脑洞大开，TDA1547另外一种用法

xinghuaman · 发表于 2016-11-27 20:22

Clocking and jitter

Good clock stability is probably the single most important issue separating good-quality analogue interfaces from the rest.  With the linearity of modern A/D and D/A converter chips beginning to rival and exceed the performance of the best analogue circuits, digital recordings would already be  ‘beyond reproach’ if clock stability did not so often degrade their potential quality.

Why is good clock stability so rare?  Probably because most conversion equipment has to compromise between clock stability, operational requirements and cost.  The ideal clock system in an A/D or D/A converter would be ultimately stable, i.e. would exhibit no jitter (frequency variations) at the point of conversion, whether operating from an internal clock or from an external synchronization reference of any format and at any sample rate.  But this is a very tall order for circuit designers, especially if they are on a budget.

Why are good clocks so rare?

Most analogue interfaces can provide workmanlike performance when internally clocked, since this is only a matter of providing a stable clock oscillator (or range of oscillators) at a fixed frequency (or frequencies) – although even this is not always well-executed.  The real problem is that in many installations the analogue interfaces can almost never operate from their own internal clocks since they must be slaved to an external reference sync, or maybe to a clock from a host computer.

The externally-clocked design challenge has traditionally been a trade-off. since the more stable a clock oscillator is, the less is its range of frequency adjustment: but we would ideally like an oscillator which can operate over a wide range of sample rates, perhaps from <44.1kHz to >48kHz, plus multiples thereof.  But such an oscillator would inevitably have poor stability – at least in terms of the stringent requirements for high-quality audio conversion.  On the other hand, if we limit the range of rates at which the oscillator needs to operate to small ‘islands’ around the standard sample rates we could use a bank of oscillators, selecting the appropriate oscillator according to our desired sample rate.  But this is expensive and, in any case, the 'pull-range' of an ordinary quartz crystal oscillator is still generally insufficient to meet the tolerance demands of the digital audio interfacing standards.

As well as a very stable clock oscillator, a good sounding converter must have a PLL (phase-locked loop) with a loop-filter which steeply attenuates incoming reference jitter towards higher frequencies. Unfortunately, even if sourcing equipment provides a reference clock with low jitter, cabling always adds unacceptable amounts, especially poor quality or high-capacitance cable, which results directly in sampling jitter in the analogue interface if jitter-filtering is inadequate.

Prism Sound's unique CleverClox technology breaks these traditional constraints, allowing a low jitter clock to be re-created from any reference sync, no matter how much jitter it has and no matter what its frequency.

But why is clock jitter so important?

Analysis of sampling jitter

Analysis of sampling jitter (small variations in the sampling intervals of an A/D or D/A converter) shows that it produces a similar effect to phase modulation, where distortion components appear as  ‘sidebands’ spaced away from the frequency of a converted tone by the frequency of the jitter itself.  These components get louder as the amount of jitter increases, but also as the frequency of the converted tone increases. So sampling jitter produces distortions which should sound much worse than conventional analogue harmonic distortions, since the spurious components appear at aharmonic frequencies.  High audio frequencies should suffer worse distortion than low frequencies.  For low-frequency jitter, the resulting distortion sidebands appear close in frequency to the audio signals which produce them – this should mean that they are ‘masked’ from our hearing by the same psycho-acoustic phenomenon upon which are based sub-band (perceptual) coding schemes such as MPEG.  This is fortunate, since it is quite difficult for a PLL to remove jitter to a good degree even at moderate frequencies, but for very low frequencies it would be very difficult indeed.

The graph below shows the effects of 'JTEST', a special test stimulus to expose jitter susceptibility of D/A converters.  JTEST is basically an fs/4 tone (12kHz at fs=48kHz) which is specially coded to cause an AES3 or S/PDIF carrier transmitted over a lossy cable to become very jittery by the time it reaches the receiving D/A converter.  The jitter produced has regular frequency components fs/96 apart (500Hz at fs=48kHz).  The quality of the D/A converter's jitter rejection is shown by the degree to which it suppresses the resulting 500Hz-spaced side-tones.  In the example below, the upper trace shows the poor jitter rejection of 'conventional' D/A converter design, where the conversion clock is derived directly from the AES3 or S/PDIF receiving chip, without any further jitter filtering.  Remember that none of these side-tones is present in the digital audio signal - they are caused only by jitter.  The lower trace shows almost complete jitter rejection across the band by the CleverClox process in Orpheus.

