|
Vinyl/Tape to CD Wave Corrector | |
|
This page shows previous topics which have been published on our website.
Topic 1. New Digital Audio Formats. July 2001
Topic 2. Loudspeakers. August 2001
Topic 3. Negative Feedback. September 2001
Topic 4. Pickup Cartridges. October 2001
Topic 5. Record Playback Equalisation. November 2001
Topic 6. Digital to Analogue Conversion. December 2001
CD has now been around for a good few years and is being challenged by the new SACD and DVD audio, both of these new formats claim significant improvements in audio quality, extended playing times and true surround sound. Their appearance in the market place has confused a significant number of the buying public.
Both new formats suffer from a serious lack of available repertoire and this is not likely to improve in the next year or so, if the record companies restrict their new format output to genuine 24 bit original masters then it will take a very long time to equal the existing repertoire.
DVD audio is mainly backed by the US. and is an extension to DVD video, it uses a shorter wavelength laser than CD. At present the discs can only be played on dedicated DVD audio players, first reviews of these players suggest that replay quality is not as good as a medium priced CD player. Additionally the DVD players usually require a tedious navigation through several menus to achieve the correct playback mode.ie. stereo or mulichannel or with pictures, this means that you cannot simply load a disc into the machine and press play as you can with a CD player.
It is possible to play CDs in most of the DVD players but it has been noted in a number of reviews that the overall quality seldom reaches the level equal to that of a medium priced CD player.
SACD is backed by Philips/Sony who co-developed the original CD standard, like DVD a red laser is used to allow for the extra data storage. although a SACD will only play on a dedicated machine it is planned to issue most discs as dual layer format allowing automatic playback (in 16 bit) on any standard player. When playedback on an SACD machine the user will be required to choose standard stereo or multichannel surround sound.
Reviews suggest that very good quality can be obtained from a good SACD machine, though the discs are very expensive to buy. It is expected that the price of all machines will fall significantly over the next year.
It would be a sensible conclusion to
stay with CD for the foreseeable future, although it is claimed
that new systems are much better because they offer 24 bit at a
higher sampling rate, this is very difficult to justify. After
all, the present CD 16 bit standard will provide a dynamic range
of over 90db. It is doubtful whether a microphone exists to
capture an original signal of that level without contributing
noise above that level.
Modern mid priced CD players are now capable of effective bit up sampling to extend the 16 bit code to 24 bit, and although this will not in any way increase the true overall information it will dramatically lower distortion and noise. It is difficult to imagine that SACD/DVD discs will in practice provide any practical improvement over the existing 16 bit players. It is also very unlikely that many discs will be manufactured to a genuine 24 bit standard.
It seems that the most sensible way forward for most people will be to exploit such programmes as wavecor to maximize the quality of their existing vinyl collections and transfer to CDR. When played back on a reasonable machine the quality will be very good and they will be listening to performances ( of tempo and dynamics) the like of which are not present in the repertoire of the major record labels today.
Dave McGhee
Ganymede Test & Measurement
email: dave@wavecor.co.uk
web: http://www.wavecor.co.uk
4th July 2001
The job of a loudspeaker is to change the varying electrical current at the amplifier output into a varying sound pressure level, without adding any coloration or distortion to the signal. Compared to a perfect input signal a typical speaker will introduce harmonic, frequency and phase distortion. In practice the latter two will dominate because it will be very difficult to provide phase/frequency accuracy over the range 20-20Khz. Most practical speakers still use the principle developed in the 1920s where a coil of wire in a magnetic field moves a diaphragm.
The choice of a good replacement loudspeaker is always a very difficult task compared to a similar exercise of replacing your amplifier or CD player; the difference in sound between of amplifiers over a range of prices is usually very small and mainly relates to greater power and less distortion. There is however a wide and obvious difference in the characteristic of sound from a range of similarly priced and sized loudspeakers, a visit to any good shop or demo. room will reveal loudspeaker sounds to suite all tastes.
Inspection of speakers in the mid price range will reveal cabinets made from several types of material (wood/plywood/MDF) and filled with similarly varying types of acoustic damping material. Then there are the drive units which will have the moving diaphragms made from metal/paper/plastic/fiber or a range more exotic materials each with their own sound characteristic. Next of course there will be the choice of baffle type, this may be in the Popular form of ducted port reflex or infinite baffle. All this means that the designer has the very difficult job to marry the different types of item and end up with a commercially valid loudspeaker which produces a good sound at a reasonable price.
