HMV Model 328A

Concept

This project is a compact self-contained entertainment unit.  Its primary role is a high quality music player.

Speakers positioning is not negotiable for serious listening.  The only thing that can be done is to make them as small as possible and supplement them with a common subwoofer.  The positioning of the subwoofer is not critical.  If you want video, you need a screen at the front of the entertainment area, at a convenient viewing height.

That leaves the remainder of the system.  Ideally, you would combine all of these things into a single unit.  In the past, this was probably a giant cabinet with your turntable, cassette deck, video player, amplifiers and so on.  Now we can replace all of this with something much smaller.  We don't need lots of separate players, except for the purpose of transferring stuff from old media.  We could also include the subwoofer in the same unit.

7.1 Speaker Layout
7.1 Layout.  For stereo, the angle should be between 30° - 45°, but most audiophiles lean towards 30°.

Design Goals

The system must be designed to be simple and convenient to use, with minimal controls.  The system should include:


Design Philosophy

I want the core system to last as long as possible.  With things like the cabinet and the audio electronics this should not be a problem.  I built an amplifier in 1968 which I still use.

One thing we can count on in the consumer market is that "standards" change!  If you buy a dedicated entertainment system, it will almost certainly become obsolete within a few years.  For example, the transition from analogue to digital TV broadcasts has taken place since the start of this project.  The HD DVD standard never happened.  Basing an entertainment system around a PC seems to be by far the best option at this time, allowing for the upgrading of individual components as long as possible into the future.

Although this project uses PIC microcontrollers, in general I do not believe in using specialised components, especially exotic audio chips, which have a nasty habit of becoming "obsolete" and unobtainable within a few years.  I would rather use commonly available components where possible, even if this means an increase in circuit complexity.  This concept worked well for MCI, whilst many newer so-called professional audio products became useless within a short space of time, due to the unavailability of specialised parts.


Loudness Control

The human ear tends to be less sensitive to low and high frequencies at low listening levels.  Traditional hi-fi amplifiers include a loudness switch to compensate for this effect.  In low-end systems, this is just a fixed low frequency boost.  In more expensive systems, the bass boost is higher at low volume settings, reducing to flat as the volume control is turned up.

However, very few systems have true loudness compensation.  For a start, the overall listening levels need to be calibrated for the loudness compensation to work correctly.  In the late 50's / early 60's, some manufacturers experimented with elaborate switched controls with hand-wired networks to perform this compensation.  This was expensive and with the advent of stereo, most people conveniently tried to forget about the idea.  Today, if anything, suitable rotary switches are even harder to obtain and the number of steps are limited.

The idea here is to construct a digitally controlled FET switching network using cheap and readily available CMOS chips (4051, 4052 and 4053).  By carefully choosing the operating conditions of the surrounding audio electronics, it is possible to construct a high quality fully contoured loudness control with low distortion.  A PIC microcontroller is ideal for controlling the CMOS switches.


Subwoofer Design Concept

Most medium priced commercial units I hear have a peak around 60Hz to accentuate the kick drum to make them appear to have a lot of bass, but fail to get to the lowest note of a bass guitar, let alone Low Frequency Effects (LFE).  For that matter, most recording studio monitors don't get there either!

A well designed fully sealed enclosure is ideal for a subwoofer and will produce a smooth response, but it needs to be very large in most cases.  Build a concrete enclosure the size of a refrigerator and I promise you will have bass!  Ported designs allow much smaller boxes to be built to do the job, but they have limitations.  Below the tuned frequency of the system, there is no back-loading for the speaker cone and the speaker can be easily overloaded or modulate the higher frequencies - the last thing you would want for a speaker that could be exposed to very low frequencies.

For studio monitor or domestic listening situations, one solution is to put the subwoofer into an undersized sealed box, which will have a predictable low frequency roll-off of 12dB/octave at a frequency which you can calculate, then use an electronic equaliser to compensate for this.  The resonance of a large speaker in an undersized box will be relatively high, producing a peak at some higher frequency.  However, this can also be compensated for with the electronic equaliser.

The down side of this is that you require an enormous amount of power for this to work.  For example, if you require 20dB of boost at 20Hz to make the system flat, that means 100 times the amount of power needed to produce the same level at (say) 1KHz.  The exact relationship between the power requirements for the main speaker system and the subwoofer depends on the relative sensitivity of the two systems.  In practice, the energy levels at very low frequencies in most recordings are relatively low, reducing the demands on the subwoofer.  As a rough guide, if you wanted to run your main speakers at around 30W, the subwoofer amplifier might need to be capable of handling 300W!


Some Thoughts About Amplifiers

 

How Much Power?

This is a difficult question to answer, due to the number of variables involved.  Sound levels are inversely proportional to the square of the distance between the speaker and listener.  Speaker efficiency (which in all cases is very low) varies enormously.  For example, I have a pair of B&W DM5 speakers which are so efficient they sound surprisingly loud on a 2W computer sound card amplifier!  These were designed to work with valve amplifiers, where speaker efficiency was a key priority.  However, most modern speakers trade off efficiency for other parameters such as cabinet size, since it is no longer a problem to build higher power amplifiers.  Speaker efficiency is usually specified as the sound level in dBA at a distance of 1M with an input power of 1W.  In practice the efficiency of different speakers varies by up to 10dB and this represents an enormous difference in power requirements to produce the same loudness level.  This makes talking about how powerful your system is in watts almost as meaningless as the price!

