The Truth About Digital Audio Latency PDF

PreSonus | Learn - The Truth About Digital Audio Latency
3/11/13 9:28 PM
The Truth About Digital Audio Latency

LEARN
By Wesley Elianna Smith
In the audio world, latency is another word
for delay. Its the time it takes for the sound
from the front-of-house speakers at an outdoor festival to reach you on your picnic blanket. Or the time it takes for your finger to
strike a piano key, for the key to move the
hammer, for the hammer to strike the string,
and for the sound to reach your ear.
Your brain is wired so that it doesnt notice if sounds are delayed 3 to
10 milliseconds (ms). Studies have shown that sound reflections in an
acoustic space must be delayed by 20 to 30 ms before your brain will
perceive them as separate. However, by around 12 to 15 ms (depending on the listener), you will start to feel the effects of a delayed signal. It is this amount of delay that we must battle constantly when
recording and monitoring digitally.
When Good Latency Goes Bad

Roundtrip latency in digital-audio applications is the amount of time it
takes for a signal (such as a singing voice or a face-melting guitar solo)
to get from an analog input on an audio interface, through the analogto-digital converters, into a DAW, back to the interface, and through
the digital-to-analog converters to the analog outputs. Any significant
amount of latency can negatively impact the performers ability to
http://www.presonus.com/community/Learn/The-Truth-About-Digital-Audio-Latency
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play along to a click track or beat making it sound like theyre performing in an echoing tunnel (unless they have a way to monitor
themselves outside of the DAW application, such as a digital mixer or
one of our AudioBox VSL-series interfaces).
Whats Producing the Delay: a Rogues Gallery

In practical terms, the amount of roundtrip latency you experience is
determined by your audio interfaces A/D and D/A converters, its internal device buffer, its driver buffer, and the buffer setting you have
selected in your digital audio workstation software (Mac) or Control
Panel (Windows).
Converters. Analog-to-digital converters in your interface transform
an analog signal from a microphone or instrument into digital bits
and bytes. This is a ferociously complex process and takes a little more
than half a millisecond on average. On the other end of a long chain
were about to describe are the digital-to-analog converters that
change the digital stream back into electrical impulses you can hear
through a monitor speaker or headphones. Add another half-millisecond or so.
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Buffers. A buffer is a region of memory storage used to temporarily

hold data while it is being moved from one place to another. There are
four of these in the digital signal chain.
USB Bus Clock Front Buffer
ASIO (Driver) Input Buffer
ASIO (Driver) Output Buffer
USB Clock Back Buffer
Each buffer contributes to the total delay present between the time
you play that hot guitar solo and the time you hear it back in your
headphones.
Fast Drivers and Slow Drivers

The biggest variable that contributes to how long this process will
take is driver performance.
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In computing, a driver is a
computer program allowing
higher-level computer programs to interact with a
hardware device. For example, a printer requires a driver to interact with your
computer. A driver typically
communicates with the de- Studio One buffer setting.
vice through the computer
bus or communications subsystem to which the hardware connects.
Drivers are hardware-dependent and operating-system-specific.
One of the primary goals for engineers who design audio-interface
drivers is to provide the best latency performance without sacrificing
system stability.
Imagine that youre playing an old, run-down piano and that there is
a catch in the hammer actionso great a catch, in fact, that when you
strike a key, it takes three times longer than normal for the hammer to
strike the string. While you may still be able to play your favorite
Chopin etude or Professor Longhair solo, the feel will be wrong because youll have to compensate for the delayed hammer-strikes.
You will have a similar problem if the buffer-size setting is too large
when you overdub a part while monitoring through your DAW.
Take a Couple Buffers and Call Us in the Morning

