An opinion piece this week designed to get your backs up and make you think. You read a lot of brouhaha these days about how musicians are not making any money out of streaming services. There are so many streaming services available - some even offer high-resolution lossless content - and much like Netflix in the video domain, we as consumers can now access a lot of content for a nominal (i.e. affordable) outlay. How, you might wonder, can the musicians who create the music in the first place be making any money out of it?
Recently, a study has been doing the rounds which purports to analyze the revenues of the streaming service Spotify, and indicates how that revenue is divvied up among the Streaming Service itself, the Record Labels, the Writers/Composers, and the Artists. The report is available on the web site of Music Business World, and was prepared by the accounting firm Ernst & Young, so it has at least a minimum acceptable level of credibility.
Ask yourself this - according to the report, for every dollar you spend on Spotify, just how much of it ends up in the pocket of the artist whose music you are listening to? Before you go on to read the the answer, I want you to ponder the issue for a moment and ask yourself how much you think OUGHT to go to the artist? Also, stop for a moment to consider the rationale behind your calculation, so that it is a little bit more than a number you pulled out of thin air. On what basis should the artist receive whatever it was that you thought was appropriate?
So what did you come up with…? 50 cents? 20 cents? 10 cents? The actual answer is less than 7 cents. Not seven cents every time you listen to a track, but 7 cents out of every dollar you spend. If you subscribe to their premium service that’s about $10 a month. So your subscription to Spotify generates 70 cents a month to be shared among all of the artists that you listen to. Lets imagine that you listen to 20-25 tracks a day, and lets assume that the money gets split evenly among the artists on a per-play basis. In that scenario you are playing about 700 tracks a month. So each time you play a track, the artist you are listening to earns something like one tenth of one cent.
In some circles, this has aroused the anger of musicians who feel that the Spotifys of this world are screwing them out of their rightful earnings. First Napster, then bit torrents and file sharing, and now this!
There are two problems with this. The first is that, as best as anyone can tell, none of these streaming services are actually making any money! It is one thing to argue a case against someone who is making scads of money off the backs of others, but another thing entirely to vent your spleen at someone who isn’t even profitable - unless your complaint is about the lack of any profit itself, which isn’t the case here.
Can it really cost that much money to run Spotify? Which brings me to the second problem. What happens to all the money that you pay to Spotify? The answer is that Spotify in turn pays the majority of it in fees to the Labels. Spotify pays about 17 cents on the dollar in taxes, and uses 20 cents to run its own operations. The rest - amounting to nearly two-thirds of their revenues - is paid directly to the Record Labels who manage the distribution to the Artists. In other words, Spotify doesn’t get to decide how much of their take goes to the Artists - that is entirely within the purview of the Record Labels themselves.
So now lets take a look at the money that the Labels receive - how do they distribute that? According to the Music Business World report only 10% of what they receive goes to the Artists, and 15% goes to the Songwriters and Publishers, which means that the Labels keep a whopping 75% of the pie for themselves. That’s a lot of pie.
It is therefore wrong-headed for the Angry Artists to get all stroppy about Spotify eating their lunch. It is the Labels who are doing all the munching. And it has been thus for as long as there has been a music industry. But, the argument goes, it is a different world in 2015. Labels used to have to pay for record stamping plants, or even CD stamping plants. They had to maintain a sales force to get their product stocked by the music stores, and a promotional force to get their customers into the stores. Plus the costs of transporting the product internationally. Today this doesn’t happen any more. All of the above is theoretically replaced by an “Upload” button that someone has to punch.
But even taking all of that into account, it still misses the point entirely for the Artists to be taking pot shots at the Labels. If the Artist feels that the Label is charging too much for what they provide, then their solution is simple - they don’t have to sign with a Label. Like just about any transaction, if you don’t like the price, you don’t have to make the purchase. Unfortunately, though, for the majority of Artists, they don’t even have the option to not sign for a Label. The reality is that as an Artist you hope to generate enough buzz that a Label - any freakin’ Label will do! - will deign to offer you a deal. The idea that you can shop around and choose the one that offers you the best deal is a pipe dream for all but the privileged few.
For the Artist, what are the alternatives? The obvious one is that they can start their own Label. Sure they can … there’s nothing to stop them. Well, except one thing. You’ll need some money. And as an Artist without a record deal capable of putting one tenth of a cent in your pocket every time someone plays one of your tunes on Spotify, you won’t actually have any money.
The view from the other side of the fence is not all roses either. As a Label, you are hopefully making money from your roster of Artists. But they come and go, as do their sales. You always need to be replenishing your portfolio. For every new Artist that you have a budget to take on there are a hundred who are convinced that they are The One. You need to be really smart about which ones you sign and which ones you pass on. After all, you’re not as dumb as the Decca executive who passed on The Beatles because guitar music was going out of style, are you?
Once you’ve signed a new Artist you are going to need to pay for some studio time to record their new album. You’ll need to pay people to design the cover work and take publicity shots. You’ll probably need professional video work doing. You’ll need to schedule radio and TV spots if you’re sufficiently gung-ho about their prospects. And you’ll need to cut that deal with Spotify. All that expense must be incurred without any guarantee that you’ll ever get a penny in return. And for every Artist who generates a handy revenue stream for you, there will be four or five who fail to make any sort of impact at all. On top of it all, it may be you who screws up. The Artist may leave you and sign for another Label, and under their guidance hit the big time.
