBR file without Xing header (Or something like that) you can get a aproximate position but not occurate and sometimes something totally wrong (Think in 1000 frames at 256 Kbit and 1000 frames at 32 kbit, if you seek 50% you will be positioned in the 4% of the song :-?¿?¿?) The other main problem is the configuration in the compressor, ACM only have one field for configuration that represent in ASCII format all possible combinations, applying this to lame (3.80) you have: 18 different bitrates 4 panning modes (mono, stereo, joint and dual) 6 freqs. This means 432 combinations (:-?) for CBR mode. If you add VBR this number simply explodes: 2 modes (VBR and CBR) 18 different Max bitrate 18 different Min bitrate 10 VBR quality levels Ummm!!!... where is my calculator... oh! here... well... 2.799.360 combinations. Think in a drop combobox with 2 million entries... and you need the last one!!! :-) and thinking in text mode :-) 2 million using around 20 chars. by line means that you need 50 megabytes of RAM to configure partially your LAME-ACM codec. I think that the only one solution is create an ACM codec that is a subset of Lame options: bitrates like: 256, 192, 160,128,32 Mono and joint stereo VBR Quality High, medium and low (1 MinaxBit=128/256,4 96/192,9=128/32) +/- 22050 and 44100 freq. This give us 60 combinations that are something reasonable. ACM support quality high and low, it could be high lame and low lame qualities or high and normal. Note: Othe solution for the ACM bitrates is a "setup" program for the ACM codec and store the desired bitrates in the windows registry. -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] Normalize audio input
At 09:24 28/05/00 +0200, you wrote: Normalization consists of determining the output leven of the audio data, and then multiplying the audiodata with a certain factor so that the highest value becomes 95 or 99% of the absolute maximum (plus or minus 32768). The trick is twofold: First of all, determining the output level is not that obvious. Many programs just count the highest peak in the data, but one peak can thus influence the rest of the data considerably. Other programs calculate the maximum and the desired level with more sophisticated algorithms (root mean squared, which aproaches the human hearing better than a peak scan) You could consider an adaptive algorithm, which would divide the data in blocks, determine the highest peak in each block, and would then use a gradually sliding factor to multiply the samples with. Hello, Yes, but if you use RMS you must need compression to avoid clips, and compression means a reduction in the dynamic range, and less dynamic range reduces the "quality" but it sounds stronger. Recent CDs are RMS normalized (Usually around -8 db) and they sound really horrible (IMO). I think that MP3 coding for a -8 db RMSed song need more bits and is more difficult to detected masked freqs. Am I wrong ? I think that kind of normalization should be done in the decompression process (it means information destruction) like a radio station compressor does. -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
[MP3 ENCODER] Audio format
Hello, Does anybody know what is the audio format of the EXE (Windows) files that are located in this server ? http://www.globalmusic.com/cybermp4/ They say that it is MP4, but what is MP4 ? AAC ? If this is really off topic replay to my mailbox [EMAIL PROTECTED] -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] FhG Codec Versions
At 02:14 12/04/00 +0600, you wrote: Fraunhofer Internal Codec [Released 1996] I know that at least 4 versions of this codec has been released all of them with different results. Probably they have the same engine but... -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
[MP3 ENCODER] (MP3 ENCODER) Downsampling
At 13:20 03/04/00 -0600, you wrote: After talking to Scott quite a bit about this in #mp3encoder, (and not [...] does a lot of lowpass filtering which removes most of the aliasing caused by bad resampling) Hello, I am looking for free routines to up/downsampling audio for my freeware project. Is this new routine free of copyrights ? Can I use it in my project ? Is "easy" to isolate it from the full LAME project ? Ok, I know that this are a lot of questions, feel free to simply drop this message in your trashcan :-) -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] using lame363 for shuttle transmission
At 07:48 15/02/00 -0700, you wrote: Declaring avg_slots_per_frame "volatile" (line 547 in lame.c) fixes it in some cases. I was using the following test options, with fools.wav as my source Delcaring it volatile (what does that do?) or static changes the As far as I know "volatile" means that the variable/s should not be optimized, it should not be loaded in any procesor register. The usually reason is "The value could be changed by any other process". -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: Re[3]: [MP3 ENCODER] highq mode
At 17:13 03/02/00 -0600, you wrote: oh one stupid suggestion for the stand alone executable : it could be a good idea to allow "| more" or "/page" or " file.asc" to see the command help, now it "goes away" without seein the beginning of it Yes, that would be very useful, especially for Windows where you cannon scroll up to see the beginning of the help. For Linux this is less important, because of the possibility to scroll up and down. Can windows users redirect stderr like us poor saps stuck using Free Software? Maybe he can try "lame 2 file.asc more file.asc". Well, it works here on OS/2 anyway. Dont know about Windows. Windows Command.com does not support stderr redirection AFAIK, but if you use 4DOS from JPSoftware you can write: Lame | more or better "Lame | List /s" -- Jose Mejuto http://www.cdngo.com CD Audio ripper compressor -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] Question about LAME_Enc.DLL
At 15:07 03/02/00 +0100, you wrote: Hi, I'm trying to use Lame_Enc.DLL. The import of the functions seems to be good. I can get the version Number. But when I want to call beInitStream it fails with an error 2. With BladeEnc.DLL all works well. Did anything change in the structures format, or anything else ? Thanks for answer on [EMAIL PROTECTED] If you can't answer the question, do you know anyone who could... ? Gabriel Hello, You can use Lame_Enc.DLL in the same way as BladeEnc.DLL, of course Lame_Enc.DLL have more options. Check that you do not have a mixture of function pointers if you use the same code. This is my first message in the list. Hello! :-)
