Short-range wireless digital voice transmission is used extensively in contemporary consumer electronics. Products such as cordless telephones, wireless headsets (for mobile and landline telephones) and baby monitors are just a few of the items that use digital techniques to wirelessly communicate voice information.
Wireless environments are inherently noisy, so the voice coding scheme chosen for such an application must be robust in the presence of bit errors.
Pulse coded modulation (PCM) and its derivatives are commonly used in wireless consumer products for their compromise between voice quality and implementation cost, but these schemes are not particularly robust in the presence of bit errors. Adaptive delta modulation (ADM) is a mature technique that should be considered for these applications because of its bit error robustness and its low implementation cost.
ADM is a voice coding technique that quantises the difference between the current sample and the predicted value of the next sample. It uses a variable 'step height' to adjust the predicted value of the next sample so that both slowly and rapidly changing input signals can be faithfully reproduced. One bit is used to represent each sample in ADM [1]. The one-bit-per-sample ADM data stream requires no data framing, thereby minimising the workload on the host microcontroller.
As mentioned before, wireless environments are inherently noisy, and bit errors are present in any wireless application. Most voice coding techniques provide good audio quality in an ideal operating environment, but the challenge is to provide good audio quality in the presence of the inevitable bit errors.
Traditional performance metrics (eg, signal-to-noise ratio) do not accurately measure perceived audio quality for various voice coding methods and input signals [2]. Mean opinion score (MOS) testing overcomes the limitations of other metrics by successfully quantifying perceived audio quality. The MOS testing uses a scale of 1 to 5 to represent audio quality, with 1 representing very bad speech quality and 5 representing excellent speech quality. A MOS score of 4 or higher represents 'toll quality' speech, which is equivalent to that audio quality obtained during a traditional telephone call [3].
Figure 1 illustrates the relationship between MOS scores and bit errors for three different voice coding schemes, CVSD (continuously variable slope delta - a member of the ADM family), μ-law PCM and ADPCM. While the perceived audio quality of all three schemes degrades as the number of bit errors increases, the graph indicates that ADM (in the form of CVSD) sounds better than the other schemes as bit errors increase.
![Figure 1. MOS comparison for various voice coding methods [3]](articles/Dataweek - Published by Technews/dw3273a.png)
Since ADM provides robust performance in the presence of bit errors, error detection and correction typically are not used in an ADM design, and this contributes further to a reduction in host processor workload. The superior noise immunity, coupled with a significantly reduced processor workload, strongly supports consideration of ADM as a voice coding method for wireless applications.
The benefits of ADM for wireless applications are demonstrated in the following example reference design. This small form-factor, low-power design includes all of the building blocks necessary for a complete wireless voice product, including:
* ADM voice codec.
* Microcontroller.
* RF transceiver.
* Power supply including rechargeable battery.
* Microphone, speaker, amplifiers, etc.
* Schematics, board layout files and microcontroller code written in C.
This design was implemented on a 2,5 cm x 3,3 cm four-layer PCB, as shown in Figure 2. The use of smaller components and tighter spacing can allow an even smaller form factor than that pictured. The block diagram for this ADM reference design is illustrated in Figure 3.
The design concept utilises a master unit and slave unit that communicate wirelessly in a licence-free RF band. Time-division duplexing (TDD) is used to simulate a full-duplex communication link.
Operation is similar on both the master and slave units - see Figure 4. The CML Microcircuits CMX649 ADM voice codec encodes the microphone input with 27,8 Kbps ADM voice coding. The encoded voice data is passed to the Texas Instruments MSP430F1232 microcontroller for 3B4B coding and frame formatting. The coded data frames are then passed to the Micrel MICRF505 RF transceiver for filtering, up-conversion and transmission in a licence-free band. The transmitted data rate is 100 Kbps, and the exact channel frequency is selected by software configuration.
After reception, the RF transceiver passes the recovered data frames to the microcontroller for de-formatting. The microcontroller then sends the raw data stream to the voice codec for signal reconstruction, and the recovered audio can be heard on a speaker.
Data framing is used in this project for TDD, which is required to achieve the perceived full-duplex effect. Data is transferred over the RF link in a half-duplex manner, but the perceived effect is full-duplex because the MICRF505's data rate is more than twice that of the CMX649 ADM voice codec. The TDD scheme is implemented with a firmware-based buffering scheme that manages the difference in data rates between the RF transceiver and the voice codec. The microcontroller's on-chip memory is used to buffer the transmitted and received voice data until ready for further processing.