Listening experience

In practice, it seems that the benefits of careful clock design are very apparent in listening tests.  On the other hand, it can sometimes be difficult to expose the shortcomings of converters with poor clocks, because these units often have other analogue problems whose severity might obscure jitter-related effects.

In general, some of the widely-noted effects of sampling jitter are not surprising – for example the muddying of brass, strings and high-frequency percussion and the loss of stereo (or multi-channel) imaging.  These are well explained by the worse distortions which result in the lab at loud, high frequencies, and the way that sampling jitter produces quiet, aharmonic components, perhaps only subliminally perceptible, which blur our impression of the ambience which creates a soundstage.

Other effects are harder to explain – for example there is wide observation that large amounts of sampling jitter can take the edge off extreme bass rendition.  Such reports are probably too widespread to be ignored, but defy explanation within current theory.

Orpheus and CleverClox

Orpheus is designed to source clocks which are as stable and accurate as possible, and also with the aim of being insensitive to the quality of incoming clocks.  It is designed to remove jitter from any selected reference sync source before it is used as a conversion timebase, so as to eliminate any audible effects of sampling jitter, whatever sync source is used.

Orpheus does this with the help of Prism Sound's unique CleverClox clock technology, which removes the jitter from any selected clock source down to sub-sonic frequencies, without the need for a narrow-band quartz VCO.  CleverClox can adapt to any reference, irrespective of frequency, and regardless of how much jitter it has, derives an ultra-stable conversion timebase.

xinghuaman · 发表于 2016-11-27 20:24

本帖最后由 xinghuaman 于 2016-11-27 20:25 编辑

Dither and noise-shaping

Orpheus can dither or noise-shape its digital output to produce high-quality 16 bit output (for, say, a CD master) from 20 bit or 24 bit recordings.  This section discusses the principles and choices involved in word-length reduction.

Truncation and dithering

There are many points in a digital audio signal path where precision can be lost.  For example, in a digital transfer from 24-bits to 16-bits, or in an analogue to digital conversion.  In this situation it is not sufficient just to discard low-order bits – this causes truncation distortion, characterised by aharmonic frequency components and unnatural, harsh decays.

Instead, it is preferable to use some sort of ‘dithering’ process, whereby the truncation process is linearized by modulating the signal prior to the truncation, usually by the addition of a small amount of noise.  By adding a random element to the truncation decision, small components as far as 30dB below the noise floor can be accurately represented, and an analogue-like low-signal performance can be realised.  This is achieved at the expense of slightly raising of the noise floor, although with some dithering schemes such as noise-shaping, linearization can be achieved with no noticeable increase in noise.

How can dithering allow information to be preserved below the least-significant bit?  It seems impossible.  Consider a simple example where the audio samples are numbers between one and six, and we are going to ‘truncate’ them (i.e. reduce their resolution) so that numbers from one to three become zero, and those from four to six become one.  Clearly much information will be lost, and all excursions of the signal between one and three and between four and six will not affect the output at all.  But if we throw a die for each sample, add the number of spots to that sample, and translate totals of six and below to zero and totals of seven and above to one, we have a simple dithering scheme.  Input samples of three will be more likely to result in outputs of one than will inputs of one.  The throw of the die is our dither noise.  Since all the faces of the die have an equal chance of occurring, this is known as ‘rectangular probability distribution function’ (RPDF) dither, which in fact does not produce perfect linearization.  We actually use ‘triangular probability distribution function’ (TPDF) dither, which is like throwing two dice with a resultant increase in the probability of medium sized numbers – totals of two and twelve occur much less often than seven.

Noise shaping

It is possible to reduce the subjective effect of the added dither noise by either using spectrally weighted ('‘blue'’) dither noise, which is quieter in the more sensitive registers of the ear, or by an even more effective technique called ‘noise shaping’.

Noise shaping is just like conventional dithering, except that the error signal generated when the unwanted low-order bits are discarded is filtered and subtracted from the input signal.  You can’t get something for nothing – the error cannot be simply cancelled out, because we already know that the output hasn’t got enough bits to precisely represent the input.  But by choosing an appropriate shape for the error filter, we can force the dither noise / error signal to adopt the desired shape in the frequency domain – we usually choose a shape which tracks the low-field perception threshold of the human ear against frequency.  As can be seen from the plots below, this has the effect of actually lowering the noise floor in the more sensitive frequency bands when compared to the flat dither case.