A buffer is designed to buy time for the processor; with the slack the
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buffer provides, the processor can handle more tasks. When the buffer
size is too large, its delaying the dataadding latencymore than is
necessary for good computer performance.
But if the buffer size is too small, the processor has to work faster to
keep up, making it more vulnerable to overload, so your computerrecording environment becomes less stable.
Consider this scenario: Youre playing your favorite virtual instrument, trying to add one more pad part to a nearly finished song. All
42 tracks are playing back, and all of them use plug-ins. And then it
happens: Your audio starts to distort, or you start hearing pops and
clicks, or, worse, your DAW crashes because your CPU is overloaded.
The 64-sample buffer size you have set, in conjunction with the
amount of processing that your song requires, overtaxes your computer.
If you increase the buffer size, you can get the software crashing to
probably go away. But its not that simple.
The more that you increase the buffer size for example, up to 128
samples the more you notice the latency when trying to play that
last part. Singing or playing an instrument with the feel you want becomes extremely difficult because you have essentially the same problem as with that rickety pianos delayed hammer-strikes. What you
play and what you hear back in your headphones or monitor speakers
get further and further apart in time. Latency is in the way. And
youre in that echo-y tunnel again.
Lets look at our piano example again, this time with a fully functioning baby grand and not that antique piano in desperate need of repair.
For simplicitys sake, lets pretend that there is no mechanical delay
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between the time your finger strikes the key and the hammer strikes
the string. Sound travels 340 meters/second. This means that if youre
sitting one meter from the hammer, the sound will not reach your ears
for a little more than 3 ms. So why does 3 ms not bother you a bit
when youre playing your grand piano, but a buffer setting of 2.9 ms
(128 samples at 44.1 kHz) in your DAW make it virtually impossible
for you to monitor your guitar through your favorite guitar amp modeling plug-in?
Decoding Latency
As mentioned earlier, roundtrip latency is the amount of time it takes
for a signal (such as a guitar solo) to get from the analog input on an
audio interface, through the A/D converters, into a DAW, back to the
interface, and through the D/A converters to the analog outputs. But
you can only control one of part of this chain: the input latencythat
is, the time it takes for an input signal such as your guitar solo to
make it to your DAW.
This is where driver performance enters the picture. There are
two layers to any isochronous driver (used for both FireWire
and USB interfaces). The second layer provides the buffer to
Core Audio and ASIO applications like PreSonus Studio
OneTM and other DAWs. This is the layer over which you
have control.
To make matters worse, you usually are not given this buffersize setting as a time-based number (e.g., 2.9 ms); rather, you
get a list of sample-based numbers from which to choose (say,
128 samples). This makes delay conversion more complicated. And
most musicians would rather memorize the lyrics to every Rush song
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than remember that 512 samples equates to approximately 11 to 12 ms

at 44.1 kHz! (To calculate milliseconds from samples, simply divide
the amount of samples by the sample rate. For example, 512 samples/44.1 kHz = 11.7 ms.)
The buffer size that you set in your DAW (Mac) or in your devices
Control Panel (Windows) determines both the input and the output
buffer. If you set the buffer size to 128 samples, the input buffer and
the output buffer will each be 128 samples. At best, then, the latency is
twice the amount you set. However, the best case isnt always possible
due to the way audio data is transferred by the driver.
For example, if you set your ASIO buffer size to 128 samples, the output latency could be as high as 256 samples. In that case, the two
buffers combine to make the roundtrip latency 384 samples. This
means that the 2.9 ms of latency you set for your 44.1 kHz recording
has become 8.7 ms.
The analog-to-digital and digital-to-analog converters in an audio interface also have latency, as do their buffers. This latency can range
from 0.2 to 1.5 ms, depending on the quality of the converters. An increase of 1 ms of latency isnt going to affect the quality of anyones
performance. However, it does add to the total roundtrip latency. For
our 128-sample example setting, adding 0.5 ms for each converter
brings the roundtrip latency to 9.7 ms. But 9.7 ms is still below the
realm of human perception, and it shouldnt affect your performance.
So Where Does the Extra Delay Really Come From?

The culprit is that first mysterious audio-driver layer that no one ever
discusses. This lowest layer has no relationship to audio samples or
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sample rate. In the case of USB, it is a timer called the USB Bus clock.
(There is a similar clock for FireWire processes but we will only discuss the USB Bus clock here.)
The USB Bus clock is based on a one-millisecond timer. At
an interval of this timer, an interrupt occurs, triggering the
audio processing. The problem that most audio manufacturers face is that without providing control over the lower-layer buffer, users cannot tune the driver to the computer as tightly
as they would like. The reason for not exposing this layer is simple:
The user could set this buffer too low and crash the drivera lot.
To get around this, most manufacturers fix this buffer at approximately 6 milliseconds. Depending on the audio driver, this could be 6 ms
input latency and 6 ms output latency. But like the ASIO buffer discussed earlier, even if these buffer sizes are set to the same value, the
resulting output latency can differ from the input latency.
For our example, lets keep things simple and say that latency is 6 ms
in both directions. Our mystery is solved: With most audio interfaces,
there is at least 12 ms of roundtrip latency built into the driver before
the signal ever reaches your DAW, in addition to the 9.7 ms latency
we calculated earlier.
Thus, you set 2.9 ms of delay in your DAW and end up with 21.7 ms
of roundtrip latency. (All of the numbers in our examples are based
on averages. However, some manufacturers are able to optimize driver performance to minimize these technical limitations.)
Overcoming the Problem