In addition to the TDD scheme, the microcontroller firmware also accomplishes a pairing procedure that causes the master and slave units to communicate together while ignoring signals from other sources. The CMX649's voice activity detector is used to determine when the circuit is placed into sleep mode. If voice is absent for more than 23 seconds, the boards will enter power save mode. The master and slave boards, if paired, will wake from power save mode when voice is present at the audio input to the master board.
The CMX649 was selected for this design because of its robust ADM voice coding, low power consumption, and highly integrated feature set. The CMX649 offers extensive flexibility in its ADM voice coding settings, and the settings chosen for this project were empirically derived to offer optimal voice quality in this application. In addition to performing ADM voice coding, other CMX649 features used in this design include:
* Programmable anti-alias and anti-image filtering.
* Microphone amplifier.
* Programmable scrambler.
* Volume control.
Two interfaces are used between the CMX649 and the MSP430F1232. Control signals are sent to the CMX649 via its 'CBUS' serial interface, while voice data is transferred between the microcontroller and codec over its 'burst mode interface'.
The MSP430F1232 microcontroller was selected for this project because of its low power consumption and rich feature set. It provides 8 KB of Flash and 256 KB of RAM. The entire RAM is used for variables and stack pointer space, and less than 4 KB of Flash are used in this project.
The MSP430F1232 is placed into its active mode during voice communications and performs all required system management functions. The microcontroller operates from a 3,3 V power supply. The 8 MHz crystal sources the microcontroller's 'basic clock module' during normal operation, and the digitally controlled oscillator provides the timing signal during sleep mode operation.
Two USARTs were required for this design, one for the codec-to-microcontroller communications and one for transceiver-to-microcontroller communications. The MSP430F1232 with its one SPI port was selected over other family members because of its cost savings. The microcontroller's single SPI port services the communications between the transceiver and the microcontroller. A second SPI port, needed for the communications between the voice codec and the microcontroller, is emulated with a 'bit-banging' routine.
The watchdog timer is disabled during normal operations. When the design is placed in sleep mode, the watchdog timer occasionally causes the circuitry to power up and check for audio signals.
The Micrel MICRF505 'zero-IF' single-chip transceiver was selected for this design because of its low external component cost, device flexibility, and its ability to block co-channel interference. It operates from a dedicated 2,5 V voltage regulator to optimise noise performance.
The transmit data is applied directly to the MICRF505's VCO in this design. The inherent high-pass filter characteristic of PLL-based synthesisers means that direct VCO modulation of a data signal can result in undesired attenuation in the signal pass-band [4]. To prevent this situation, the transmit data is 3B4B-encoded in the microcontroller to ensure that the bandwidth of the data signal is high enough to be passed by the synthesiser. Encoding increases the transmit data rate by one-third, resulting in the 100 Kbps rate of data transfer both in and out of the MICRF505. The transceiver uses an internally generated data clock signal to control the exchange of data with the microcontroller.
A 200 mAH lithium polymer rechargeable battery from Ultralife Batteries was selected for this design because of its small form-factor and large energy density. The design can also easily accommodate other types of batteries.
The flexibility of the CMX649 and the MSP430 allow operation with very low power consumption. While the exact current consumption is a function of many variables (eg, user's voice volume, how often the user speaks into the microphone, etc), the observed current consumption during normal operation is approximately 19,25 mA, which translates to an estimated 'talk time' of 10,4 hours with the 200 mAh battery. The observed current consumption for sleep mode is 90 μA, and this translates to a standby time of approximately 92 days. (Note: these values are based on 100% battery capacity being available for use, but the low discharge rate of this design makes this assumption reasonably accurate.)
Low-dropout regulators provide 2,5 V to the RF transceiver and 3,3 V to both the microcontroller and voice codec. The speaker driver is connected directly to the battery to minimise current-surge noise from propagating to the rest of the circuitry.
Conclusion
Wireless digital voice transmission is very popular in today's consumer electronics. Items such as cordless telephones, wireless headsets and baby monitors are just a few of the items that use digital techniques to wirelessly communicate voice information. ADM voice coding should be considered for these applications because of its bit error robustness and low implementation cost.
Sound files that demonstrate ADM voice quality for various sampling rates can be heard at www.cmlmicro.com/products/applications/649/649sound/649sound.htm
References
[1] R. Steele, Delta modulation systems, Pentech Press, London, England, 1975.
[2] CML Microcircuits Application Note, 'Continuously variable slope delta modulation (CVSD): A tutorial', www.cmlmicro.com.
[3] N.S. Jayant and P. Noll, Digital coding of waveforms; principles and applications to speech and video, Prentice-Hall, Englewood Cliffs, N.J., 1984.
[4] CML Microcircuits Application Note, 'Using two-point modulation to reduce synthesiser problems when designing DC-coupled GMSK modulators', www.cmlmicro.com .
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