It is misleading to talk about a loudspeaker having an accurate sound, i.e. if it is tested in an anechoic chamber and found to have an ideal response, it will sound significantly different in a normal listening room where the reflected sound from walls and ceilings will add to the main wave. The important point to remember when choosing a speaker is to take the room into account, most obviously large speakers do not work well in small rooms. A loudspeaker has synergy with a room when it produces a sound stage in keeping with the room, so in a relatively small room say 3 x 4 m. you select a small speaker with a good but limited base response, this will produce a balanced sound without the overbearing base response which would occur if a larger speaker were used. When the correct choice is made you will be able to sit and listen almost anywhere in the room and hear a balanced sound, a speaker which is to large will exit the room resonance's producing to much base in one area and far to little in another.
Whilst talking about small or bookshelf speakers is worth mentioning that they usually employ two drive units, a base/midrange operating up to around 3 kHz and a tweeter covering 3 to 20 kHz. This means that the single crossover point is bang in the middle of the operating range. However on larger more expensive speakers there are likely to be several driver crossovers. In loudspeaker design it is the maintenance of correct amplitude/phase response at crossover that determines the true quality.
You should take advantage of room characteristics to achieve the best balance of sound, if you choose a good quality small speaker its base response will be enhanced or degraded by moving it relative to the walls or corners. The ideal height for speakers is that which puts the speaker at the same level as the head when seated in the listening position.
In truth a good loudspeaker is worth its money, if you have such a loudspeaker you will find that you can listen for hours on end to your favourite music. It will have that foot tapping quality which involves you in the music without causing fatigue. This is a condition which is not easy to establish; it requires trial and error and the help of a good dealer.
Finally you should consider the loudspeaker connecting leads, these should be capable of carrying the current from the amplifier to the speaker. In practice 5-10amp. capacity will be very adequate. You should take care to ensure that all connections are well made and do not offer any resistance to the current flow. Put politely you should ignore all suggestions that very expensive cables offer any advantage. They do not; their main function is to separate you from your money.
Dave McGhee
Ganymede Test & Measurement
email: dave@wavecor.co.uk
web: http://www.wavecor.co.uk
1st August 2001
Several recent technical reviews have been published on audio amplifiers which claim to use zero negative feedback, the objective of this article is to provide a limited insight into the application of negative feedback and demonstrate that it is most unlikely that any transistor amplifier is a truly zero feedback device.
The principles behind negative feedback are these:
1) Negative feedback reduces distortion. By taking a small portion of the output and subtracting it from the input any non linearity in the amplifier can be reduced, this is so because the feedback signal will produce an anti phase distortion signal at the amplifier input which will cancel( or reduce) the distortion caused in the amplifier. In the same way an amplifier with a poor frequency response will be improved by the application of negative feedback, because when the output is high more feedback is applied and the overall gain is reduced making a flatter or more even response.
2) Negative feedback is also used to change impedance at both the application and derivation points.
Negative feedback was first applied to amplifiers in the early days of valves, its main purpose being to reduce distortion. Valve amplifiers suffered from relatively low levels of distortion and had a frequency response dictated mainly by the characteristic of the output transformer. Hence small amounts of feedback were used; somewhere between 6 to 12dBs would be typical. As a rule of thumb the distortion would be reduced by the amount of feedback applied. This was of great advantage to the designer as the overall performance could be improved at the cost of providing a small amount of additional voltage gain.
The introduction of transistors to audio amplifiers caused a major change in design. Whereas in the valve era it was common practice to use only two stages(three if a phase splitter was used) of amplification, many more stages were used in semiconductor amplifiers. The open loop gain of a typical modern amplifier could be several thousand with the feedback being used to set the overall gain. Now this is the problem: if the open loop gain is so high and the semiconductors introduce a small time delay, then as the signal goes through the amplifier and transient distortion occurs, this will be amplified to overload levels before the feedback can act to correct the error. This causes is small but perceptible distortion and occurs in most semiconductor amplifiers. In this case, the introduction of single loop negative feedback to an amplifier will aggravate an initially small problem. This is why low level high order harmonics which are derived from crossover distortion in most types of transistor amps.are not improved by large amounts of single loop voltage negative feedback.
The important thing to remember is that negative feedback can only reduce an error once it has occurred.
In practice all the problems mentioned above can be eliminated by the application of current derived negative feedback at each individual stage in the amplifier. This removes the overall feedback loop and its disadvantages but allows the full advantages of negative feedback to be realized. When this is done the designer/manufacturer will claim a zero feedback design. This is untrue its just that the application of feedback has been changed.
The good news is that transistor ampflifiers do sound much more natural when distributed current feedback is applied. Conversely it is these amplifiers which will probably produce the poorest measured performance.
