[ardour-users] It is not loud enough

Kevin Cosgrove kevinc at doink.com
Wed Oct 26 12:42:43 PDT 2005

On 26 October 2005 at 9:25, "Don Fredricks" <ardour at handsfull.net> wrote:

> > I'm trying to increase the signal as much as possible during
> > the recording phase.
> This is misguided in a 24-bit world, IMO. In the 16-bit days
> you had to hit close to zero or risk losing signal in the
> quantization noise. If you're using 24-bit converters, your
> signal (at the converter) should average somewhere around
> minus 12 to minus 14 dBFS during tracking, and peaks should
> hit around minus 6 dBFS. This leaves you 6 dB of "insurance"
> headroom.

Yes, I'm coming from a 16-bit world -- see my other post note
referring to an ADAT.  Now I have 24-bit converters in my M-Audio
Delta 1010.  When I trimmed up the signal path my loudest peaks
from a single drum strike would be -2dBFS on the associated
channel -- not much headroom.  When I actually play a song, the
range of maximum peaks across all channels is -10 to -4dBFS.

> > The maximum peak indicators in the mixer window are really
> > great for this.  But, what decibel level corresponds to
> > clipping?
> 0 dBFS at the A/D or D/A converters.

Yippee.  I got that one right.

> > I notice that the mixer peak meters go quite a bit above 0dB.
> This is because internally, Ardour uses a 32-bit floating point
> format to represent digital samples. This gives you 700+ dB of
> headroom *internally*. The signal must be reduced to 0 dBFS
> before it hits the D/A converter, though, or clipping will
> occur.

Makes sense now.

> > Also, waveforms are shown in the editor window.  What scale
> > is the magnitude of those waveforms?  Would 0dB be the
> > top/bottom of the editor stripe? Is it linearly scaled or
> > logarithmicly scaled?  Knowing this will help me to achieve a
> > maximum input level before clipping.
> A larger waveform represents a louder signal. A sine wave at 0
> dBFS would fill the stripe completely. At minus 3 dBFS it would
> occupy half of the stripe. The scale is linear according to the
> voltage that the decibel level represents, if that makes any
> sense to you. So it's both linear and logarithmic, depending on
> how you look at it.

It sure would be easier to have the scale be logarithmic, at
least to me.  The way it is only 1/2 of the scale is really
usable.  Maybe having this be configurable, if only in the
ardour.rc, might be good.  Thoughts?  Yes, your description makes
perfect sense; my day jobs is that of an electronics design

> > If it matters, I'm recording drums and have a soft limiter on
> > each channel which I've set to just start activating at what
> > I think the maximum recording level before clipping would be.
> Limiting past the A/D conversion stage won't stop clipping, as
> I believe someone else has pointed out.

Limiters are before the A/D.

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