[continued from previous message]   
      
   But, even as such, I can note (when downloading and looking for the   
   audio files for LibreQuake) that, even despite being fairly recent, they   
   are still doing most of their sound effects at 11025 (apart from the   
   random straggler files at 22050 or 44100).   
      
      
   Well, anyways, I guess I can fiddle more with trying to use a   
   combination of ADPCM and a pattern table to get to a lower bits-per-sample.   
      
   At least, the decoding process isn't too expensive.   
      
      
   > But everything else is usually very far from perfect.  A particular   
   > issue is the dynamic range - 16-bit linear coding does not have enough   
   > range for a lot of music.  Either quite sounds are "pixelated", losing a   
   > lot of important information, or the dynamic range is compressed before   
   > the CD quality image is generated - giving the music a "flat" sound.   
   > When compressed audio formats are used, they may start off at higher bit   
   > depths and sample rates, but in effect the bit depth also gets   
   > compressed and you lose resolution as well as sample rate and high   
   > frequency information for high compression ratios.  And just as high   
   > jpeg compression produces artefacts for some images, such as ghosting,   
   > so does high audio compression.   
   >   
      
   Yeah, I am aware that seemingly a lot of music uses compression to try   
   to achieve a sort of semi-uniform "loudness wall".   
      
   I can note that when looking at music in an audio program, it tends to   
   use the full amplitude range for pretty much the whole song.   
      
   Contrast, the audio from TV shows tends to use closer to around 25% to   
   33% of the amplitude range, and with a lot more variability in loudness   
   between sections.   
      
      
   Though, if there is one merit to compression, it does make it easier to   
   hear low frequencies.   
      
   So, while a 440Hz pure sine wave is very quiet; a 440Hz sine wave fed   
   through the compressor filter is a lot louder, even if the overall   
   amplitude hasn't really changed much. Possibly because compression often   
   makes the shape of a sine wave closer to that of a square wave.   
      
      
      
   Ironically, one could almost make a case here for using an A-Law variant   
   with the low-order bits XOR'ed with the sign, in which case it could   
   function as both a higher dynamic-range format and as 8-bit PCM.   
      
   Though, using it as 8-bit PCM would still have the "noise floor"   
   annoyance; where it almost invariably adds a low intensity hiss to the   
   audio.   
      
   --- SoupGate-Win32 v1.05   
    * Origin: you cannot sedate... all the things you hate (1:229/2)