Most of the sounds we hear are made up of many different frequencies all vibrating together at the same time. The energy in a wave depends on its amplitude and frequency. The higher the amplitude, the more energy. With sound, amplitude is related to loudness. Also the higher the frequency, the more energy. The amplitude part of this makes intuitive sense. The frequency part does too, but it is less obvious.
Consider a musical instrument playing a sound. Since energy depends on amplitude and frequency, if it puts equal energy into all the frequencies it emits, then the higher frequencies must have a smaller amplitude. Musical instruments don’t actually put equal energy at all the frequencies they emit, but this does hold true roughly or approximately. If you do a spectrum analysis, they are loudest at or near the fundamental (lowest) frequency and their amplitude drops with frequency. Typically, roughly around 6 dB per octave. That is, every doubling of the frequency roughly halves the amplitude.
For example, here is amplitude vs. frequency for a high quality orchestral recording:
This graph shows amplitude dropping as frequency increases. Since energy is based on amplitude and frequency, this means roughly constant energy across the spectrum (all frequencies).
This implies that low frequencies are responsible for most of the amplitude in a musical waveform. So, if you look at a typical musical waveform, it looks like a big slow bass wave with ripples on it. Those ripples are the higher frequencies which have lower amplitudes. Further below I have an example picture.
Audio devices are not perfectly linear. They are usually designed to have the best linearity for low level signals, and as the signal amplitude approaches the maximum extremes they can become less linear. This is generally true with analog devices like speakers and amplifiers, and to a lesser extent with digital devices like DACs.
For example, consider a test signal like 19 and 20 kHz played simultaneously. If you encode this signal at a high level just below clipping, it’s not uncommon for DACs to produce more distortion than they do for the same signal encoded at a lower level like -12 dB. I’ve seen much smaller level changes, like a 1 dB reduction in level giving a 30 dB reduction in distortion! The same can be true for amplifiers.
Furthermore, the lower the level of a sound, the fewer bits remain to encode it. 16-bit audio refers to a full scale signal. But a signal at -36 dB has only 10 bits to encode it because the 6 most significant bits are all zero. Because the high frequencies are usually at lower levels, they are encoded with fewer bits, which is lower resolution. The Redbook CD standard had a solution to this called pre-emphasis: boost the high frequencies before digital encoding, then cut them after decoding. This is no longer used because it reduces high frequency headroom and most recordings are made in 24 bit and are dithered when converted to 16-bit.
The Importance of Bass Response
One insight from the above facts is that bass response is more important than we might realize. At low frequencies (say 40 Hz), the lowest level of distortion that trained listeners can detect is around 5%. But at high frequencies (say, 2 kHz), that threshold can be as low as 0.5%.
So one could say who cares if an audio device isn’t perfectly linear? Because of the energy spectrum of music, the highest amplitudes that approach non-linearity are usually in the bass, and we’re 10 times less sensitive to distortion in the bass, so we won’t hear it.
But this view is incorrect. It is based on a faulty intuition. The musical signal is a not a bunch of frequencies propagating independently. It is a single wave with all those frequencies superimposed together. Thus, the high frequencies are riding as a ripple on the bass wave. If the bass wave has high amplitude approaching the non-linear regions of a device, it is carrying the lower amplitude along with it, forcing even those low amplitude signals into the non-linear region.
A picture’s worth 1,000 words so here’s what I’m talking about, a snippet from a musical waveform. The areas marked in red are the midrange & treble which is lower amplitude and normally would be centered around zero, but riding on top of the bass wave has forced them toward the extreme positive and negative ranges:
This reminds me of a practical example. Decades ago, I owned a pair of Polk Audio 10B speakers. They had two 6.5″ midrange drivers, a 1″ dome tweeter, and a 10″ tuned passive radiator. The midrange drivers produced the bass and midrange. As you turned up the volume playing music having significant bass, at some point you started hearing distortion in the midrange. This is the point where the bass energy is driving the 6.5″ driver excursion near its limits where its response goes non-linear. All the frequencies it produces are more or less equally affected by this distortion, but our hearing is more sensitive in the higher frequencies so that’s where we hear it first.
Obviously, if you turn down the volume, the distortion goes away. However, if you use EQ or a tone control to turn down the bass, the same thing happens – the distortion goes away. Here the midrange frequencies are just as loud as before, but they’re perfectly clear because the distortion was caused by the larger amplitude bass wave forcing the driver to non-linear excursion.
Other Applications: Headphones
The best quality dynamic headphones have < 1 % distortion through the midrange and treble, but distortion increases at low frequencies, typically reaching 5% or more by the time it reaches down to 20 Hz. The best planar magnetic headphones have < 1% distortion through the entire audible range, even down to 20 Hz and lower.
Most people think it doesn’t matter that dynamic headphones have higher bass distortion, because we can’t easily hear distortion in the bass. But remember that the mids and treble are just a ripple riding on the bass wave, and most headphones have a single full-range driver. If you listen at low levels, it doesn’t matter. But as you turn up the volume, their bass distortion will leak into the mids and treble and become audible.
Thus, low bass distortion is more important in a speaker or headphone, than it might at first seem.
Other Applications: amplifiers and DACs
Amplifiers and DACs have a similar issue, though to a lesser extent. This concept could apply here as well – especially when considering the dynamic range compression that is so often applied to music these days.
Consider a digital recording that is made with dynamic range compression and leveled too hot, so it has inter-sample overs or clipping. Sadly, this describes most modern music rock/pop recordings, though it’s less common in jazz and classical.
Most of the energy in the musical waveform is in the bass, so if you attenuate the bass you reduce the overall levels by almost the same amount. This will entirely fix inter-sample overs, though it can’t fix clipping. Remember the 19+20 kHz example above? With most music, attenuating the bass will fix that too, since the higher frequencies are usually riding on that bass wave.
In an amp, the idea would be to attenuate the bass before the gain-feedback loop, then boost it back to normal afterward. In a DAC, the idea would be to attenuate the bass digitally before converting to analog, then boosting the bass back to normal in the analog domain.
Doing this essentially makes a tradeoff: lower distortion in the midrange and treble, higher distortion in the bass. Psychoacoustically, this tradeoff should be a net win because it prevents big low frequencies from leaking distortion into the midrange and treble.