Words that describe audio

An article on Audiophile Review talks about a research project whose aim is to find the best words, or categories, with which to describe and quantify audio performance. In the end they come up with just three: “timbre”, “space” and “defects”.

At face value, it strikes me that this is two words too many. Wouldn’t it be simpler to forget “timbre” and “space” and simply list the “defects”?

The answer is that this list shows that audiophiles do not believe that there is an unambiguous correctness (or neutrality) that a system can aspire to. They think that a good audio system adds its own character to the recording and that one chooses it just as one would choose a violin, fine wine or work of art. This character, or musicality, may be specified in terms of timbre and space and is independent of any defects.

If only there was a word to describe this aspect of audio. It is analogous to anthropomorphisation i.e. the attribution of human motivation to inanimate objects and animals. In this case it is the attribution of the characteristics of musicians, music and musical instruments to electronic hardware. Something has gone seriously awry. People fully understand that a television set should be as neutral as possible in order to experience art, performance and culture through it. I think they used to understand this about hi-fi (hence the name) but not any more.


Hardware Fetishism

[28/03/15 UPDATED]

More than any hobby I know of, hi-fi is dominated by ‘fetishism’ for the hardware. £10000 audiophile cables are the most obvious and extreme manifestation of it, and hilarious to ‘civilians’, but I would guess that the majority of audiophiles are susceptible to this vice to a greater or lesser extent. The question is, why?

One reason, I would guess, is the direct association that we perceive between the hardware and the rather marvellous thing that is music. In our minds, our systems transport us to concert halls and recording studios, providing spectacularly fine experiences which are, perhaps, as good as driving a fast car or eating the finest food. Maybe we even think our systems are musical instruments. On that basis, why wouldn’t it be worth spending a few extra quid on the best hardware?

Well, the answer, I would say, is that the hardware is merely a means to an end. It is not the music, nor is it a musical instrument. The hardware merely needs to be adequate, like kitchen equipment only needs to be adequate to the task of preparing a fine meal. What possible difference can it make whether the knives cost £30 or £3000? Above a certain level of quality the dinner guests cannot possibly perceive a difference. Even the most ardent gastronaut would never dream of spending £3000 on a kitchen knife. (It seems like a ridiculous analogy, but…  £10,000 cables… at least it takes some skill and knowledge to design and make a kitchen knife that works).

I suspect that most audiophiles must know, deep down, that the differences between adequate amplifiers are tiny. Ditto DACs. And surely – please! – they must know that cables are just cables. So why are so many audiophiles prepared to spend £3000 on such mundane items as amplifiers, DACs or, unbelievably, pieces of wire? Surely it’s obvious that the raw cost of building adequate versions of these items is much, much lower than that? An AV amplifier can be massive, powerful, bristling with I/O, well-constructed, well-specified and have a wealth of features including exemplary DACs, yet costs closer to £300 than £3000. On that basis a flea-powered audiophile amp should cost about £30. Clearly, in the audiophile’s mind the hardware takes on a mythical significance that is detached from reality.

In my world, several thousand pounds of disposable income represents untold mornings scraping frost off the car when I’d rather be in bed, traffic jams, meetings, desk-bound lunches, hard technical work, being in places I’d rather not be, yet more traffic jams, and so on. It represents time away from home, away from the family. Or time when I could be climbing mountains, or in the pub, or listening to music. The idea that I would sacrifice a significant chunk of the time I have left on this earth to buy a cable is utterly, utterly ridiculous!

Maybe this affliction is unavoidable for most audiophiles. Something of an ‘elephant in the room’ could be that if they never actually get to hear an “adequate” audio system, then they can never know when to stop churning the equipment. I don’t mean there’s a conspiracy between manufacturers to keep the punters shelling out by deliberately manufacturing defective systems, but more that audiophiles are like those deer in Eastern Europe that still won’t cross the boundaries of electric fences that were removed several generations ago. As I have mentioned before, I think the hobby of audio is stuck in its own groundhog day and people would rather spend vast sums of money (= the best times of their remaining lives) on imaginary gains at the very margins of fundamentally-flawed systems than to try a radical departure from what they know. However, as a direct result of this, even if they do see the light the available commercial options are limited.

