Sunday, 31 March 2013

So you think you understand Digital Audio?

I'm willing to bet that most of you don't.  In principle it is all very straightforward.  But in practice it really doesn't quite pan out that way.

Lets start off with all the stuff that you do know.

In digital audio, the music waveform is "sampled" on a regular basis.  Sampling means that the instantaneous magnitude of the waveform is measured and the resultant value stored somewhere.  The Sample Rate tells you how often these samples are measured.  The standard used in Compact Discs is 44.1kHz.  This means that the music waveform is measured 44,100 times per second, and the results of each measurement are stored.  The device that performs this sampling is called an ADC (Analog-to-Digital Converter).

Obviously, it is important that the precision with which the instantaneous waveform is measured is extremely accurate, and this precision is for the most part reflected by the "Bit Depth" with which the result is digitally stored.  The standard used in Compact Discs is 16-bits.  This means that the resultant value is stored as a 16-bit number.  16-bit numbers range from 0 to 65,535 and can take the form of whole numbers only.  Here, the largest possible amplitude of the musical waveform that can be recorded corresponds to the number 65,535, and the lowest possible magnitude (which is in fact the largest possible negative amplitude, since music signals vary between positive and negative), corresponds to the number zero.  By contrast, most high resolution recordings capture the musical waveform as 24-bit numbers.  These numbers range from 0 to 16,777,215 and so obviously are able to capture the musical waveform in a lot greater detail.

When we play back our digital music, all we have to do is re-create the musical waveform using the stored numbers, and this is where your DAC (Digital-to-Analog Converter) comes in.  Using the example of music stored in the CD format (which is commonly written in audio shorthand as 16/44.1) the job of the DAC is to grab a data value 44,100 times per second, each time creating an instantaneous voltage corresponding to the precise value encoded in the 16-bit data value.  If the DAC can do this accurately, then it will recreate the original musical waveform with a precision limited only by the extent to which the original music signal can be accurately reflected by 16-bit numbers.  Also - and this is obvious if you think about it - the timing of the 44,100 samples per second has to be exactly the same timing as when the music was originally sampled.

Now, here's the bit you probably don't know.

Unfortunately, one can be lulled into a false sense of security by this simplistic picture.  The reality is actually quite different.  You see, it turns out that it is a frighteningly complicated and prohibitively expensive task to build either ADCs or DACs that do the job I have just described.  So, given that most mobile phones contain both an ADC and a DAC, how is it we manage to get around this?

The answer is a technologically and mathematically challenging concept called "Sigma-Delta Modulation" (SDM).  This is basically an ultra-high-speed bit stream comprising only ones and zeros, such that at any point in time the magnitude of the encoded signal is reflected by the relative preponderance of ones over zeros.  If the bit-stream comprises almost all ones, then this would represent the maximum possible signal amplitude.  If almost all zeros, it would represent the minimum (or most negative) possible signal amplitude.  The beauty of SDM is that - without any signal processing whatsoever - the bitstream can be fed directly into the input of an amplifier, and it takes little more than an analog low-pass filter to convert it into music.  This is precisely how "Class-D" amplifiers work.

Although it is beyond the scope of this note to describe how, an ADC whose output is an SDM bit stream is a cheap thing to build, despite being an incredibly complicated thing to describe or even understand functionally.

So, in reality, with so few exceptions as to be not worth mentioning, all digital recordings are created using an SDM-based ADC, followed by a mathematically-driven signal processor which converts the SDM bitstream to a PCM data file.  And likewise, all DACs take the PCM data files they receive and put them through a mathematically-driven signal processor which converts them back to SDM, which is then converted to Analog using a simple Class-D output stage.

Why should you be concerned by all this?  Well, what you need to know is that, from a mathematical perspective, SDM and PCM are mutually incompatible formats.  Although both are at their roots nothing more than numbers, you cannot convert losslessly from one format to the other.  The conversion process invariably results in the musical data being irrecoverably "smeared" in the time domain.  So, in a real-world ADC, rather than actually sampling the music at fixed intervals in time, what the ADC is actually doing is calculating what the instantaneous amplitude of the music ought to be using the SDM bitstream as its reference.  Likewise for the DAC.

Many smart people will absolutely insist that PCM-based digital audio fundamentally has this, that, or another degree of perfection, based on somebody's published theoretical analysis, and if you claim to hear otherwise you are obviously fooling yourself.  And based on pure-PCM assumptions, these analyses are often very convincing.  Yet most serious audiophiles continue to hold that digital audio is on various levels less satisfying than good old analog.

I have developed a strong suspicion that many, many of the ills which we ascribe to digital audio may in fact be caused not by fundamental limitations of the PCM format, but by the sonically disruptive SDM-to-PCM and PCM-to-SDM converters that live unheralded in the ADCs and DACs that inhabit the playback chain.  Over time - quite a long time, I expect - I plan to experiment with this idea.