For PA applications, the amplifier and speaker power ratings are usually matched to minimise the risk of damage to the speakers.  For hi-fi and private studio applications, the sound quality is often improved by using a higher power amplifier.  This is for two reasons.  Firstly, the amplifier is running well within its limits and usually performs better.  Secondly, transients are reproduced better when the speaker is running close to its limits.  For example, if you combined an amplifier with a continuous rating of 30W RMS and a music power rating of 40W with a speaker rated at 30W, the maximum average level you could play the system at would be 40W less the peak to average ratio of the recording.  If the peak to average ratio was 10dB, that would mean the average power level could only be 4W (1/10th) if all the peaks in the recording are to be reproduced cleanly.  If you replace it with a 300W amp, you can now run the speaker up to an average level of 30W (8.75dB louder than before), with peaks going up to 300W.  Provided you do not exceed the continuous power rating of the speaker or its maximum cone excursion, a speaker system will usually sound cleaner with an over-sized amp.  Naturally, you have to be very careful not to fry your drivers!


Why do Amplifiers Sound Different?

If we represent a speaker load with a resistor, we can calculate the current that an output transistor has to handle to deliver a given power.

Real world speaker loads are not pure resistors but instead a complex network of resistors, inductors and capacitors as a result of the speaker cables, the crossover network, the speaker voice coil itself and so on.  But the story doesn't end there.  Speakers are also generators.  If you move the speaker cone, you will generate a voltage at the speaker terminals.  Therefore, mechanical properties such as the speaker cone mass, suspension stiffness and enclosure volume also form part of the complex speaker load.

Imagine a point in time when the speaker cone is rapidly hurtling forward whilst playing music.  There will be a voltage across the speaker terminals due to this movement.  What happens if the input waveform is suddenly reversed?  At this instant, the speaker load looks like a voltage source, reverse-connected across the amplifier output!  In a similar way, the electrical reactance of the complex speaker load can produce "out of phase" voltages across the amplifier output.  In a worst-case scenario, an output device in an amplifier has to deal with up to double the voltage across it for a given current and up to double the current for a given voltage when running into a complex speaker load as opposed to a theoretical resistor.  There are often diodes connected between the amplifier output and its power rails to prevent the voltage across the speaker load from exceeding the power rail voltages.

In addition, the impedance of a speaker load is not constant with frequency.  It is possible for the impedance of some speakers to drop well below their nominal impedance at some frequencies, placing further demands on the amplifier's capabilities.

These unexpectedly high current and voltage demands on the amplifier output devices can sometimes exceed their ratings.  An unprotected bipolar transistor output stage will usually be damaged in the time it takes for a quick-blow fuse to blow if the output is overloaded or shorted.  V-I limiting usually involves the addition of two transistors to limit the operating region of the output transistors to their rated Safe Operating Area (SOA).  V-I limiting often reduces the ability of the output stage to deliver the necessary short-term heavy currents to the load and sometimes amplifiers without V-I limiting sound better when operating close to their maximum power.  In domestic and small private studios (but not PA!) applications, you might be prepared to take the risk with an unprotected amplifier and just be careful.

Any situation where the amplifier is overloaded, clips or V-I limiting takes place results in at least one of the transistor stages going into saturation or being totally cut off, due to the feedback circuitry trying to compensate for the error.  It takes time for a transistor to recover when from these conditions and this can result in audible artifacts in the sound.  A well designed amplifier would clip and recover cleanly.

None of these effects would necessarily show up in the normal specifications of an amplifier (such as frequency response, THD and so on), yet these are significant factors that can result in audible differences between amplifier designs.


Amplifier Design

A FET output stage without V-I limiting tends to withstand short-term overloads better and a well designed stage will usually safely blow the fuses if shorted without destruction of the FETs.

FETs have a softer "knee" characteristic and therefore have lower inherent crossover distortion than bipolar transistor versions.  However, FET amplifiers require a relatively heavy bias current to operate in their linear region and usually run hot when idling.  This is an unnecessary waste of energy.  Although I have heard some very good sounding FET amplifiers, I am yet to be convinced that there is any inherent sonic advantage to using FETs.

The other thing that turns me off using FETs for this project is concern about the long-term availability of specific FET devices and the reduced likelihood of finding a suitable replacement in the future.

It seems you cannot go past a conventional bipolar differential input stage followed by a Class A voltage gain stage feeding a Class AB output stage.  However, for higher performance and higher power designs, the simplest form of this design starts to put demands on the transistor specifications.  It becomes tempting to throw in a current mirror to increase the gain of the differential pair, or a buffer to increase the drive power of the Class A stage, or an extra driver transistor to increase the current gain of the output stage.  But sometimes this can result in potential instability and bad behaviour under overload conditions.

Interestingly, some of the better high power bipolar amplifiers I have heard have either series output transistors or operate in bridged mode.  In both cases, this doubles their slew rate capabilities.  This suggests that the speed of the output transistors is an important consideration in larger amplifiers.