Digital Sound & Music: Concepts, Applications, & Science, Chapter 5, last updated 6/25/2013
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An issue we're not considering in this section is applying dynamic range compression as
one of the final steps in audio processing. We mentioned that the dynamic range of car radio
music's listening environment is about 25 dB. If you play music that covers a wider dynamic
range than 25 dB on a car radio, a lot of the soft parts are going to be drowned out by the noise
caused by tire vibrations, air currents, etc. Turning up the volume on the radio isn't a good
solution, because it's likely that you'll have to make the loud parts uncomfortably loud in order to
hear the quiet parts. In fact, the dynamic range of music prepared for radio play is often
compressed after it has been recorded, as one of the last steps in processing. It might also be
further compressed by the radio broadcaster. The dynamic range of sound produced for theatre
can be handled in the same way, its dynamic range compressed as appropriate for the dynamic
range of the theatre listening environment. Dynamic range compression is covered in Chapter 7.
5.2.3 Latency and Buffers
In Section 5.1.4, we looked at the digital audio signal path during recording. A close look at this
signal path shows how delays can occur in between input and output of the
audio signal, and how such delays can be minimized.
Latency is the period of time between when an audio signal enters a
system and when the sound is output and can be heard. Digital audio systems
introduce latency problems not present in analog systems. It takes time for a
piece of digital equipment to process audio data, time that isn't required in fully
analog systems where sound travels along wires as electric current at nearly the
speed of light. An immediate source of latency in a digital audio system arises
from analog-to-digital and digital-to-analog conversions. Each conversion adds
latency on the order of milliseconds to your system. Another factor influencing latency is buffer
size. The input buffer must fill up before the digitized audio data is sent along the audio stream
to output. Buffer sizes vary by your driver and system, but a size of 1024 samples would not be
usual, so let's use that as an estimate. At a sampling rate of 44.1 kHz , it would take about 23 ms
to fill a buffer with 1024 samples, as shown below.
Thus, total latency including the time for ADC, DAC, and buffer-filling is on the order of
milliseconds. A few milliseconds of delay may not seem very much, but when multiple sounds
are expected to be synchronized when they arrive at the listener, this amount of latency can be a
problem, resulting in phase offsets and echoes.
Let's consider a couple of scenarios in which latency can be a problem, and then look at
how the problem can be dealt with. Imagine a situation where a singer is singing live on stage.
Her voice is taken in by the microphone and undergoes digital processing before it comes out the
loudspeakers. In this case, the sound is not being recorded, but there's latency nonetheless. Any
ADC/DAC conversions and audio processing along the signal path can result in an audible delay
between when a singer sings into a microphone and when the sound from the microphone
radiates out of a loudspeaker. In this situation, the processed sound arrives at the audience’s ears
after the live sound of the singer’s voice, resulting in an audible echo. The simplest way to
reduce the latency here is to avoid analog-to-digital and digital-to-analog conversions whenever
possible. If you can connect two pieces of digital sound equipment using a digital signal
Flash
Tutorial:
Reducing
Latency
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