Sunday, 24 March 2013

Tchaikovsky's 6th Symphony

Tchaikovsky is a composer to whom it has become slightly de rigeur to gaze at down your long academic nose, if I can put it that way.  Simply because, in my opinion at least, his music is so fantastically accessible, which offends that school of modern musicians whose own efforts are not.  His grasp of melody remains, arguably, second to none, and his orchestrations seem so flawless, so effortless, yet so rich.  And I truly, madly, and deeply love his 6th symphony, the “Pathétique”.

Tchaikovsky wrote seven symphonies, and they can be quite readily divided into two groups, the ‘lightweights’ Nos 1, 2, 3, and ‘Manfred’, and the ‘heavyweights’ Nos 4, 5, and 6.  Symphony No 6 was his last work, and he died, age 53, only nine days after conducting its premier.  It is a colossal work, and stands close comparison to the recognized symphonic colossus of the 19th Century, Beethoven’s 9th.  Both were compositional tours-de-force which make profound, lasting impressions on their listeners.  Both enjoyed immediate enduring critical approval, although, interestingly enough, not at their actual premiers.

Indeed the structures of both works bear some comparison.  Both begin with an elegiac, extended first movement, move on to a wonderfully melodic slow movement followed by a standout rhythmic and dynamic scherzo, and conclude with an astounding statement finale.  But where Beethoven’s finale is majestic, uplifting, and extrovert, Tchaikovsky’s is foreboding, introspective, and wrenchingly emotional.  It is truly, truly magnificent music.

Fortunately, we are blessed with many excellent recordings to choose from.  The one which is most widely acclaimed is Evgeny Mravinsky’s reading with the Leningrad Philharmonic, and really, you cannot go wrong with it.  Herbert von Karajan’s 1956 recording with the Vienna Philharmonic is classy, but with a dated sound.  Claudio Abbado with the Chicago Symphony, Valery Gergiev with the Vienna Philharmonic, and Mariss Jansons with the Oslo Philharmonic are all highly recommended modern recordings with great sound.  Leonard Bernstein produced a stunning but idiosyncratic recording with the New York Philharmonic which sustains a deathly slow pace from the first bar that only Bernstein could get away with.

What is my own choice?  Well, I must admit to a lasting love affair with Bernard Haitink’s recording with the Royal Concertgebouw Orchestra of Amsterdam.  It is not perfect, but none of the others I have mentioned can claim to be either.  For sure, there are going to be those who would argue that none of my choices should be on anybody’s recommended list.  And that’s good.  Because one of life’s great joys is seeking out unfamiliar recordings of this powerful work, hoping to unearth the one miracle like Kleiber’s Beethoven’s 5th.

Happy listening!

Saturday, 16 March 2013

Digital vs Analog Volume Control

With the rise of computer-based high-end audio playback, a very interesting question is whether it is best to perform volume control in the digital or analog domain.  I have done some experiments in this area, and I thought I might share some of my thoughts on the subject.

The very best DAC/preamplifier combos have sufficiently low noise that they can resolve perhaps the 21st bit and even the 22nd bit of an audio signal.  However, you may have to choose between that and buying a fast car to achieve that level of performance!  The merely "very good" are capable of resolving the 19th to 20th bit as a rough guideline.  If you implement volume control in the digital domain, every 6dB of attenuation results in the loss of one bit of resolution.  If you play back music music with 24-bit bit depth, then all volume control results in irrecoverable loss of data via bit-depth reduction.  However, if you play 16-bit music, and pass it to your DAC in 24-bit format (something which most high-end DACs require anyway), then, depending on the quality of your DAC/preamp combo, you can in principle dial in up to 18-36dB of attenuation without audibly truncating the the music data.  All this assuming that your digital volume control is done in a first-class manner using a 64-bit audio engine or its equivalent, as implemented in BtPerfect.

So that's the theory.

On the other hand, volume control performed in the analog domain requires passing the signal through some sort of variable attenuator – such as a potentiometer, an active electronic equivalent, or a switched resistor ladder.  These components do actually degrade the sound, and to quite an alarming degree!  If you are in the habit of "tweaking" your audio equipment, you will know that a hardy market exists for after-market volume control potentiometers costing up to thousands of dollars each (!!!) to try and eliminate these sonic defects.  So the answer to the question boils down to whether or not the sonic degradation introduced by Bit Depth reduction is less intrusive than that introduced by a preamplifier's volume control.

As it happens, I have done some extensive listening tests on this subject, and I have surprised myself by the conclusions I have drawn.  Regardless of whether the music is 16-bit or 24-bit, I have found that performing volume control in the digital domain is qualitatively superior to performing it in the analog domain.  And the difference is not subtle – it is really quite massive.  No contest, actually.  I will temper that statement by saying that it for sure depends on the preamplifier you are using and the volume control circuitry it implements.  For example, I had a chance to discuss this with Dan d'Agostino, and while he agrees with me, he assures me that his new $30,000 preamplifier has a volume control that introduces no sonic degradation whatsoever!

There is a significant practical downside to performing volume control in the digital domain.  Basically, your DAC is connected to a preamplifier permanently set to maximum volume.  Depending on the rest of your audio equipment, the consequences of accidentally playing music at maximum volume may represent a risk that you are just not willing to take.  Most computer playback systems have a user interface that has not been designed with this concern in mind!  You would have to be very particular indeed about the procedures you go through each time you start to play music.

