Telos ZephyrExpress user manual Overview, Introduction to Audio Coding

OVERVIEW

Introduction to Audio Coding

Audio takes up a lot of data.

Without data reduction, CD-quality quality audio — 16 bits at 44.1kHz sample rate — requires a transmission capacity of about 705 thousand bits per second (kbps) for each

?audio channel. But the wires we use for remote broadcasting are on a telephone system designed for voice-grade communications: 8 bits at 7kHz sample rate, or 56kbps per channel. That’s less than 8% of what we need.CURIOSITY NOTEYou can arrive at these same numbers with nothing more complicated than

grade-school math. Just multiply the sample rate by the sample depth: 44,100 samples per second * 16 bits per sample = 705,600 bits per second for CD-quality mono audio.

You can reduce the data requirements by lowering the quality somewhat. 13 bits would yield a respectable 78 dB dynamic range, certainly adequate for home listening. And a 32kHz sample rate — with careful equipment design — will give you flat response to 15kHz, the practical limit for analog FM broadcasting in North America. Unfortunately,

? that still leaves us with telephone data channels about 86% too small to do the job. Besides, 13 bits is an awkward bit depth for computers to deal with, and the audio it produces isn’t clean enough to survive today’s transmitter processors.

CURIOSITY NOTE

Bit depth and sample rate translate easily into audio specifications. Digital audio must have a sample rate of at least twice the desired bandwidth, so 15kHz audio requires (after a safety margin) 32kHz sampling. Each bit of sample depth represents slightly more than 6dB of dynamic range.

The first practical coding methods used a principle called ADPCM, Adaptive Delta Pulse Code Modulation. This takes advantage of the fact that it takes fewer bits to code the difference, or delta, between successive audio samples compared to using the individual values. Further efficiency is had by adaptively varying the difference comparitor according to the nature of the program material. G.722 and APT-X are examples of ADPCM schemes. They achieve around a factor of 4 reduction in bitrate.

G.722 achieves additional efficiency by allocating its bits to match the patterns in the human voice, and it’s considered adequate for news and talk programming over ISDN. But for high-fidelity transmission, algorithms with more power are required. These are based on psychoacoustics, where the coding process is adapted to the way we hear sounds. There are several algorithms available, with varying complexity and performance levels.

Some years ago, the international standards group ISO/IEC established the ISO/MPEG (Moving Pictures Expert Group), to develop a universal standard for encoding moving pictures and sound for digital storage and transmission media. The standard was

5-2 CODING