Best Practices in Q-SYS Gain Structure (Part 1)
| Site: | QSC |
| Course: | Q-SYS Quantum Level 1 Training (Online) |
| Book: | Best Practices in Q-SYS Gain Structure (Part 1) |
| Printed by: | Guest user |
| Date: | Saturday, 23 November 2024, 1:38 AM |
Description
Video Transcript
00:07
Achieving good gain structure is a key component
00:10
to getting the best audio performance out of your Q-SYS system.
00:13
Unfortunately, there is a lot of misinformation out there about how to achieve it
00:18
so we are going to start from scratch and correct some of those bad habits.
00:22
So let’s start off by establishing some basic vocabulary.
00:26
“Clipping” is the point at which you can no longer reproduce a signal
00:30
that looks or sounds like what is usable.
00:33
The “noise floor” is the base electronic noise of the circuit.
00:37
The signal level between those two points is what’s called “dynamic range”.
00:41
The typical pro audio device has a clipping point of about +24.
00:46
Both Q-SYS and most other DSPs on the market will clip +24 dBu.
00:52
For example, If you look at the Q-SYS CIML4 input card,
00:56
the noise floor around -81 dBu,
00:59
so that means your “dynamic range” is greater than 105 dB assuming clipping at +24.
01:06
If the program material in the audio system is dynamic,
01:10
there could be a lot of variation in the ‘average’ signal level and ‘peak signal level’.
01:15
We want to make sure that peaks will not clip the audio input.
01:19
The space between the average signal level and clipping is called the ‘headroom’.
01:24
Now remember, our ears are not linear.
01:27
In general, it takes about a 10 dB increase in SPL to be perceived as twice as loud.
01:34
If we allow for program peaks to be four times the average signal level,
01:38
which is the industry standard, this means a headroom of 20 dB.
01:43
While we want to leave headroom to avoid clipping,
01:45
we also want to make sure the signal into the device isn’t noisy.
01:49
The amount of signal vs. the amount of base noise in the circuit
01:53
is called the Signal to Noise Ratio, or SNR.
01:57
So in summary, what we want is a signal that is sufficiently above the noise floor,
02:02
but leaves the right amount of headroom before clipping.
02:05
We refer to this as ‘nominal’ level.
02:07
This is the ‘sweet spot’ between those two considerations.
02:11
In standard pro audio circuits, that would be +4dBu.
02:15
Consumer level audio circuits are typically -10dbV nominal (with respect to 1V).
02:21
However, not all of the audio sources that we bring into Q-SYS are going to be +4 nominal, right?
02:27
Microphones, for example, typically come in at around -40 nominal.
02:32
This is why Q-SYS and almost every other DSP makes use of an analog preamp.
02:38
This allows you to increase or decrease the gain of that device
02:42
before it flows into an A-to-D converter,
02:44
at which point we jump from the analog domain to the digital domain.
02:49
An analog preamp typically provides the most control with the least noise.
02:54
A-to-D conversion takes small samples of the analog audio waveform,
02:58
where the signal enters the ‘digital’ domain.
03:00
These samples represent the magnitude of the waveform at that point.
03:05
A signal is typically sampled at a fixed rate.
03:08
According to the Nyquist Theorem,
03:10
you must sample at twice the highest frequency that you plan to reproduce.
03:15
As long as we're sampling at least double that 20k upper limit,
03:19
we shouldn't experience aliasing or have incorrect sample wave form.
03:23
A Q-SYS sampling rate for example is 48k.
03:26
Different A-to-D converters will have different supply voltages, so the input signal is scaled down from that value.
03:33
That introduces the idea of dBfs, or full scale.
03:38
Many integrators get confused because certain DSP manufacturers
03:41
represent their input meters in dBu and train on gain structure accordingly,
03:46
recommending 0dBu nominal input.
03:49
Other DSP platforms such as Q-SYS use input meters in units of dBFS, not dBu.
03:56
So if you apply that other knowledge in Q-SYS,
03:59
every time a microphone has a peak, it would force it into clipping.
04:03
Units really matter in this situation:
04:06
if you look at +4 dBu nominal as compared with +24dBu clipping,
04:12
if we leave +20 dB of headroom in our digital system (0dBFS clipping),
04:18
our nominal level is going to be -20 dBfs.
04:21
If you're a Biamp user and you set that to zero then you've essentially eaten all that headroom
04:27
because your nominal level is essentially 0 dBFS.
04:31
In Q-SYS meters, our nominal becomes -20 dBfs.
04:35
Of course, after we take samples, we can do anything we want, right?
04:39
Q SYS is a floating point DSP, which means that I can apply an infinite amount of gain and attenuation.
04:45
The alternative would be a fixed point DSP,
04:48
which means that introducing gain beyond a certain amount of dynamic range will start to introduce error.
04:53
In Q-SYS, you can only clip samples at the conversion point to and from the analog domain,
04:59
however you can apply infinite attenuation or gain.
05:03
Alright, let's take a break right there and we'll come back for the next section.
Lesson Description
(Part 1 of 4) Learn to do's and don't for setting good gain structure within Q-SYS, particularly within a meeting room scenario.