Re: [linux-audio-dev] Simple Jack mixing problem - Linux-audio-dev

5 Jan 2007

Hi thanks for answering,

Ok i think i'll leave it to the user, because the goal is to make a simple
sampler with kit/instrument creation and control.

Flo

David Olofson a écrit :
...
  On Friday 05 January 2007 01:44,
florentberthaut(a)no-log.org wrote:
  Hi everyone,

 I'm working on a sampler and i have a really silly problem.

 I simply want to mix sounds in a stereo buffer within the
 jack "process" function.
 So for each frame i sum the samples from each sound but then the
 values go above 1 and i get clipping.
 So my first thought was to divide by the number of samples playing,
 but it ends up with a decrease in volume.
 I tried to limit the values between -1 and 1 but it 's also
 clipping -> distortion. 
 Well, that's what JACK is doing anyway - or there would be wrapping in
 the float -> int conversion, which sounds a lot worse than clipping.
 It's just not possible to represent larger amplitudes than [-1, 1] in
 the integer domain (that is, [-32768, 32767] or whatever, depending
 on resolution), which is what DACs deal with.

  I looked into some sources and it seems i would
actually just have
 to sum the samples. 
 Yep, that's all there is to it, really - and it's the user's
 responsibility to keep volumes low enough to avoid driving the output
 into clipping.

 This is true on hardware synths and samplers too. The only reason some
 of them appears not to have this problem is that they use various
 tricks to hide it.

 For example, the old Roland JV-1080 had 18 bit DACs, where the two
 extra bits were pretty much used for extra headroom above
 the "normal" max amplitude of a single voice. Even so, it was pretty
 easy to drive it into clipping by using resonant filters and/or lots
 of loud voices.

 I wouldn't be surprised if some synths and samplers use 20 och 24 bit
 DACs for even more headroom, waveshaping to make the last 12 dB or so
 non-linear (soft saturation, sort of), dynamic output gain control
 and stuff like that.

 Now, if you're dealing in digital, extra DAC bits and dynamic output
 gain control is out, obviously. (Goes for plugins and JACK clients as
 well as hardware devices with digital outputs.) So, either you set
 your sampler's 0 dB level at -12 dB or something (allows mixing four
 maximized samples at maximum volume without clipping), or you leave
 that to the user.

 If you really want to make the impression sqeezing in more than
 there's room for, you'll have to add some dynamics processing. There
 is of course no way of doing this without some sort of distortion,
 but even a simple waveshaper to "flatten out" peaks above 12 dB below
 clipping sounds a great deal better than hard clipping.

 However, I'd rather see that kind of stuff left to the user. Those of
 us who use 20+ bit sound cards would rather have it all linear and
 undistorted right into the amplifier. There is always the option of
 inserting some serious multiband compressor last thing in the JACK
 chain. Might actually be a good idea anyway, if you like playing loud
 and don't want to ruin your ears if some synth or effect freaks out.

 //David Olofson - Programmer, Composer, Open Source Advocate

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