Let me tell you of my experience of “adequate” audio. My system is built on a shoestring but is nevertheless very large and powerful. It is not a conventional audiophile system, but uses active DSP-based crossovers with correction based on measurements of the drivers. With this system there is no aspect of the sound with any obvious room for improvement. Needless to say, with a string quartet, solo piano or girl-and-guitar it is superb (not that I listen to those types of music much). However, the true test of any system is the big symphony as this brings out problems with intermodulation and colouration that just don’t show up with solo performers and small ensembles. I happen to like that sort of music so it is important for me that the system can cope with it. Please note that I am not suggesting that some systems are better suited to certain types of music than others per se. I think the truth is that many systems are capable of giving a plausible (but not necessarily accurate) rendition of a string quartet or small jazz ensemble, but that as the complexity of recordings increases, systems fall by the wayside until only the best (i.e. “adequate”) can cope. With a complex recording the reproduction does have to be accurate as the listener has too many diverse cues to go on, and the large amount of information will collapse into a tangled mess if there’s significant distortion.

With my “adequate” system, if I play the most complex, huge recordings possible (and I would cite this piece as an example), there are no deficiencies crying out for mysterious magic from a £10,000 amplifier or the indefinable warmth that only a £100,000 turntable can give. Basically I have before me a large space that stretches beyond the walls of the room, a symphony orchestra that is just ‘there’ – no need to analyse it – in perfect clarity, and a massive pipe organ that physically shakes the room. I play the piece loud, and it’s thrilling. The final crescendo is so powerful that it is literally overwhelming – I’d swear it even has a visual element to it. But as the thunderous sound dies away, my ears aren’t ringing, just as they wouldn’t after a real symphony concert. I have just had the unmistakable experience of sitting through a huge symphonic performance from the best seat in the house (or even better). It’s physical not just auditory.

Or I can hear the best possible performances by The Who in the studio (the band right in front of me), or Jimi Hendrix. Or Neil Young – no need for Pono.

The point is that through my adequate system I am experiencing ‘the real thing’. Why would I need to change it? There is nothing about it that seems wrong. Why would it occur to me that a different amplifier (or selection of amplifiers) could make it sound significantly better? And as for cables, well the idea is just preposterous. I think that only when the system is more-or-less correct like this can we rationalise the potential for improvements vs. cost.

I can easily see that by going conventional then all bets would be off. Take away the visceral but clean bass and replace it with the contortions of a ported speaker, or simply ditch the bass altogether, and suddenly we are in the realms of unnatural sound: loss of acoustic cues, loss of detail, unpleasant, ear-tormenting timbre at higher volumes. Go passive and lose the effortlessness of directly-connected independent amps, lose the cone damping, lose the steadfast accuracy of DSP crossovers, lose the driver correction, lose the ability to play at realistic volumes. Yes, I too would be casting around, searching for a miraculous silver bullet.

But even just talking like this we are already in the realms of hardware fetishism: breaking the sound down into technical gobbledygook that has nothing to do with music. If our equipment is already at a certain minimum standard and we can identify anything in the sound that is reminiscent of these non-musical terms then the battle is already lost; the basic topology is wrong and the most expensive amplifier, DAC or cable in the world isn’t going to make more than 0.01% difference to the sound. On the other hand, if there is nothing wrong with the sound now then what degree of improvement could we expect by substituting different DACs, amps and cables that all measure only slightly differently from what’s there already?

The Audiophile Cable Fantasy

I was going to leave cables alone as it felt too much like shooting fish in a barrel, but I keep seeing articles that conflate all critical comments and questions concerning audiophile cables with “trolling”. Of course there are bad mannered gangs of internet crusaders out there, but reasonable, sceptical questions are not “trolling”.