Dave McGhee
Ganymede Test & Measurement
email: dave@wavecor.co.uk
web: http://www.wavecor.co.uk
1st September 2001
Modern pickups come in in two main types, moving magnet and moving coil. In theory whether you generate electricity by moving a magnet in a coil or a coil in a magnetic field the results will be exactly the same. Iin practice the two designs produce noticeably different results. The crystal cartridge was very popular in the sixties but is very rare now and is not covered in this article.
Moving coil cartridges are constructed by making a very small coil of extremely fine wire that is attached to the stylus via a cantilever. If the coil is very light it will have a very small mass and will be able to move quickly and have a very good transient response. The required magnetic field is produced by a fixed magnet. The down side of this design is that the electrical output will be very low at around 200µvolts. A typical amplifier will require 10 to 20 mvolts to produce a noise free output. The low output impedance means that a transformer can be used to increase the output to the required level. The advantage of this design is that all moving parts are very light and produce little moving mass at the stylus tip.
Some modern moving coil cartridges are designed to have a higher output, this is achieved by using more turns of thinner wire. This increases the moving mass at the stylus and will reduce the advantage of the original design.
Moving magnet cartridges use a very powerful magnet positioned at the end of the cantilever (where the coil was in the m/c ). This has to be as small as possible to minimize the mass at the stylus. A fixed coil of many turns is positioned close to the magnet and the result is a much higher output than was possible with the m/c design. The down side here is the compromise necessary in magnet weight to achieve a good level of output whilst keeping the moving mass as low as possible.
Many variations in design exist to overcome this problem, it is possible to use a fixed magnet which is positioned to induce a magnetic field in another piece of ferrous material which would have a smaller mass than the magnet. The B&O mcc (moving micro cross) design is a good example.
A very unusual design of a moving iron cartridge was produced by Decca in the early 1960's. This involved attaching the stylus directly to the moving iron. In practice, this meant that the coil had to be very close to the stylus and therefore the disc surface. This had the effect of producing a superb transient response because the cantilever had been eliminated. Unlike all other cartridge designs which used individual left and right channel coils the Decca cartridge was also notable for having separate sum and difference coils, their outputs being mixed in the cartridge to produce accurate stereo. This cartridge was one of the most respected ever made because it produced the most dynamic and lifelike sound available at the time.
The stylus is the most important part of the cartridge, they are made of industrial diamond and are available in various shapes such as conical, elliptical and proprietary eg. fine line.
Typical stylus diamensions are:-
Conical mono 40µ (0.001 in.)
Conical stereo 20µ (0.005 in.)
Eliptical 15 x 7µ (0.0006 x 0.0003 in.) or18 x 5µ to (0.0007 x 0.0002 in.)
78rpm 108µ (0.0027in.); 128µ (.0032in.) typical values
When used to play vinyl LP's, a good stylus will last for about a thousand playing hours, the best life being achieved if tracking distortion is kept low by using a playing weight close to the upper limit. Records should be cleaned before playing.
Early mono micro groove records had a deeper groove and hence the wider stylus. 78rpm records were made with several groove widths so the best compromise would be about 110µ. You should note that playing records with a different shaped stylus to the one normally used will often give improved noise and distortion results. This is because the stylus is using a less worn part of the groove wall.
All the early types were conical which present the same width from any angle. However, as the cutting stylus is chisel shaped (it is wider head on), the lateral cut will be slightly less wide when it cuts a high modulation groove. For this reason, it was thought that an elliptical shape would improve tracking by maintaining better overall contact with the groove. In practice the benefits are very small and offset by increased record wear and a tendency to collect dust from the record surface.
The best shapes are fine line. This means that it is shaped so that it has a longer contact with the groove wall greatly reducing wear and improving trackability.
Dave McGhee
Ganymede Test & Measurement
email: dave@wavecor.co.uk
web: http://www.wavecor.co.uk
1st October 2001
Equalisation is the process by which the amplitudes of different audio frequencies are varied to produce an overall flat (equalised) frequency response.
When the master record is made, the groove is cut by a magnetically operated cutter head. This moves the cutter rather like the cone of a loudspeaker is moved by the coil. The characteristics of a moving coil are such that signal amplitude is inversely proportional to frequency. In other words, the groove amplitude gets smaller and smaller as the frequency increases. Low frequencies thus make inefficient use of the groove space whilst high frequencies tend to get lost in the surface noise. To overcome these effects, the recording engineer attenuates the bass signals and boosts the treble signals before they are applied to the cutter. This evens out the range of signal amplitudes as cut into the vinyl and makes optimum use of the medium.
The playback equalisation applies the inverse response of that applied at the recording stage. Bass signals are now boosted and treble signals attenuated in order to restore the signal as originally recorded.
