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.
There may also be aspects of stereo recording that mean that such ‘ambience enhancement’ is something of a lottery.
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:
Vinyl
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.)
Valve Amplifier
- 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
- distortion
- 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
- beaming
- 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.
- lobing
- 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
- distortion
- 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 10/07/18]
The Rational Audiophile 
Your comments RE passive crossovers (in contrast to active) seem rather disingenuous. The factors that can negatively affect passive speakers can (and are) taken into account in well designed loudspeakers. Your choice of a 2 way design is of course the worst case scenario for a passive speaker. You seem to be simply selectively picking out the worst possible case to perform some sort of slam-dunk about the passive versus active speakers. I don’t necessarily disagree that active DSP / crossovers have great potential advantages but if the benefits are so obvious, why write a piece that makes gross generalisations and disregards the plethora of well designed subjectively and objectively good “shameful” passive speakers?
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Well, it’s my view of “the average audiophile system”. My ‘thesis’ is that a high proportion of passive speakers are two-way, even at the fairly expensive end, because of the difficulties in making three-way passive systems. In other words, the choice of the number of ways is not independent of the implementation of the crossovers. DSP crossovers, in my opinion, not only work better in isolation, but allow a three-way speaker to be designed as easily as a two-way.
I know that passive speakers can sound OK, but before we even listen to them we know that they are a compromise. If cost and complexity of the hardware were overriding factors, then it would be perfectly understandable at the low end. But once we get into the realms of many thousands of pounds, and cables raised on ceramic pots etc., it seems to me that we should be looking for something better. Another one of my ‘theses’ is that we don’t need empirical results to validate our ideas, and this is an example of such rational thinking. I may be on my own on this!
Irrespective of cost and complexity, can you think of any reasons why passive crossovers might actually perform better than active?
(Thanks for the comment.)
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Whiile I personally use all DSP and active crossovers, may I play Devil’s Advocate? A speaker designer I highly admire, Tom Danley, even though I’ve not heard (nor can I afford!) his speakers, makes this point (more or less): his taret market is the pro/arena user, and they want an easy hook-up. Thus most, if not all, of DSL speakers use passive crossovers. They are, I assume, very highly engineered to the specific drivers, since phase/coherence/etc. are hallmarks of Danley’s designs. So if a passive x-over is needed, it is certainly within reach of modern modelling and engineering.
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The issues with a basic reflex port are only problems when improperly designed. I believe you think that they are a compromise in order to extend frequency response because large woofers and large sealed boxes are impractical. However there are speaker with very large enclosures and large woofers that still use bass reflex ports. For these speakers I’m sure it wasn’t some trick and is an part of the design that was considered for maximum fidelity.
The Meyer Sound X-10 is a good example. It has a 15″ woofer, huge box and still has bass ports. It is a fully active speaker with DSP and all sorts of other things you advocate. Made by audio engineers for the pro-market, not the audiophile segment.
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Thanks very much for the comment.
Are you sure that the basic problems go away when properly designed? It seems to me that in order for them to ‘work’ they have to resonate, and it is that resonance that causes the smearing and rapid roll-off below resonance. But at the same time they undeniably help the power handling of the speaker, so I guess for the professional market that would make perfect sense..? With the couple of speakers I have built, I have had the luxury of not having to worry about playing them at professional levels. If I was making them commercially for professional use I would probably spend 90% of the design effort (no exaggeration) on making sure they can be pumped at 200W continuously in an ambient temperature of 40 degrees C or whatever. My instinct would be to do it with ‘intelligent’ DSP protection, but for the highest SPL levels I would have to go with bass reflex as well.
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There’s a reason why I picked the Meyer X-10. It is not for live sound reinforcement but for studio monitoring. They are obsessed enough about accuracy where they have a microphone mounted in front of the woofer for feedback to control the bass response. Just look at the impulse response in this link
Click to access x-10_ds.pdf
With this attention devoted to accurate sound reproduction I highly doubt that a bass reflex port would have been included in the design if it was ‘smearing’ the signal.
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So why do you think they use bass reflex? And why do other people use sealed?
In my mind I am clear why sealed is better for a pure hi-fi speaker. Quoting wikipedia:
“The low frequency driver in a resonant speaker enclosure system such as a ported cabinet or passive radiator cabinet cannot start and stop instantly like it can in a sealed-box cabinet. In order to achieve their bass output, ported speaker enclosures stagger two resonances. One from the driver and boxed air and another from the boxed air and port. This a more complex case than an equivalent sealed box. It causes increased time delay (increased group delay imposed by the twin resonances), both in the commencement of bass output and in its cessation. Therefore, a flat steady-state bass response does not occur at the same time as the rest of the sonic output. Instead, it starts later (lags) and accumulates over time as a longish resonant “tail”. Because of this complex, frequency-dependent loading, ported enclosures generally result in poorer transient response at low frequencies than in well-designed sealed box systems. “
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I didn’t design the speaker but I assume that the woofer has a large motor for precise control, especially if you are employing the feedback system that they are using. Sealed cabinets are typically employed because of their damping properties to reduce ringing. For cheap drivers with weak motors this makes sense, the driver would continue ringing without additional damping. However if you have a strong motor which allows accurate control, then you don’t need additional damping and in fact can lead to overdamping. Thus a vented design would be preferable in this case.