In no particular order I would make these points:

live and let live

The notion of “live and let live” is a very fine one, of course, but merely by publishing a statement (like the one you are reading) the writer is implicitly criticising those who don’t agree with it. Hopefully they have something to back up their claims. The cable fraternity is effectively saying that all sceptics are philistines who cannot hear the subtleties that they can, yet without any reasoned argument for this claim. Their views influence all hi-fi discussion forums and confuse newcomers. Feathers are ruffled. It isn’t surprising that some people react negatively.

Developing one’s own audio system focuses the mind

If you are using DSP to create crossovers and to correct your system then you are very much in touch with what the signal is, and does. You are pulling the signal apart, processing it based on actual measurements of the system, and then putting it back together again in the analogue domain. There is no need for there to be anything mysterious or magical going on in the cables. If the cable provides adequate shielding and tolerable values of L, C and R (this is nothing special), can handle the current comfortably, is linear from the highest down to the lowest possible signal levels, and has no unwanted exotic properties like microphony then nothing more is required of it. All of this can be measured and quantified objectively, although it would be a pretty uninteresting exercise as almost all cables would be very similar.

An active crossover system throws a further spanner into the works of the audiophile cable fantasy. If the cable is only passing a small part of the spectrum then is it likely to have any of the following attributes taken from a real cable review: “substance, color, clarity, presence, scale, drama, flow, momentum, impact, humanness”? What happens when we combine cables with a mixture of these supposed characteristics? It gets even sillier when the signal is converted into numbers (digitised) and yet the cables used to convey the digits are thought to possess those self-same attributes.

Does science work or not?

If audiophile cables “work” then science and engineering do not. Science and engineering have no explanation for how one cable could give the sound “drama” and another not. (Well they do, and it’s called “placebo” or “expectation bias”). They can, however, explain how the capacitance of one cable, in the non-critical context of an audio interconnect, might roll off the treble by 0.01dB more than another cable. But any mythical ‘goldilocks’ RLC characteristics could be copied in cables costing $2 per metre, anyway. (If a troublesome troll insists on making that point, the cable industry goes on to imply that it’s more than just RLC.) Science and engineering certainly can’t explain how one ethernet cable could sound different from another. This stuff wouldn’t be controversial if there was any published research anywhere that showed how a cable can physically affect sound in any but the most benign, predictable ways. There isn’t. And yet science and engineering produced the basic hardware upon which the “tweakers” work their miracles. Funny that.

“placebo” is a real phenomenon

There is actual research that shows that placebo is real. Audio would be the perfect area to study it. Placebo is sufficient to explain all cable-related sonic effects, and until objective measurements show otherwise it must be the default assumption for any cable-related claim.

“I ran a cable design company…”

“I ran a company designing cables and we explained to people how they work” – is, to a rationalist, no different from saying “I run a crystal healing centre and we have leaflets that explain how all our procedures work.” More in the form of published, verifiable research is needed before a judgement is possible. Simply making statements like “Our copper is 99.9999% pure” does not tell us anything about how it influences the sound (i.e. not at all). Even saying “Our copper is 99.9999% pure which makes it 1.006 times as conductive as ordinary copper” doesn’t tell us anything either, if the only effect of a tiny increase in conductivity is an immeasurably-small increase in volume and nothing else. It isn’t enough to make scientific-sounding statements – that may even be correct – if they mean nothing in terms of the effect on the sound. Leaving the punter to draw the wrong conclusion would be a cheap trick.

The signal and power have already come through miles of cheap cable…

…so what difference can the last metre make? I asked this on one cable-related blog but was told I just didn’t understand otherwise I wouldn’t be asking such a stupid question.

On a related point, why don’t they use special cables in recording studios? Or research establishments? Or hospitals. Or the military? If lives depended on it then you’d have thought the professionals would have cottoned on by now.