When micro groove records were introduced in the 1950’s, there were half a dozen or so different equalisation options. A few years later these were reduced to one by the RIAA (Recording Industry Association of America). Through tests, the RIAA decided on the best compromise combination of bass cut and treble lift. There are however some disadvantages to this strategy which are caused by having a large amount of bass boost in the playback chain. Mains hum was nearly always a problem especially with valve amplifiers and turntable rumble was made worse. However these were soluble problems and the overall improvement was very worthwhile.
Adoption of this standard meant that a 12 inch LP with a good dynamic range would play for about 18 minutes (this was a dramatic increase over the 3/4 minutes possible on a 78 rpm disc). In the 1960s ‘varigroove’ records were introduced and increased the playback time up to 32 minuets. Varigroove was a very simple idea; it varied the groove spacing to take account of the dynamic range of the music. This meant that the cutter head was told one revolution ahead of the forthcoming level, so if the level was low groove spacing could be significantly reduced and then quickly increased to accommodate music peaks. This gave the LP longer playing time and increased the peak playback level.
Another type of equalisation is applied to LP’s to take account of the falling high frequency output as the playback stylus moves toward the centre of the disc. An increasing level of high frequency lift is applied as the cutter head moves to the centre, the boost (usually at about 10kHz) rises to 4dB at the final groove. As you can see this is a minor change and is intended to give the LP a flat frequency response over its playing area.
In these days of digital recording the enormous dynamic range possible on a good LP is often forgotten, particularly when you consider that the CD has an overload margin of zero whilst the LP could manage any level that the cutting engineer thought practical.
Dave McGhee
Ganymede Test & Measurement
email: dave@wavecor.co.uk
web: http://www.wavecor.co.uk
1st November 2001
D to A conversion is a very interesting subject and its implementation can take many forms. Digitally encoded audio or pulse code modulation (PCM) provides a stream of binary numbers which represent the audio signal. No matter what the transmission path and its deficiencies may be, if the binary numbers can be recovered then the audio can be regenerated free of interference.
There are three main types of D to A conversion.
Multi-Bit decoders contain a cascade of current (or voltage) generators representing each of the bits in a sample. Thus in a 16-bit system, there are 16 generators each of which is switched on or off depending on the values of the respective 0’s and 1’s in the 16-bit word. The outputs of the 16 generators are summed to produce the analogue sample.
With 16-bit PCM, the device must be able to produce 65,536 discrete levels and as music is represented by an ac signal this may be thought of as plus and minus 32,768 levels. These numbers imply a very high degree of accuracy, if the device is to be work properly to the least significant bit. For example, a transition from 0111111111111111 to 1000000000000000 will require a resolution of 1/32,768 as the most significant binary generator takes over and all the others are reset to zero.
To put this problem in context, imagine making a voltage divider to represent the various levels. This is clearly impossible, as you would not be able to produce the necessary degree of component accuracy and even if you could it would drift off calibration very quickly.
The problem was neatly got round by Philips in there respected TDA 1541 by having an on board square wave generator and shift register, this concept allows the output to be dependent upon the accuracy of the square wave rather than the individual binary current sources. A full description of the technique is given in the Philips manual but put simply each binary weighted current source is switched within dividers (4 times per sample) such that an amplitude error results in equal and opposite errors at the output. In ideal conditions these errors are smoothed out by the low pass filter at the output. Hence the linearity is dependent upon the square wave timing and not the accuracy of individual current sources.
Bitstream. The previous description shows how difficult it is to accurately produce the levels needed for each binary number. Bitstream is a solution to this that uses only one bit or pulse. In this device, the output level is represented by a number of pulses. The least significant bit would have a single short pulse and the number of pulses would rise in accordance with the binary code. If this stream of pulses is integrated in a simple filter then the output is an exact copy of the audio, but without any errors due to inaccurate current sources. This is a very neat idea because it makes the output entirely dependent upon the size and amplitude of a single pulse. Even if the pulse parameters change with age the output is still linear over the full code range. There is the disadvantage that the timing of the pulses is important; any inconsistency in timing will result in distortion at the output.
Pulse Position. This uses exactly the same principle as bitstream except that the width of the pulse is varied in accordance with the binary code and the result smoothed by the output filter.
A very recent addition to the previously mentioned converter types is called "continuous calibration". This gets around the original problem by ensuring that each binary current bit generator is repeatedly calibrated against an on board reference. This guarantees good linearity and does not need the high clocking frequencies that bitstream requires.
Dave McGhee
Ganymede Test & Measurement
email: dave@wavecor.co.uk
web: http://www.wavecor.co.uk
1st December 2001
You may make copies of this page so long as you include the references to the author.
 |
Back to Wave Corrector Home Page |