If you see the Linkwitz LX521 open baffle project, he also employs bass drivers with very low Qts and large magnets. These drivers are made custom for open baffles and accurate bass. Sometimes its not about the port response but what the driver itself needs to perform it’s best.
And really what matters in the end are measurements (and how it sounds). The time response plots for the X-10 are excellent regardless what enclosure they use. If you look at plots and THD measurements for both sealed and ported, they both perform similarly when done right.
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OMG CaptainSpirou you answered your own question without realizing it.
THERE IS NO SIGNAL NOR MOTOR CONTROL OF THE PORT. The live or recorded signal is increased through a power amplifier in order to create a force in a linear motor at NEARLY THE SPEED OF LIGHT, delayed slightly at low frequencies by the inherent inductance of the coil winding. The mass of the soft parts determine the upper frequency response of the acoustic system, and again makes a small delay effect at the low end. The mass also effects the natural resonance of the driver, and the settling time of the mechanical/acoustical system within an enclosure without influence from the power amplifier (open circuit settling time). With the amplifier terminals connected to the driver with low impedance wires, there is back EMF damping applied to that same motor system that rapidly decreases driver output following cessation of signal. In contrast, the port air has NO CONNECTION to the input signal. Much like a doctor tapping your knee with a hammer, the REFLEX RESPONSE IS DELAYED because the port is compliant at rest, and becomes “stiff” when resonating out of phase with the pressure of the front of the cone. Think about clapping your hands. Clapping at a hand moving away from you results in no sound. Clapping at a hand moving towards you makes a loud sound. The trouble is that ALL PORTED SPEAKERS HAVE INITIAL PORT BEHAVIOR THAT IS OUT OF PHASE WITH THE FRONT OF THE CONE. The front of the cone moving outward pushes the port air IN, and the back of the cone sucks the port air IN during the start of the transient. I am going to repeat this because you and 99% of humanity don’t seem to get this, at all. THE FIRST CYCLE OF A TRANSIENT AROUND THE PORT TUNING FREQUENCY IS CANCELLED TO NEAR ZERO ENERGY. It doesn’t get delayed. It is erased into the bottomless hole of destructive interference. Even if you put the speaker in a corner. Even if you stick your ear between the port and diaphragm. GONE. So for any instrument in the bass range with a transient envelope behavior (anything other than a pipe organ), the most energetic portion of its sound is snipped off and discarded. And what does the port air do if the signal suddenly ceases? Whatever the hell it wants (no damping from input signal, amplifier damping factor impedance ratio, nor servo feedback network). Kiss involvement with the music goodbye. Kiss fidelity goodbye. Kiss maximum suspension of disbelief you are not present with the artist goodbye.
The reason for this permanent mass delusion in loudspeaker design is two-fold. One part is irrational, the other part is entirely rational. Once upon a time in 1950 it was very difficult to get more than 7W out of a tube amp. Loudspeakers had to be power efficient so everything was horn loaded, with short xmax simple motors using weak magnets, and ported to squeeze every last dB out of a Watt that was possible. A good frequency response was considered 100-10,000 CPS (Cycles Per Second, as opposed to Hertz, which means the same). We don’t have the same power limitations in the 21st century, when you can get a 1000W Class D amplifier built from $100 in parts off of eBay. Following an outdated set of design considerations as a result of blind tradition is foolish and shortsighted. The second reason almost all designers port EVERYTHING is expediency, otherwise known as cheapness. If you want a LOUDER SEALED CABINET, as a single physical unit with a certain dB output for a given cabinet volume, all you have to do is ADD A SECOND BASS DRIVER, and maybe a larger or second amplifier channel (In some cases you get more amplifier output power simply by halving the load impedance). But this 2nd driver costs extra money. So they cut a hole in the box, slap in a $2 cardboard tube (or a really fancy flared plastic tube for $4) and make more money, while you miss the most important part of the low range of music; accurate dynamics in time with the rest of the pass band.
No exceptions. Not because the loudspeaker designer has experience designing ported speakers. Not because they went to college. Not because they used computer simulation. Not because they used a complex and accurate measurement system. Not because they used a better driver with a better motor system design or a bigger magnet. Not because they spent more money making a stiffer 11 birch ply box with better hardware. Not because they applied DSP or servo feedback in the amplifier. Not because they charge more money. All ported loudspeakers behavior is the same; they fail to track the input signal accurately on transients near their tuning frequency, and have rapidly falling output below this frequency (-24dB/oct) that room gain cannot compensate for (+12dB/oct) and where it is dangerous for cone motion to apply an EQ boost.
Period.
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I’m pretty sure the “mixing stereo bass to mono” on vinyl thing is a myth. Plenty of old stereo mixes feature hard panned bass. Having bass in the center is just a matter of good stereo imaging, especially on headphones.
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I’m not so sure. Check out my earlier post “Vinyl Sales Overtake Digital”. I quote a real vinyl mastering web site (the first one I looked at, actually) and one of its tips is to mix bass to the centre, and avoid large phase differences. I agree that some old 60s stuff had the bass at one side only (a particular Small Faces track springs to mind). Not sure whether that is technically ‘mono’ though..? I think the biggest problem is when left and right are in anti-phase.
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