Micro-diodes are rectifying our signals

Cable people often say that copper oxide is forming “micro-diodes” that are rectifying the signal (it’s not just about R,L and C you see). A copper oxide/copper diode can occur and has a forward voltage of 0.2V, so the micro-diodes idea might hold water if there were any micro-diodes in the signal path – but there aren’t. The connectors are bonded to the copper directly (if not then they’re defective) and the signal makes its way down the multiple copper strands where it is immaterial whether they are in direct contact with each other or separated by “micro-diodes”. There is no potential difference between the strands because they are bonded to each other directly at the connectors (and are no doubt in direct contact elsewhere) so no current flows across the “micro-diodes”. The multiple strands behave as a single piece of copper. If there’s anything wrong with this explanation it would be a lovely piece of research work for someone to do. The rectification must show up as distortion, or it doesn’t exist.

No one likes to see people being ripped off

Finally, there is an element of not wishing to see people being ripped off. In most areas of business there are standards and laws to prevent gullible people from being taken advantage of –  even if they are happy about their purchases. I could launch a new brand of petrol that used the power of placebo to convince people that it gave them more miles-per-gallon and made their engines last longer. I might even be sincere, believing that my new additive really worked, based on some measurements-free pseudo-science I had made up. People would buy it, and be happy about it, but there are laws to prevent me from making money that way unless I can prove my claims objectively. Not so in the audio industry.

Thoughts on antiprogress


Luddites opposing progress

The core ‘functions’ of an audio system haven’t changed in decades. In reality, the only functions that the audiophile really needs are two speakers and a volume control. But industry uses technical improvements and extra features to get at the customer’s wallet, and the result is usually progress that benefits everyone in the long run. So digital audio supersedes analogue, and computerised systems offer greater convenience, providing such wonders as streaming, downloads, searchable databases, artist info and playlists.

But as time has gone by, audiophiles have found themselves engaging with music less than they used to. Casting around for something to blame, they have developed a meme that says it is the new technology that is killing music. This ignores possible factors such as older people simply ‘having heard it all before’ and, as the old codgers they are now, being incapable of re-living the days when music was inextricably linked with fashion, youth culture and contact with the opposite sex. Today’s younger people may be less connected with music than they used to because they have never been exposed to boredom. Maybe they have more to do with less time available: tweets to send, new emoticons to express. Music with tunes is now seen as so un-hip it’s positively Clarkson-esque, and anyway there’s a theory that the only way modern youth can rebel against their parents is to be cautious and conservative. All of which could conspire to render new music as somewhat bland.

Whatever the actual reasons, it appears that something is killing music for both old codgers and da yoof. The chosen narrative is that it must be the new technology.

As a result of this perceived disconnection from music, part of the zeitgeist of the audio world is a view that progress in itself is a bad thing. In its place there is a push for the introduction of artificial inconvenience and imperfection in the hope it will make the experience of playing a record more authentic. In the world of commerce, this is surely an unusual phenomenon. The larger companies that would naturally have been innovating and introducing new features to tempt the customer to replace their systems, are left with nothing to offer. Smaller companies and one-man garage operations find themselves able to operate entirely within their comfort zones, with no need to keep up with the larger companies at all. All they have to do is to re-hash technology that hasn’t changed for many decades – a very unusual technological industry indeed. One of the biggest advantages that large companies have, that of being able to make advanced products at extremely low prices, does them no good. And if they try to compete by flogging a flimsy turntable that is less advanced than the ones they stopped selling in 1989, it just looks like pathetic bandwagon-jumping. (I recently saw such a ‘named brand’ turntable in Richer Sounds, selling for nearly £150…)

Antiprogress means that audiophiles are only impressed by equipment that is perceived to be hand made and sold at a “reassuringly expensive” price. By even trying to enter the current ‘high end’ market, the large companies are effectively legitimising antiprogress, and killing their own future business. By pushing the vinyl thing, the music industry has further tainted its biggest product, the digital download.

