Re: Mixing manually vs. Mixer AudioUnit and avoiding clipping
Re: Mixing manually vs. Mixer AudioUnit and avoiding clipping
- Subject: Re: Mixing manually vs. Mixer AudioUnit and avoiding clipping
- From: Admiral Quality <email@hidden>
- Date: Fri, 18 Jun 2010 16:16:35 -0400
Exactly like David says...
- AQ
On Fri, Jun 18, 2010 at 4:07 PM, David Reaves <email@hidden> wrote:
> When summing multiple sources of audio of equal volume, but whose signals
> are NOT coherent (i.e., totally unrelated program in each source), they will
> not sum to a 6dB (2.0x) increase as would identical programs, but rather 3
> dB (1.414x). In other words it is more like a summation of power, rather
> than voltage.
>
> So a mixer can sum channels, pretty reliably without clipping, with a
> reduction of input gain of -3 dB (0.707) per input more than one (-3dB * N-1
> inputs).
>
> Example:
> 2 sources summed: each source's gain in to the mixer should be dropped by 3
> dB (multiply each by 0.707).
> 3 sources: each should be dropped by 6 dB (multiply each by 0.707 squared,
> which is 0.5).
> 4 sources: each should be dropped by 9 dB (multiply each by 0.707 cubed,
> which is about 0.35), etc.
>
> If you have, say, 30 inputs, you could seriously compromise the signal to
> noise ratio by following that rule, so typically you use the number of
> inputs that are *actually* used at any one time, rather than the total
> number of available inputs.
>
> Also, if you have adequate internal overhead above 1.0, you can more simply
> multiply the summation rather than each individual input.
>
> Dynamic processing is not needed or warranted.
>
> David Reaves
>
>
>
> On Fri, 18 Jun 2010 11:58:57 -0700, <email@hidden> wrote:
>>
>> On Jun 18, 2010, at 11:44 AM, Gregory Wieber wrote:
>>
>>> You have to think in terms of the max output signal being "1". You need
>>> to divide that by the number of busses you have and set the volume for each
>>> bus accordingly.
>>
>>
>> I tried something like that but it didn't work very well. Take, for
>> instance, the case where you have one note playing, 5 notes are silent and
>> two notes are playing at very low levels. Dividing the mixed signal by 8 is
>> not going to sound right.
>>
>> I think the trick is to reduce the volume of the very high notes, but not
>> to reduce the lower notes.
>>
>> Google has pointed me to some articles on compression or Dynamic Range
>> Compression... maybe that's the direction I need to move in...
>>
>>
>
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