In my case, I run my Light Harmonic Da Vinci DAC directly into the inputs of a 300W/ch Classe CA-2300 power amplifier, feeding B&W 802 Diamond loudspeakers.  All of my serious listening is done with about 20-30dB of attenuation dialled in by BitPerfect using the iTunes volume slider.  I find this to be massively superior to routing the signal through my Classe CA-800 preamplifier with all digital attenuation turned off and a truly "bit perfect" signal passed into, and attenuated by, the preamplifier.

As this subject is taken up by a wider audience, and more different system configurations are evaluated, it will be very interesting to see what sort of a consensus emerges. 

[UPDATE] Since writing this post, the notion of digital volume control - if properly done - being potentially superior to analog volume control, has more or less evolved to become mainstream thought.  Only the very highest-grade (and, generally, highest-priced) preamplifiers can offer analog volume control with a remotely comparable performance.

My reference system has also changed.  I now use a PS Audio DirectStream DAC, and I listen to a lot of music in DSD format.  The DSD aspect has a great deal of impact on  the concept of using digital volume control, because you cannot perform that type of manipulation on a 1-bit bitstream.  At least not easily, and the issue of its impact on sound quality remains unresolved.  I cannot therefore dial in 20dB of attenuation in BitPerfect using the iTunes volume slider.

The DirectStream DAC processes the incoming digital data in its own unique way.  It converts all the incoming data to a 28MHz 30-bit PCM format, and then converts that to DSD128 which is fed natively to the output stage.  This enables them to perform essentially lossless digital volume control on the signal while at the 30-bit 28MHz stage.  I can access this stage of volume control using the DirectStream's remote control.  My Classe CA2300 power amplifiers have now been replaced by PS Audio BHK-300 Signature monoblocks, also 300W/ch, so I still spend most of my listening at a setting with 20-30dB of attenuation dialed inThe DirectStream offers a neat solution by providing a switch to toggle the gain of the output stage up or down by 20dB.  By engaging this switch, I can introduce an essentially lossless 20dB of attenuation at the analog output stage, and run the digital volume control in the 0-10dB range, which is close to lossless.

To add to the fun and games, PS Audio have released a 'BHK Signature' preamplifier, which sits between the DAC and the power amplifiers, with the DirectStream's volume control set to maximum.  I haven't heard this product, but Paul McGowan of PS Audio claims that the system actually sounds better with the preamplifier installed, which goes against all apparent logic.  If it does, though, then the only explanation to my way of thinking must be a less-than-ideal electrical interface between the output of the DirectStream and inputs of the monoblocks, which is corrected by using the new preamplifier.  Aaaah yes, the black art of analog electronics design.  In many ways, digital is soooooo much simpler :)

Thursday, 14 March 2013

Concord On A Summer Night

My wife is not a big fan of jazz, so I don’t get to listen to it much whole she’s around.  She likes music with a hook, a simple melody, a reliable rhythm, and an element of comfortable predictability.  If you step outside of those bounds in an effort to express your musical individuality, she’s not going to want to listen.  Here’s a jazz album I think I can get her to sit down to. And it’s a doozy!

Dave Brubeck’s jazz has all of the above attributes, yet somehow manages to “civilize” them into an accessible form.  Take his best-known song “Take Five”.  It has no obvious hook, a melody in 5/4 time, modulations of the beat pattern within extended 5/4 phrases, and extended improvisational solo workouts, yet somehow it is imbued with an easy-to-follow, easy-to-absorb character that is a pre-requisite of “popular” music in its many forms.   Indeed, this this is a pretty fair description of most of Brubeck’s work.

Perhaps one of the aspects of Dave Brubeck is that, while his bands employ musicians of the highest virtuosic quality, virtuosity is never allowed to intrude on the compositions. Brubeck's recordings are always expressions of the music, and never of the musicians.  In Concord On A Summer Night, this becomes especially evident on the ubiquitous Take Five, where Brubeck’s piano playing at one point soars with an orchestral majesty that very few pianists ever deliver, and yet by that time the listener is so absorbed in the music that the virtuosity is just not the focal element – the majestic exposition of the melody just carries you away with it.

Concord On A Summer Night is one of the finest live jazz recordings ever made.  One of these rare events when a magical performance is captured in a magical recording.  It was recorded in Brubeck’s home town, Concord, CA, in 1982, and among other things is notable for inclusion of a clarinet instead of the usual saxophone.  Brubeck was born in Concord in 1920, and his mother was a some-time concert pianist and piano teacher.  Dave had two older brothers who were active musicians, and it was felt that Dave should go into a career as a veterinarian.  At college he was persuaded to switch to music, but even so was nearly expelled when –incredibly – it was discovered that, despite his evident talents, he could not read music!

Happily, Concord On A Summer Night is available as an absolutely superb high-resolution recording from HD Tracks, although, to be fair, the CD version is not too shabby either.  Either way, it is that rare bird – an album that both hard-core jazz fanatics, and those who really don’t like jazz, can agree upon.  One of these nights, when she’s not expecting it, I will play it for my wife!