Put it all together, and it could look as though the audio and music industries have sleep-walked into some major marketing problems, and audiophiles have trapped themselves in Groundhog Day. We may find soundbars, Bluetooth speakers, surround sound, and all-in-one streaming systems in ordinary high street shops – yes, it makes me feel queasy to think about it. We may find turntables, valve amplifiers and passive speakers that cost £20,000 in specialist shops – an equally nauseating prospect. We may find mid-fi separates here and there. But I suggest that we are unlikely to ever see a large, Meridian-style DSP active speaker system for sale at under £1000 anywhere. For me, that would be the worthy successor to those systems of the 1970s; purist’s hi-fi that is startlingly good, and is not ‘discreet’ or hidden away, and doesn’t need added ‘authenticity’ and a comically-large price tag.

A Waste of Energy

The EU is engaged in a continuous clampdown on inefficient appliances including light bulbs, vacuum cleaners, central heating pumps, and now coffee makers.

I wonder whether they will ever take an interest in hi-fi equipment. After all, someone running a 100W stereo Class A amplifier can expect to burn 600W or more of power continuously! And that could be for several hours each day. This would more than wipe out the energy saved when the EU banned filament bulbs and forced the audiophile to buy LEDs for his house.

They might also look at the Devialet Phantom, which burns 3000W to achieve a sound level that could be reached at much lower powers simply by using a larger box – and will probably sell in much greater numbers than the Pass XA100.5. Sure it’s Class D and probably just as efficient as a Class AB system at lower volumes, but it’s still a flagrant ‘up yours’ gesture in the face of the green-ness that we’re all supposed to espouse – the green-ness that means that non-home owners are forced to subsidise homeowners’ profits from solar cells on their roofs, while other people burn off that energy for no actual benefit whatsoever.

I don’t suppose the EU will take an interest because the numbers aren’t big enough. But even if they were, I believe that the audio world is so far away in the land of the fairies that if the EU consulted the industry ‘experts’ on the matter, the officials who are taking such a hard line on coffee makers would actually be persuaded that there simply is no substitute for 600W of continuous wasted heat if we want to reproduce the sound of an acoustic guitar really well.

Trying Linux

UPDATED 16/03/15ubuntu-logo-8647_640 Approximately every two years I find myself inspired to have a go with Linux. I install Ubuntu on an old PC and congratulate myself on having finally made the right choice. Everything works fine: all the devices are auto-detected correctly, and although the graphics and text are a bit lumpy, it looks as though it can do everything Windows can do. It never lasts. Within a short time I try to do something beyond the basic web surfing and word processing and it doesn’t quite work. So I go to the web, and of course there’s usually a solution buried in a forum somewhere, and it invariably involves editing a config file. But along the way I may have found several other ‘solutions’ that didn’t work, and for each I maybe edited a different file or changed something using some little app I’ve installed. At the end, even though the system may be working, I am never quite sure how I got there, nor confident I could reproduce the same working system on another PC.

Well, the time has come again, and I am typing this using the latest version of Ubuntu. Everything is wonderful so far, and even Spotify is running flawlessly. Specifically, though, I want to get my active crossover system working on Linux, not Windows. My experience with Windows 7 running on slightly older PCs is not good. I have a laptop approximately 5 years old which will grind almost to a halt for several minutes every day, performing some sort of scan of itself, and I don’t know enough to do anything about it. The desktop PC that I use for the active crossover is slightly better, but it, too, takes quite a while to ‘warm up’ and is also prone to the occasional glitch while playing music, due to deciding to update its anti-virus database – I am sure it was not a problem with Windows XP. In contrast, running Ubuntu on an older desktop PC without much RAM, the experience is one of ‘solidity’. I am not experiencing the operating system going AWOL for several seconds at a time. But it comes at a price. I really, really don’t want to have to understand the details of any operating system, and Windows is good for the person who maybe wants to dip into a bit of programming (a distinctly different activity from IT) without having to worry too much about the really low level details. Windows feels as though it is ‘self-healing’. Every time the PC is turned on it starts scanning itself, checking for inconsistencies, downloading updates. New hardware is detected automatically and the user never edits configuration files. Ubuntu feels a little different. By all means correct me if I am wrong, but the impression I get is of a system that is dependent on lots of configuration files that are not hidden from the user. Of course these files get changed by the operating system itself (just as Windows must change its hidden configuration files) and there are little applications that you can install that simplify changing the parameters of various sound cards, say (more on this later). But occasionally the configuration files must be edited by the user using a text editor. One typo, and the PC may refuse to boot!

As I mentioned, I am hoping to run my active crossover stuff on Linux, not Windows. In order to achieve this I must loop continuously doing the following:

  1. Extract a chunk of stereo audio from an ‘input port’ that receives data from my application of choice (media player, Spotify etc.)
  2. Assemble the data into fixed-size buffers to be FFT-ed.
  3. Process with FIR filters to produce a separate, filtered output for each driver.
  4. Inverse FFT.
  5. Squirt the results out to six or eight analogue channels, or if feeling ambitious, HDMI (that would be the dream!).

It’s a very specific, self-contained requirement. I can handle numbers 2 to 4, no problem. 1 and 5 are the tricky ones, and seem to be a lot trickier than they, perhaps, might be. They weren’t all that easy in Windows, either, but I eventually came up with a scheme that kind of worked.

Here’s where it gets very specific: under XP I was able to use a single Creative X-Fi surround sound card as both the ‘receptacle’ for PC audio which I could then access with my application, and also as the multichannel DAC that my application could squirt its output to. Under Windows 7 the driver for the sound card was ‘updated’ and I could no longer access it as the receiver for general PC audio – I could still have used it for S/PDIF, analogue Line In etc., however. In the ideal world, the ‘receptacle’ would just be some software slaved to the output sample rate, I think, but I don’t know how to create such a piece of software – it would appear to Windows to be a driver I would guess. I could buy a piece of software called Virtual Audio Cable but I could never be sure whether that would always be re-sampling the data, and I’d rather avoid that. In the end, I used a method that I knew would work: I slaved a ‘professional’ audio card to the X-Fi using S/PDIF from the X-Fi. The M Audio 2496 can slave its sample rate to the S/PDIF (using settings in the M Audio-supplied configuration application) so I was able to send PC audio to the M Audio and my application could extract data from its ‘mixer’ at the same sample rate. Keeping the input and output on separate cards like this has some advantages when it comes to making measurements of the system while it is working, I think.

As a start I will probably try to do the same thing under Linux. I am attempting to use an Asus Xonar as the multichannel DAC, and another M Audio card I had lying around as the slaved source. It’s almost certain that I could achieve the objective without a second sound card, but I really don’t know how to do it [update 30/06/15: maybe I do know how to do it now]. Linux audio seems to have several ‘layers’ that I don’t understand (but as yet I have no view of them as layers, more as spaghetti). Really, I would like not to have to know anything about them at all, but this seems unrealistic. I have established the following:

  • I can do lowish-level audio stuff using the Alsa API. I can refer to specific cards by names that I can bring up with certain command line (shell) queries. Are these names guaranteed to stay the same in between boots? I don’t think so, but there are ways of editing the config files to associate names I choose to specific cards – I think.
  • There is a highly comprehensive system called JACK that allows “JACK-aware” programs to have their audio routed via a user-configurable patchbay. It can handle re-sampling between separate cards transparently. Brilliant, but I don’t think Spotify is “JACK-aware” for example so I’m not bothering with it. [Update 30/06/15: I want to avoid any form of re-sampling anyway]
  • Ubuntu has PulseAudio installed already (I think) and using an application (that I had to install) called Pavucontrol I can direct Spotify, and presumably other apps, to send their outputs to any of the sound cards in the system. Does this get written to a file and saved when I exit it? I think so. PulseAudio may be the thing I need, possibly being capable of creating software “sources” and “sinks”. But is it always resampling the audio to match sample rates even when that is not needed? More investigation needed. [Update 30/06/15: Pulseaudio cannot be guaranteed not to resample. I have removed it from the machine].
  • I installed a little program called Mudita24 that gives me most of the functionality of the app that is supplied for M Audio cards under Windows. It will let me slave the M Audio to S/PDIF. But without a lot of rummaging around on the web, finding this solution was not obvious. Will the results be saved to a file so I don’t have to call this up every time? I don’t know. [Update 30/06/15: the M Audio-compatible drivers don’t seem to work properly. I have abandoned this idea].
  • I found a “minimal” example program that can send a sine wave to an output via Alsa. The program is anything but minimal and allows the user to select from a large number of alternative sample rates, bit depths etc. etc. and has copious error reporting. My version of “minimal” is much shorter! I adapted the program for eight channels, and am sending a separate frequency to each of the Xonar’s outputs. It seems to be working quite solidly. I can’t be absolutely sure that the Xonar isn’t applying surround sound processing to the signals yet, though. Question: should I be programming using Alsa or PulseAudio? [Update 30/06/15: answer is most definitely ALSA only].

I don’t mind if everything is low level, nor do I mind if the operating system handles everything for me. What I am not keen on is a hybrid between the operating system doing some things automatically, and yet having to manually edit files (I haven’t done that yet, though) or having to install little apps myself. How are they all tied together? I don’t know.

UPDATE 10/03/15 Installed Ubuntu on my erratic Windows 7 laptop. On the hard drive I had to delete the ‘HP Tools’ partition to do it, as a PC can only have four partitions, apparently, and HP had used all four to install Windows – the things you learn, eh?

For the things I use the laptop for mainly, Ubuntu is knocking Windows 7 into a cocked hat. It actually responds instantly and doesn’t hang for tens of seconds with the disk light on constantly and the mouse pointer frozen. It’s taking some getting used to!

UPDATE 15/03/15 It is becoming clear to me that there is only one sensible solution for what I am trying to achieve (an active crossover / general DSP system under my control that can be applied to any source including streaming) that is guaranteed not to resample the data, nor is dependent on sound card-specific features, or needs two sound cards. Let me run this by you:

  • Media player apps need something that looks like a sound card to play into. Some apps will only play into whichever card is set as the default audio device.
  • If it’s a real sound card that’s being played into, I need to extract the data before it reaches the analogue outputs. This just may not be possible with many sound cards, and it is impossible to know without trying the card – no one cares about this issue normally.
  • I process the data into six or eight channels and then I need to squirt the results out to, effectively, some DACs (or HDMI). This is most likely a real, physical multi-channel sound card.
  • I believe that the media player’s sample rate is defined by the sound card it is playing into. If so, this is akin to asynchronous USB mode i.e. the media app is slaved to the sound card’s sample rate.
  • I would like to avoid sample rate conversion (and this would still be needed to convert between 44.09999 kHz and 44.10001 kHz i.e. there is no such thing as “the same sample rate” unless they are derived from the same crystal oscillator).

There is a Linux driver called snd-aloop which can act as a virtual audio node, recognisable by media player apps as a sink, but also recognisable by other apps as a recording source. I could send media player output into this virtual device, recognise it as a source for my application, process the data and send the multi-channel audio to a consumer-level DAC card without it needing any special features. However, there is a subtle problem: aloop’s sample rate is derived from the system-wide “jiffies” count. It will not match the sample rate of the DAC card even if they are both nominally 44.1 kHz.

I see just one sensible solution: I have to modify the aloop code so that, when the information is available, it gets its sample rate synchronisation from the DAC card. I could either modify aloop and send it this synchronisation information via a ‘pipe’ or shared memory (if that’s possible) from my active crossover application, or I can make my active crossover application a virtual sound card driver itself. Either way, I would need to register the driver with the system so that it can be set up as the default audio device (using the usual GUI-based sound preferences).

To any Linux programmers out there: does this sound sensible and do-able?

More later.

Update 30/06/15: It seems that there is an updated version of the snd-aloop driver which incorporates a dynamically-adjustable sample rate via the Alsa PCM interface. This could be precisely what I need.

