Some gathered info on Audio meters

Hi All,

All these new meters and acronyms for audio measurement! What’s it all about ? I gathered up what I hope is a technical explanation for all the new loudness measurement tools out there. Some was drawn from Wikipedia and some from an excellent section included in the DMG Trackmeter manual. It will either help you understand it or make your head cook like an egg …:wink:

The original VU meter is a passive electromechanical device, namely a 200 µA DC d’Arsonval movement ammeter fed from a full wave copper-oxide rectifier mounted within the meter case. The mass of the needle causes a relatively slow response, which in effect integrates the signal, with a rise time of 300 ms. 0 VU is equal to +4 [dBu], or 1.228 volts RMS across a 600 ohm load, or about 2.5 milliWatts. 0 VU is often referred to as “0 dB”. The meter was designed not to measure the signal, but to let users aim the signal level to a target level of 0 VU (sometimes labelled 100%), so it is not important that the device is non-linear and imprecise for low levels. In effect, the scale ranges from −20 VU to +3 VU, with −3 VU right in the middle. Purely electronic devices may emulate the response of the needle; they are VU-meters in as much as they respect the standard.

The VU-meter (intentionally) “slows” measurement, averaging out peaks and troughs of short duration, and reflects more the perceived loudness of the material than the more modern and initially more expensive PPM meters. For this reason many audio practitioners prefer it to its alternatives, though the meter indication does not reflect some of the key features of the signal, most notably its peak level, which in many cases, must not pass a defined limit.

A peak programme meter (PPM) is an instrument used in professional audio for indicating the level of an audio signal. There are many different kinds of PPM. They fall into broad categories:

  • True peak programme meter. This shows the peak level of the waveform no matter how brief its duration.

  • Quasi peak programme meter (QPPM). This only shows the true level of the peak if it exceeds a certain duration, typically a few milliseconds. On peaks of shorter duration, it will indicate less than the true peak level. The extent of the shortfall is determined by the ‘integration time’.

  • Sample peak programme meter (SPPM). This is a PPM for digital audio—which shows only peak sample values, not the true waveform peaks (which may fall between samples and be up to 3 dB higher in amplitude). It may have either a ‘true’ or a ‘quasi’ integration characteristic.

  • Over-sampling peak programme meter. This is a sample PPM in which the signal has first been over-sampled, typically by a factor of four, to alleviate the problem with a basic sample PPM.

In professional usage, where consistent level measurements are needed across an industry, audio level meters often comply with a detailed formal standard. This ensures that all meters that comply with the standard will give the same indication on a given audio signal. The principal standard for PPMs is IEC 60268-10. It describes two different quasi-PPM designs that have roots in meters originally developed in the 1930s for the AM radio broadcasting networks of Germany (Type I) and the United Kingdom (Type II). The term Peak Programme Meter usually refers to these IEC-specified types and similar designs. Though originally designed for monitoring analogue audio signals, these PPMs are now also used with digital audio.

Traditionally, audio metering has been based on these PPM and/or RMS (VU) meters, which respond to voltage, and do not provide a good perceptual match to human experience of loudness. The ITU set about working out a remedy to this.

The first stage towards this was to determine more precisely how to take an RMS measurement, and to compensate for the fact that different frequencies are experienced as having different loudness. The latter part was resolved with what’s known as the K-weighting filter, which includes a high-pass filter at roughly 60Hz, and a high shelf at about 1.5kHz with about 4dB of boost. This simple filter has proven remarkably effective in listening tests at compensating for the frequency-nonlinearity of the ear.

The RMS measurement was now to be performed on K-weighted audio. Signals were to be processed per-channel, and then squared and summed. The RMS mean is computed over some number of samples. When that number of samples equates to 0.4 seconds, we call that the Momentary statistic. It has the feel of a traditional PPM meter, but maps better to perceived loudness. When that number of samples corresponds to 3 seconds, we call that the Short-Term statistic. It has the feel of a traditional RMS meter. The benefits are increased correlation with human experience of loudness, and unambiguous, clear definition as to how they are computed.

The ITU (and the EBU and the ATSC and many others!) also had a requirement to be able to measure the loudness of an entire program with a single number. Historically this would literally have been a TV program or radio program. It has since expanded to film, and it is expected to transition to music very soon.

Obviously a peak or an RMS meter respond in real-time to audio information, so some further processing was required. The ITU came up with two solutions. One was to compute the RMS of the entire K-weighted (filtered) signal. This is the Ungated signal, as used by ATSC A/85, and is written down as a figure in LKFS (with the understanding that LKFS means dBFS for a signal that has been measured as ITU1770 ungated).

The second solution the ITU proposed was to record all Momentary readings for the duration of the program, at a minimum of ten per second. Any readings below -70dB are discarded, and the average of what remains is computed. Then subtract 10 from that figure, and discard any readings below that number, and recompute the average. This is known as the Integrated (gated) loudness, as used by EBU r128. It is written down as a figure in LUFS (with the understanding that LUFS means dBFS for a signal that has been measured as ITU1770 integrated).

The EBU wanted a further statistical measurement for a complete program, to measure the perceived dynamic range. This is the LRA statistic. It is measured by recording all Short-Term readings for the duration of a program and discarding any readings below -70dB. They are then compiled into a histogram, where the top 5% and bottom 10% are discarded. The difference between the highest and lowest values remaining is the LRA figure, and is a figure in dB (no units,since it’s a ratio).

Finally, True Peak is an estimation of the maximum instantaneous level of a digital signal after it has been played through a DAC. It is possible for a digital audio signal which does not clip to represent a signal that does clip in the analogue domain. This is measured by simply upsampling the audio and measuring peak levels.




Before anyone breaks into a sweat… :wink:
One of the beauties of our live systems is that all we need to worry about is balance and that we’re not clipping.
If it sounds right, it is right.
We really don’t even care too much about dynamic range because the noise floor of any venue is way above the floor of a 16 bit signal, let alone the 24 bit depth that most contemporary digital gear uses.
In situations where we might employ Cantabile as a mastering buss, the likelihood is that the plugins themselves will be equipped with comprehensive metering and dithering options, and is pretty much a foregone.

Useful info though, Dave. :+1:

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In the digital world I generally need to know just one thing- did it go over 0? At ll? In any way, shape or form? Because there be dragons :smiley: