Maintaining Audio Quality in the Broadcast Facility 2011 - Orban

in the Broadcast/Netcast Facility 25
There are a number of sound card and USB audio devices that suffer from bit-slip
due to a reversed left and right audio clock. The result is digital audio that is not
correctly time-aligned, which causes an interchannel phase shift that increases with
frequency. Consequently, the sum of the left and right channels does not exhibit a
flat frequency response. The amount of attenuation at a given frequency will depend
upon the sample rate. A one-sample slip at 32kHz sample rate will produce a
notch at 16kHz and almost 6dB of loss at 10kHz; 44.1kHz will produce almost
–3dB at 10kHz and–6dB at 15kHz; 48kHz will produce –2dB at 10kHz and –5dB at
15kHz. A one-sample slip is definitely audible in the mono sum, so devices with this
problem are inappropriate for broadcast audio applications, especially for mastering
a library. Many of these devices were based upon a Texas Instruments USB Codec
chip that had its hardware clock reversed. TI has acknowledged the problem and has
released revised parts. However, there are still many audio interfaces and codecs in
use with this problem. This hardware should be scrupulously avoided.
Level metering in sound cards is highly variable. Average, quasi-peak, and peak responses
are all common and often inadequately or incorrectly documented (see
Metering on page 26). This is relevant to the question of line-up level. EBU R68
specifies reference level as –18dBfs, while SMPTE RP 155 specifies it as –20dBfs.
Unless the sound card’s metering is accurate, it is impossible to ensure compliance
with the standards maintained within your facility. Many professional sound cards
have adequate metering, while this is far less common on consumer sound cards.
Further, consumer sound cards often cannot accommodate professional analog levels,
balanced lines, or AES/EBU inputs and outputs.
Bit Accuracy
Digital systems that do not have sample rate converters in their signal path are considered
synchronous if there is no further digital signal processing. A master clock
signal, typically in wordclock or AES11 format, is required for all devices in the audio
signal path to remain in sync. To be bit-accurate, a system must be synchronous and
must not change the word length of the audio data being conveyed.
Digital systems that have sample rate converters in their signal path are not bitaccurate
and are considered asynchronous even though they can also be locked to a
master clock. The advantage to asynchronous digital audio is that just about anything
can be connected to just about anything else without difficulty. However,
there are cautions.
Computer audio can be very confusing, especially when one is trying to achieve bitaccurate
audio. Because audio has many different sources and destinations, computer
operating systems contain software sample rate converters for both record
and play to make things easier for the average user. For the professional, this can
cause performance problems and sample rate conversion can occur when it is not
wanted or expected.
To work around this problem, most professional users use the ASIO interface and
drivers. However, not all audio applications support ASIO. Furthermore, if recording
or encoding from another application is required, this is not supported by ASIO.