Here are my thoughts on what is wrong with the average audiophile system, based purely on objective measurements and the idea that the best system should be the most accurate in reproducing the recorded waveform. Please feel free to disagree. I would be very interested if anyone could put the other case: that a hi-fi system shouldn’t just be linear, but should contribute something of its own to the signal. Why is this, and what should that something be?
Is Stereo sufficent?
There is an argument to be had over whether hi-fi’s primary limitation is its reliance on plain two-channel stereo versus alternatives that seek to reduce unwanted crossfeed such as ambiophonics and BACCH, or multi-channel surround sound, etc. For the foreseeable future, however, it looks as though plain stereo will remain dominant.
The distortions of vinyl, valves and passive crossovers
Assuming the use of plain stereo, the job of the hi-fi system would appear to be that of ensuring that the two channels that the record producers have created for us make it into the room as close to intact as possible. When this is done properly, to my ears it can sound pretty spectacular.
However, at every stage in the recording and replay process, distortions are added to the signal. It would seem fairly straightforward that we would like to minimise those distortions. But while audiophiles and the hi-fi industry may use measurements and specifications that fit with this idea, they also believe that there are aspects of the sound that are beyond measurements.
If we are interested only in reproducing the waveform accurately, here is a list of the distortions that typical audiophile hardware imposes on the signal:
Noise and distortion
- basic noise floor of -70dB (A-weighted) if we’re lucky – stylus scraping along a groove in plastic, vinyl has finite grain size
- pops and clicks: scratches and dust
- electrical hum and noise: cartridge produces a tiny signal, and high gain pre-amplification is needed
- rumble: bearings, motor
- warped records cause various problems
- stylus wear
- stylus contamination: dust, dirt, vinyl particles
- stylus misalignment – may vary as arm moves across record
- record wear – the Decca test disc for measuring cartridge frequency response was specified for only 5 plays for the tracks with frequencies above 10 kHz!
- record contamination: dust, dirt, vinyl particles
- fundamental limitations in linearity of vinyl cutting/replay system
- diameter loss: speed of groove decreases throughout LP, increasing noise and distortion and reducing upper frequency response
- pre-echo: adjacent groove modulation
- microphony: sound from speakers feeds back into the pickup
- Channel separation: varies with frequency and typically only 20-30 dB at maximum
- Record may be pressed towards end of life of the stamper, resulting in increased levels of various distortions
Arbitrary processing needed for vinyl mastering
- compression (raises the quietest sections in volume to make them audible above the background noise, reduces the loudest sections to economise on groove spacing)
- de-essing (reduce treble response for high amplitude, high frequency sounds)
- mixing stereo bass to mono (otherwise the needle jumps out of the groove)
Wow and flutter
- off-centre pressing
- motor speed, belt, etc.
Inaccurate frequency response
- RIAA record and/or playback curves are often only approximate
- cumulative effect of factors above causes imprecise frequency response (arbitrary processing when mastering, diameter loss, etc.)
- With fashionable ‘retro’ topologies THD can be of the order 1%-10%
- some people think that harmonic distortion sounds nice if it is of the right type, but harmonic distortion automatically creates objectionable intermodulation distortion (IMD) on all but the simplest signals. You think the amp sounds nice on girl-and-guitar, but it will sound atrocious on a full symphony orchestra.
- limited output power
- transformer coupling at output (some valve amplifiers claim to be output transformerless ‘OTL’, but this may be a case of throwing the baby out with the bathwater)
- distortion and limited frequency response
- high output impedance
- lack of cone damping
- inaccurate frequency response
- microphony: all valves are microphonic to some extent and so sound from speakers can feed back into the amp
- constantly degrading performance as valves age
- indistinct stereo imaging due to discrepancies between channels suffering from differing intermodulation distortion with different composite signals.
Two-way passive speakers
There are a lot of ‘anomalies’ in the reproduction of a signal using a two-way passive speaker:
- Lack of bass due to small bass/mid driver, compact enclosure – a major, unnatural discrepancy between the original signal and what emerges into the room.
- Lack of damping
- Passive crossover adds impedance between amp and driver: speaker cones are not under precise control of the amplifier
- Series impedance of crossover increases distortion from nonlinear driver load
- passive crossover is extra, awkward load for amplifier
- wastes amplifier power
- amp has to work harder
- higher distortion
- inaccurate crossover
- varies with power and temperature – system changes character with output volume
- EQ control is ‘blunt’
- EQ non-adjustable for room/speaker combination and placement
- phase response is not flat
- colours the sound of transients independently of apparently-flat frequency response
- smears detail
- intermodulation distortion and doppler distortion
- mid frequencies ‘ride’ on top of large bass woofer displacements
- mid frequency output power limited by driver’s ability to handle bass
- woofer doubles up as mid range: large diameter cone becomes directional at top end, mismatched with tweeter that has wide dispersion at crossover frequency
- indirect sound has different frequency response compared to direct.
- vertical separation of adjacent drivers causes frequency-selective cancellation pattern. This becomes significant in a two way speaker compared to a three-way.
- iron-cored inductor (if used) saturation
- impairment of filtering ability at high power
- breakup, power handling
- woofer and tweeter reproducing frequencies outside their comfort zones -> distortion
- bass reflex
- introduces more time domain smearing
- port produces distortion and “chuffing” noise even if woofer cone is displaced less and therefore supposedly less distorted
- port efficiency decreases with output power – sound becomes harsher as volume increases
- inverted, delayed signal mixed with direct signal
- may measure fine with steady state sine waves but causes distortion of transients and music waveforms
- midrange bleed from interior of box out of the port
- unnaturally-rapid roll-off
- cannot take advantage of room gain
- effectively a deep hole in the response compared to what was recorded
- uncontrolled cone below resonance
- woofer still has to produce mid frequencies as cone flaps about
- many speakers do not have ‘time-aligned’ drivers i.e. woofer and tweeter’s outputs do not reach the listener’s ears at the same moment.
- in a non-DSP speaker this can only be achieved by setting the tweeter back from the plane of the woofer’s front edge.
- displacement must be varied for different vertical listening positions
- cumulative effect of all of these factors on stereo ‘imaging’.
- Any errors, even if supposedly duplicated in both channels must have an effect on imaging because the two channels are not reproducing identical signals. For example, Doppler distortion affects everything in the mid range, but will be worse in one channel than the other if that channel happens to be playing more bass at that moment (= a form of intermodulation distortion). Imaging becomes ‘blurry’ and the brain has to do more work to try to resolve what it is hearing.
Clearly the ‘high end’ hi-fi system that most audiophiles aspire to own is anything but objectively transparent.
[Last edited 17/03/17]