Conservation of power is a critical design goal in many wireless products, including wireless headsets for digital radio communications. One "power-wise" approach to wireless audio transmissions involves sampling, digitizing, and compressing audio signals before they are transmitted as the modulated portion of a RF carrier. Upon reception, these signals are decompressed and converted back to analog audio signals. By using time-division-duplex (TDD) techniques, it is possible to realize a low-cost, half-duplex RF transceiver (the CC1000 from Chipcon) that can transmit and receive audio signals with a low-component-count design that is suitable for wireless headsets.

Stricter regulations regarding the vehicular use of mobile telephones, as well as fear of RF radiation close to the brain, have prompted increased sales of wireless headsets for use with mobile telephones. Bluetooth has been touted as a "driving" standard for such applications as wireless headsets, but it remains to be seen if manufacturers of Bluetooth semiconductors can meet demanding market requirements for a low price. As a result, designers of wireless headsets are anxiously seeking low-cost, low-power alternatives to Bluetooth integrated circuits (ICs), such as the single-chip model CC1000 RF transceiver from Chipcon (Oslo, Norway).

Based on 0.35-µm complementary-metal-oxide-semiconductor (CMOS) technology, the CC1000 represents a combination of low cost, high integration, high performance, and flexibility in a low-power device. Designed primarily for frequency-shift-keying (FSK) systems in the industrial-scientific-medical (ISM) bands at 315, 433, 868, and 915 MHz, the single-chip transceiver can be programmed for use from 300 to 1000 MHz. It operates on supply voltages from +2.1 to +3.6 VDC, and only consumes 7.4 mA at +3 VDC in receive mode. The leakage current is only 0.2 µA, ensuring miniscule draw on batteries during nonoperating times. The transceiver chip features receiver (Rx) sensitivity of −110 dBm (at 433 MHz and 1.2 kb/s) supports data rates from 0.6 to 76.8 kb/s.

In a digital telephone system, analog voice signals are converted to digital signals through pulse-code modulation (PCM), using three steps: sampling, quantizing, and coding. Band-limited (4-kHz-wide) voice signals are sampled at a rate of 8 kHz (in agreement with Nyquist theory). The amplitude of the voice signal is sampled (measured) every 125 µs. Each sample is then quantized (or truncated) into a number, usually an integer. For example, if 13 b is used, this integer could be between −4096 and +4095. Coding is the way these quantization levels are represented as digital numbers. For example, the coding method known as "two's complement" may be used to express negative, as well as positive, numbers. Most of the information captured for human speech has a small-signal character, contained within higher-resolution, smaller amplitudes. For less-likely larger-amplitude voice signals, the use of a uniform quantizer (equally spaced digitizing steps) provides high, but unnecessary, quality. The uniform quantizer can also yield pronounced truncation effects for the more frequent small-amplitude signals. As a result, using nonuniform quantization provides a system that is more appropriate for human speech.

Nonuniform quantization can be achieved by passing a voice signal through a compressor at the transmit end, and then passing the signal though an expander on the receiving end as part of a process known as "companding." By using companding, the required code-length word can be reduced from 13 b to 8 b or less, while retaining the subjective quality of the voice signal. Companding can be performed in hardware in a coder/decoder (codec) circuit or in software using a look-up table or a real-time calculation. Two international standards for creating 8-b encoded data are u-law and A-law. In the US and Japan the accepted standard is u-law, while A-law is used in Europe.¹ The 8-kHz sample rate combined with companding leads to an 8-b code word. The digital voice stream is therefore said to be 64 kb/s.

A public-telephone network is an example of a full-duplex system, where speech is transmitted in both directions at the same time. In a half-duplex system, speech travels only in one direction at one time (similar to a walkie-talkie system). As an extension of the public-phone network, a wireless headset must, therefore, be full duplex. Full-duplex systems require more complex circuit solutions than half-duplex systems, where the Rx and transmitter (Tx) can share many of the system function blocks. A half-duplex approach can save space and cost.

By using TDD, full-duplex operation can be achieved with half-duplex cost and simplicity. Using TDD, the signal is transmitted one way at a time, but the direction of transmission is switched very fast with a small latency (time delay). As long as the latency is in the order of 100 ms or less, the human ear will not detect it and a normal conversation can occur. In comparison, satellite telephone systems have a large delay, making a normal conversation flow more difficult. Using TDD makes it necessary to buffer the digital voice stream from one party while the other is transmitting. A voice stream of 64 kb/s, therefore, would require a wireless TDD data link of at least 128 kb/s. The turn-around time from Rx to Tx would require an even higher data rate.