Many audio-interface manufacturers have solved the problem of monhttp://www.presonus.com/community/Learn/The-Truth-About-Digital-Audio-Latency
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itoring latency through a DAW by providing zero-latency monitoring

solutions onboard their interfaces.
One of the earliest solutions was the simple analog Mixer knob on the
front panel of the PreSonus FirePod. This allowed users to blend the
FirePods analog (pre-converter) input signal with the stereo playback
stream from the computer. This basic monitoring solution is still available on such interfaces as the PreSonus AudioBox USB, AudioBox
22VSL, and AudioBox 44VSL. Another solution, used in the PreSonus
FireStudio family and many others, is to include an onboard DSP
mixer that is managed using a software control panel.
While both of these solutions resolve the problem of latency while
monitoring, they provide a flat user experience by giving control only
over basic mix functions like volume, panning, solo, and mute.
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Anyone who has ever recorded using one of our StudioLive mixers (anyone who
has ever tracked with any
mixer, for that matter) knows
how important it is to be able
to record a track while hear- Old-skool solution: Just grab some of the analog signal
before it goes into the A/D converters and send it back to
ing effects (as well as com- your headphones. It works but you cant hear any effects or
pression and equalization). reverb.
For example, if reverb on a vocal is going to be part of the final mix,
its almost impossible to record the vocal dry phrasing and timing are totally different when you cant hear the duration and decay of
the reverb.
The developers at PreSonus were intrigued by the idea that they could
conceivably provide the user with some level of control over the USB
Bus clock buffer and perhaps offer another way of monitoring outside
the DAW (while adding effects and reverb). After much experimentation, they discovered that most modern computers can easily and stably perform at a much lower USB Bus clock buffer than previously
thought. On average, a 2 to 4 ms USB Bus clock buffer offers both excellent performance and stability. On a powerful computer like a fully
loaded Mac Pro, theyve been able to lower this buffer to the lowest
USB Bus clock setting possible: 1 ms.
Given these discoveries, not giving the user control over the USB Bus
clock buffer and telling them that the only latency controls available
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are the ASIO and Core Audio buffer sizes seems at best duplicitous,
and at worst a failure to provide customers with the best latency performance a modern computer can provide.
This is where AudioBox VSL-series interfaces enter the picture. This
new series of interfaces takes advantage of these technological discoveries and provides users with the ultimate monitor-mixing experience,
without including expensive onboard DSP and the proportional cost
increase to customers.
Tracking with Reverb and Effects... without Being in a Tunnel

The Virtual StudioLive software that comes with our AudioBox
22VSL, 44VSL and 1818VSL interfaces looks like and performs like
the Fat Channel on our StudioLive 16.0.2 mixer.
You get compression, limiting, 3-band semi-parametric EQ, noise
gate, and high-pass filter. Weve even included 50 channel presets
from the 16.0.2 just to get you started. Plus you get an assortment of
32-bit reverbs and delay, each with customizable parameters.
Optimizing AudioBox VSL Software

AudioBox VSL monitoring software runs between the USB Bus clock
buffer and the ASIO/ Core Audio buffer on your computer, so it is
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only subject to the latency

from the USB Bus clock
buffer.
Unlike many manufacturers,
PreSonus did not fix this
buffer at 6 ms; rather, AudioBox VSL offers a choice of
three buffer sizes. To reduce AudioBox 44VSL Virtual StudioLive mixer and Fat Channel
the confusion of presenting
the user with two types of buffer settings, these USB Bus clock buffer
settings are labeled Performance Mode.
This setting is available from the Setup tab in AudioBox VSL, and it
directly affects the amount of latency you will hear in monitor mixes
from AudioBox VSL software.
At the Fast setting, AudioBox VSL runs at a USB Bus clock buffer setting of 2 ms, while Normal sets the buffer to 4 ms, and Safe sets it to 8
ms. So when you set your AudioBox VSL to run at the Fast USB Bus
clock buffer setting, roundtrip latency will be approximately 3.5 ms,
including the time it
takes for the A/D D/A converters to change analog audio to 1s and
0s and back to analog again.
To optimize these buffer settings for your particular computer:
Begin by creating a monitor mix in AudioBox VSL and setting the
Performance mode to Fast.
Listen carefully for pops and clicks and other audio artifacts at a
variety of sample rates.
Now load the AudioBox VSL with compressors, EQs, reverbs, and
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delays.
If you hear audio artifacts, raise the Performance mode to Normal.
On most machines, Normal will provide the best
performance with the
most stability. If you
CPU Performance Meter in Studio One 2 Artist DAW (comes
have an older machine free with AudioBox VSL interfaces).
with a slower processor
and a modest amount of RAM, you may need to raise this setting
to Safe. Keep in mind, however, that even at 9 ms, AudioBox VSL
is running at a lower latency than monitoring through most
DAWs at the best ASIO/ Core Audio buffer settingand the best
buffer setting will not work on a slower computer anyway.
Once you have Performance mode tuned, the next latency component of the driver to tune is the ASIO buffer size (Windows) or
Core Audio buffer size (Mac). This time, load a large session into
your DAW and experiment with the buffer settings. Again, you
are listening for pops and clicks and other audio artifacts.
If your DAW includes a CPU-performance meter (as Studio One

does), you can use this to help you find the best buffer setting for your
computer.
No matter how you set your ASIO/Core Audio buffer size, the monitoring latency in VSL is not affected. So you can set this buffer fairly
high and lower it only when you are playing virtual instruments.
Keep in mind that its still important to determine the lowest threshhttp://www.presonus.com/community/Learn/The-Truth-About-Digital-Audio-Latency
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old at which your DAW can still perform stably.
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