Into another dimension

Does it ever occur to you that the whole idea of digital audio is rather amazing? Can you recall that in the early 80s you might actually have been extremely curious to hear how music made from numbers – something apparently completely unrelated to vibrations in the air – might sound? Maybe you were disappointed to find that it just sounded like music, and from then on you found it rather mundane and became quite dismissive of it.

In fact, do you ever stop and think how amazing it is that we have the ability to conjure up music, photographs and moving images at all? I think it is more impressive than most people realise, and a good example of how ‘magic’ quickly turns into the mundane, once we ordinary people get hold of it!

If you could put yourself back a thousand years, or even back to your own childhood, then simply suggesting how natural sound and images occur would be beyond you, never mind knowing how to capture and transmit them. While we may think that modern audiovisual technology is clever, the initial, gigantic leap occurred centuries ago when it was discovered that images are simply rays of light, and that sound is merely fluctuations of air pressure. Rays of light can be focused, collected or projected with lenses, and sound waves can be detected or emitted by a vibrating diaphragm.

A subsequent step was to work out ways to record and reproduce these physical phenomena, and maybe by that stage it was relatively obvious how to do it. Images could be captured automatically using a photo-chemical reaction, mimicking the action of an artist who uses a camera obscura to trace smudges onto a surface in response to the projected image. Sound could be captured by mechanically tracing the motion of a diaphragm into a wobbly groove that could later be used to re-wobble the diaphragm. (To this day people are occasionally struck by the thought of how interesting it is that a simple paper cone can duplicate any human voice or any musical instrument or an entire orchestra. Wouldn’t you have expected to need to build an artificial larynx etc.?)

The further step, of introducing electricity into the equation, would certainly have provoked my curiosity at the time: how would ‘electricity’ sound? By converting sound into electrical signals, it is transformed from one physical domain into another. The sound can be sent instantaneously from one side of the world to the other without any moving parts involved. Astounding! In fact, a human voice is in some small way part of a person, so in effect we are suddenly in a world where something of the essence of a living person can be stored and re-animated after his death, or can be sent into space, or duplicated a million times. Was there a time when people contemplated how weird this was, or within two seconds of hearing a radio or having their first phone call did the whole thing seem terribly dull?

Going digital was another ‘obvious’ step. It is both an unbelievably simple process akin to join-the-dots, and yet a mind-blowing transformation. While electricity is a different physical domain from the mechanical vibrations of sound, digitisation takes sound out of the physical completely and turns it into an abstract quantity! Numbers are perfect. Numbers can be stored perfectly. Numbers can be manipulated perfectly using mathematics. Once turned into numbers, the sound can be transmitted and duplicated without error, or stored without degradation until the end of time. It can be manipulated in an infinite number of ways. Things can be done with it that compensate for errors in the transducers that are used when we finally wish to bring it back into the physical world.

All of this can be done to any arbitrary level of precision. All of this can be done with minuscule chips that cost 50p and never wear out. Digital audio storage and transmission is cheaper to implement than analogue. It can be built into the trashiest consumer electronics. No special care or ceremony is needed to use it. It is so brilliant and so simple that the true magic is concealed.

And once it was launched, it took about two seconds for people to find the whole idea very dull!

It’s not real without physical media?

disc musical box


In this article on Audiophile Review the author talks about streaming versus physical media, and his coming to terms with the idea that in future we may not be able to buy our music in physical form.

I find it surprising that this might worry someone in the 21st century. I would have thought that changes in the workplace over the last two or three decades would have made us aware of the advantages of making everything digital, backed-up and available via the internet or ‘The Cloud’. Certainly in my work, if I were to draw an olde worlde pencil sketch I would feel that it was the opposite of permanent, whereas if I produced the drawing digitally it would be there forever – or so it would feel. Years later, not only could I find it easily, but at any time I could produce a pristine physical copy, un-yellowed by coffee rings and the passage of time. Until something is committed to the digital world, to my mind it feels ephemeral.

Of course, if the argument is that the quality of digital audio can never match an LP, or that streaming and downloading will always be inferior to CD quality, then the poor, anxious audiophile has a problem.