The theory of noise shaping has been around for a long time – certainly since well before DSP in real-time was feasible for audio signals.  It has applications in many signal processing and data conversion applications outside audio.  It has been well researched, and is not in the least bit mysterious.  ‘Proprietary’ word-length reduction algorithms are generally conventional noise shapers.  Assuming that the basic implementation and dither levels are correct, the only significant freedoms available to the designer are to choose the actual shape of the noise floor, and to decide how to adapt this (if at all) to different sample rates.

Prism Sound SNS (Super Noise Shaping)

Orpheus provides a comprehensive choice of dithering and noise-shaping processes.  These comprise ‘flat’ dithering, plus a selection of four Prism Sound ‘SNS’ (‘Super Noise Shaping’) algorithms.  All produce high-quality 16 bit output: the choice of which one to use is purely subjective.  The four SNS algorithms are designated SNS1 to SNS4, in increasing order of the degree of shaping.  The spectra of the four SNS algorithms are shown below.  Note that, unlike some noise shaping algorithms, SNS spectra are adjusted automatically to provide optimum subjective advantage at each different sample rate.  The spectra are shown below for 16-bit output, at 44.1kHz, 48kHz and 96kHz sample rates.

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It is difficult to assess the difference in sound between different noise shapers for any given program material, since their effects are at very low amplitudes (the 0dB line on the plots below represents flat dither with an rms noise amplitude of about –93.4dBFS). It is tempting to audition noise shapers by using a low signal level and boosting the shaper output by tens of dBs in the digital domain prior to monitoring. Using this method it is easy to hear that the noise floor of more extreme shapers is clearly not white – switching, say, from SNS1 to SNS4 sounds like shhhhh..ssssss as the noise is shifted towards the higher frequencies. However, this is not really a meaningful test since the sensitivity of the ear at different frequencies is very dependent on level, and the design of the more extreme shapers is in any case intended to render the noise floor completely inaudible at normal listening levels. Ultimately, the only ‘right’ choice of noise shaper is the one which sounds best for the material. SNS2 is a good starting point for most situations.

The Prism Sound SNS logo shown above is found on many of the world’s finest CDs, and is recognised as a standard of technical excellence. The logo, and accompanying sleeve note, is available by contacting sales@prismsound.com.

xinghuaman · 发表于 2016-11-27 20:26

Analogue interconnections

To maintain the high sound quality of Orpheus, it is important to follow some basic guidelines when making analogue connections to the unit.  This section discusses some things to watch out for.

Cable quality

Use of good-quality, heavy duty audio cables is recommended.  For microphone use, quad-twisted cables may give best results.  Cables with heavy screens are recommended, especially for unbalanced use.  Owing to mechanical differences between connectors from different manufacturers, it is advised to use cables with identifiable connectors from reputable manufacturers.  This is especially true for jacks, where unreliable tip connection can owing to the slightly non-conforming shape of some manufacturers' parts.  Neutrik connectors are used in Orpheus, and these are recommended to ensure reliably-mating cables.

Balanced versus unbalanced connections

Where possible, balanced interconnections should be used, since the audio signal is represented as a voltage difference between two dedicated conductors (neither of which is ground-coupled), which are usually closely-twisted to ensure that any interference pickup is cancelled out.  In unbalanced connections, the signal is represented as a voltage difference between a single signal conductor and an accompanying ground conductor.  Where dynamic ground-potential differences exist between the source equipment and the receiving equipment, this difference is effectively added to the unbalanced audio signal.

This effect has long been familiar in audio systems as 'hum loops', where the variation in ground potential occurred at line-frequency, and was developed by the flow of line-frequency currents to linear power supplies. Hum loops were usually resolved by either steering the currents along non-critical routes by re-arranging the topology of the system ground interconnections, or by mass-interconnection the system grounds using heavy gauge cable so as to minimize the hum voltage resulting from the current.

Obviously many items of analogue audio equipment only have unbalanced connections; this is especially true of consumer equipment, which is often used for monitoring even in professional studios.  If you must use unbalanced connections, keep them as short as possible and use good-quality cables with substantial screens.  If you have a choice, keep the signal level as high as possible on the interconnection, since this will make any interference proportionally less noticeable.

Instrument connections are often particularly vulnerable to hum and other interference, since they are usually unbalanced and low-level,  and frequently employ a long cable not selected for its interference-immunity qualities.  Also, the source impedance is usually high, making the connection particularly vulnerable to interference.

