Telos Zephyr Xstream User Manual page 130

USER'S MANUAL
AAC-LD (AAC Low Delay)
When announcers use codecs for broadcast remote applications, they often need to have
natural two‐way interaction with other program participants located back at the studio, or
callers. Because it is a hot topic for engineers working in the field of Internet telephony, a
number of studies have been conducted to determine user 's reactions to delays in telephone
conversations. The data apply directly to the application of professional codecs to remotes, so it
is interesting to take a peak over the shoulder of the telecom guys to see what they have
learned.
For broadcast remotes, we try to arrange our system so that there is no path for the field
announcer's voice to return to his/her headphones. Nevertheless, sometimes echo is
unavoidable. For example, this can happen when a telephone hybrid has leakage or when a
studio announcer has open‐air headphones turned‐up loud and the audio makes its way into the
studio microphone.
When there is no echo, it has been discovered that anything less than 100 ms one‐way delay
permits normal interaction between participants. Between 100 and 250 ms is considered
"acceptable." ITU‐T standard G.114 recommends 150 ms as the maximum for "good"
interactivity. Echo introduces a different case. As you might expect, echo is more or less
annoying depending upon both the length of time it is delayed and its level. Telephone
researchers have measured and quantified reactions, and ITU‐T G.131 reports the findings and
makes recommendations.
Summary of ITU-T G.13, with recommendations for designers of telephone systems that must cope
with echo. This shows Talker Echo Loudness Rating vs delay.
There are codecs using other than perceptual technologies that have lower delay, but they are
not as powerful. That is, for a given bitrate, they do not achieve fidelity as good as the MPEG
systems we have been examining. The common G.722 is an example. It uses ADPCM (Adaptive
Delta Pulse Code Modulation), which can have delay as low as 10 ms, but with much poorer
quality. So the question arises: Is it possible to have high quality and low delay in the same
Section 6: AUDIO CODING REFERENCE 118