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Signal chain basics: Audio metering issues

Posted: 08 Jan 2013 ?? ?Print Version ?Bookmark and Share

Keywords:Audio metering? volume unit? peak program meters?

Audio metering takes place in almost any device that plays or records audiofrom cell phones displaying bar graphs of audio output levels, to home stereos with flashing LEDS, to live broadcasting.

There are multiple standards for audio metering that deal with the levels being monitored and time constants used for calculating the average. The behaviour of volume unit (VU) meters is defined in ANSI C16.5-1942, British Standard BS 6840, and IEC 60268-17. These standards define 0 VU = +4d Bm, with an integration time of 300 ms. Traditionally, with a moving needle meter, this means that from zero signal to 0 VU, the needle should reach 0 VU within 300 ms (same for decay). This makes a traditional VU meter excellent for getting an idea of loudness, but a poor method for monitoring transients. This was fine in traditional broadcast and recording systems, mainly based on tubes and tape, both of which tend to saturate elegantly.

In today's digital systems, clipping inputs or outputs sounds awful, so peak program meters (PPM) are used. PPMs are similar to VUs, but, integration time is much faster (typically around 10 ms). Note that there are multiple standards bodies with definitions for the signal levels and exact integration times. Display technology is a big part of these standards. Getting the ballistics right to move a needle through 90 degrees (or more) with one per cent of accuracy is much harder and slower than lighting a few LEDs!

Using LEDs, the quality of audio monitoring varies drastically. For example, a simple voice recorder product requires a different metering system from a mini-micro system using LEDs as a visual effect. In a live sound or broadcast environment accuracy is important, as well as seeing the difference between average loudness and the signal's absolute peak.

Most meters can be split into two sections: a front-end and comparator circuit (figure 1).

Figure 1: Two-part block diagram with a front-end and a comparator set up.

Signal conditioning
Signal conditioning happens in the front-end. Three things should be done at this point, plus an optional item:
1. Audio level gain/attenuation to configure what voltage should equate to highest LED/meter value
2. Integration time (for example, 300 ms for VU, 3. Signal rectification
4. Optional: peak-hold vs direct signal circuit

Signal rectification is required mainly in comparator-style systems. Full signal rectification is required because not all signals are symmetric. The acoustic output of some musical instruments (and some voice) shows more pressure on the microphone in one direction. To monitor this accurately, the signal should be fully rectified. Be sure to compensate for the usual 0.7 V diode drop.

Figure 2 shows a full wave rectifier circuit. Such a circuit can be implemented in a very small space, using BAV99 dual diodes in a SOT23 packages, and 100 kOhm resistor networks. For small voltage systems (3.3V), a quad low-voltage rail-to-rail output operational amplifier (op amp) such as the LMV324 can be used in a single supply system, but ground should be replaced by Vcc/2. The dual output op amp used in this example is the LF353, because of its low cost, high impedance input, and wide voltage range support (15V).

Figure 2: Full wave rectifier circuit.

Figure 3 is an example schematic of a signal rectification and VU integrator.

Figure 3: Full wave rectifier with VU integrator.

For systems that require monitoring peak versus direct signal (with its integration time), a peak-hold circuit (or code) can be used. In software this can be implemented by storing the ADC's largest peak value in a separate register and holding it for a specific number of samples. Whenever a new larger sample is received, simply overwrite the peak value and reset the counter. When the counter finally hits zero, reset the peak value to zero.

Figure 4: Example peak-hold circuit.

In an analogue circuit, the rectified output can be run through the circuit in figure 4. D1 biases the output up a diode drop (~0.7V), and is then dropped 0.7V by D2. D2 ensures that audio content can only flow into C1 and R1, not back through the feedback circuit. The values of C1 and R1 set the decay time for the peak hold. The impedance of the comparator array ADC also acts as impedance to ground.

About the author
Dafydd Roche is an audio engineer at Texas Instruments.

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