Some digital audio and computer equipment with switched-mode power supplies can cause particularly troublesome interference problems, especially for low-level, unbalanced signals.  This is discussed in the following section.

Interference

The increasing use of low-cost digital equipment and computers in the audio production process results in various potential problems for the remaining analogue devices.  It is well-known that the hostile power and EMC environment inside a typical computer is likely to be the limiting factor governing the audio quality of an internal analogue sound card. A solution to this is the use of external 'sound cards', such as Orpheus, with their own enclosures and power supplies allowing adequate space, power and electromagnetic peace and quiet for the well-being of studio-quality analogue circuits.

However, even the sound quality of external devices can be compromised by the proximity of some types of digital equipment.  Many low-cost switched-mode power supplies emit interference which can compromise system audio quality even at a distance.  The hostile mechanism is usually 'conducted interference', wherein the high-speed switching action of the power converter results in voltage and current transients being conducted back down their power cords.  If the equipment is connected to mains safety-ground, transients can also be conducted down the ground connection.  Radiated emissions (airborne radio interference) can also be a problem, but it is less common that this will have such a serious effect on audio quality.

Conducted power-line interference can cause problems in analogue equipment within the installation if its own power supply allows the transients to pass through to the audio circuits.  However, conducted ground interference can be even worse since, if the ground connection of the analogue equipment is modulated by switching interference, there is nothing that the designer of the equipment can do to combat it.

How much any conducted ground interference affects audio quality depends on many factors, mostly to do with how the various analogue boxes in the system are interconnected and grounded.  Where possible, high-level balanced connections should be used, just as in the case of hum-loops as discussed in the previous section.

Where ground-potential variations are caused by switching power supplies, the effect can be more difficult to resolve, since the signals can occur at more noticeable frequencies: although the supplies usually switch at frequencies too high to hear, the frequency is often modulated by variations in the load current over time, resulting in a continuous modem-like chirping in which can be heard particular events such as computer screen updates, disk activity etc.).  Another problem is that even heavy ground cabling may not reduce the effect of the interference, since high-frequency currents may not see much less resistance in a thick conductor than a thin one.

How do the equipment manufacturers get away with this?  Surely there are stringent regulations covering conducted and radiated emissions?  Well that's true, but the level of emissions which can result in audible degradation of low-level, unbalanced audio interconnections are well below legislation levels.  Unfortunately, computer power supplies (and especially the switching wall-warts and line-warts which power notebook computers and other small items) are amongst the worst offenders.

Vinyl decks

Orpheus is equipped with an RIAA de-emphasis filter to allow direct connection of a vinyl deck, as described in the Analogue inputs section.  Since vinyl decks usually have a low-level, unbalanced output it is important to minimise interference as discussed above when connection a vinyl deck.

Since most magnetic cartridges require a higher input impedance than that of the Orpheus microphone preamplifier input, it is usually best to connect a vinyl deck to the instrument inputs using a pair of phono-to-mono-jack cables.  The instrument gain controls can then be set to an appropriate level for the particular cartridge.  The 1MR input impedance of the instrument inputs will work satisfactorily with most magnetic phono cartridges (which are 'moving magnet' types), but with some cartridges, improved frequency response and noise levels can be achieved by fitting the cartridge's required load resistance (usually 22kR or 47kR) across the instrument input terminals; this is best achieved by soldering it inside the jack.  Moving coil cartridges have a lower output level and require a lower preamplifier input impedance.  These are best connected to Orpheus' mic inputs, or may require a dedicated preamplifier.

Most vinyl decks have a ground wire separate from the audio connectors.  Connection of this wire for lowest hum is often a matter of trial and error.  Ideally this should be connected to Orpheus' analogue signal ground (the outer of the instrument input jacks, or pin 1 of the mic input XLRs).  Since no dedicated terminal exists on Orpheus, it is usually easiest to connect the wire to the outer of one of the deck's unbalanced output connectors.  In some situations, a direct connection to local mains ground may work better.

In summary

• Use good-quality cables with reputable connectors;

• Use balanced connections where possible; if you must use unbalanced connections, keep them short;

• Ensure that signals passing between equipment do so at as high a level as is practical;

• If switching interference is heard, try to identify the source equipment by unplugging things one by one.  When you find the culprit, either re-plug it a long way from the audio equipment, or use a power filter, or both.