For this reason, most Labels are very controlling when it comes to their stable of Artists. They will control a large part of the product, how it sounds, whose arrangements are used - they’ll even kick out members of the band and bring in better session musicians. If they don’t like your songs they’ll use their own songwriters. The Labels are in the business of knowing what will sell and what won’t. They won’t always get it right, but like a profession stock trader, they’ll get it right way more often than you will. Even the poor sod at Decca who turned down The Beatles (his name was Dick Rowe) went on to sign The Rolling Stones. Consequently, the Artists very soon find out exactly where on the totem pole a place has been reserved for them, even as their backs are being patted and their egos pumped.
So, as a musician, if you can do all that then you don’t need the services of a Label, and you’ll make ten times as much from Spotify as you might otherwise have done. If not, then you have little choice but to work within the established Label system if you can get one sufficiently interested. Otherwise, as one certain Norman Tebbitt might have put it, you should consider getting ‘on yer bike’ and finding a proper job :)
Here’s the thing about musicians in particular, but Artists generally. You are only an Artist while you are creating art for your own personal satisfaction. As soon as you aim to sell it for even a modest profit you become a businessperson, no different from a restaurant owner or a plumber. Its a dog-eat-dog world, whether you’re selling art or amplifiers, and you need to have a minimum of business savvy if you are going to survive in it. You need to identify smart things to do and dumb things to avoid doing. The world has little sympathy for poor businesspeople. And it won’t pay $10 for something if there is something else it thinks might be just as good priced at $9.95. Don’t take my word for it. Spend some time in Walmart.
My advice to musicians who fret about how much they are getting from Spotify is simple. You are businesspeople first and foremost, and you had better start looking at yourselves in that light. Would you open a paint store that only sold green paint? I know I wouldn’t. The thing about business is that sometimes the best thing to do is not the same as the thing you really wanted to do. If you can’t - or won’t - see that, then your prospects for success will have a lot in common with buying a lottery ticket. Which is fine, because most of us do not make particularly good businesspeople, and rarely win the lottery. In which case you should go back to being an artist, and create art for no purpose other than your own satisfaction - in your spare time of course, since you’ll have a ‘proper’ job to do as well.
Tuesday, 10 February 2015
Tuesday, 3 February 2015
iTunes 12.1.0
I have been using iTunes 12.1.0 for a couple of days now. It seems to work fine with BitPerfect. The only issues I am aware of are those which also affected previous versions of iTunes and represent mostly edge cases and minor inconveniences. BitPerfect user can feel comfortable upgrading from iTunes 12.0.x.
Tuesday, 20 January 2015
High-Order DSD
As support for regular DSD (aka DSD64) starts to become close to a requirement for manufacturers of not only high-end DACs, but also a number of entry-level models too, so the cutting edge of audio technology moves ever upward to more exotic versions of DSD denoted by the terms DSD128, DSD256, DSD512, etc. What are these, why do they exist, and what are the challenges faced in playing them? I thought a post on that topic might be helpful.
Simply put, these formats are identical to regular DSD, except that the sample rate is increased. The benefit in doing so is twofold. First, you can reduce the magnitude of the noise floor in the audio band. Second, you can push the onset of undesirable ultrasonic noise further away from the audio band.
DSD is a noise-shaped 1-bit PCM encoding format (Oh yes it is!). Because of that, the encoded analog signal can be reconstructed simply by passing the raw 1-bit data stream through a low-pass filter. One way of looking at this is that at any instant in time the analog signal is very close to being the average of a number of consecutive DSD bits which encode that exact moment. Consider this: the average of the sequence 1,0,0,0,1,1,0,1 is exactly 0.5 because it comprises four zeros and four ones. Obviously, any sequence of 8 bits comprising four zeros and four ones will have an average value of 0.5. So, if all we want is for our average to be 0.5, we have many choices as to how we can arrange the four zeros and four ones.
That simplistic illustration is a good example of how noise shaping works. In effect we have a choice as to how we can arrange the stream of ones and zeros such that passing it through a low pass filter recreates the original waveform. Some of those choices result in a lower noise floor in the audio band, but figuring out how to make those choices optimally is rather challenging from a mathematical standpoint. Theory, however, does tell us a few things. The first is that you cannot just take noise away from a certain frequency band. You can only move it into another frequency band (or spread it over a selection of other frequency bands). The second is that there are limits to both how low the noise floor can be depressed at the frequencies where you want to remove noise, and how high the noise floor can be raised at the frequencies you want to move it to.
Just like digging a hole in the ground, what you end up with is a low frequency area where you have removed as much of the noise as you can, and a high frequency area where all this removed noise has been piled up. If DSD is to work, the low frequency area must cover the complete audio band, and the noise floor there must be pushed down by a certain minimum amount. DSD was originally developed and specified to have a sample rate of 2,822,400 samples per second (2.8MHz) as this is the lowest convenient sample rate at which we can realize those key criteria. We call it DSD64 because 2.8224MHz is exactly 64 times the standard sample rate of CD audio (44.1kHz). The downside is that the removed noise starts to pile up uncomfortably close to the audio band, and it turns out that all the optimizing in the world does not make a significant dent in that problem.
This is the fundamental limitation of DSD64. If we want to move the ultrasonic noise further away from the audio band we have to increase either the bit depth or the sample rate. Of the two, there are, surprisingly enough, perhaps more reasons to want to increase the bit depth than the sample rate. However, these are trumped by the great advantages in implementing an accurate D/A converter if the ‘D’ part is 1-bit. Therefore we now have various new flavours of DSD with higher and higher sample rates. DSD128 has a sample rate of 128 times 44.1kHz, which works out to about 5.6MHz. Likewise we have DSD256, DSD512, and even DSD1024.
Of these, perhaps the biggest bang for the buck is obtained with DSD128. Already, it moves the rise in the ultrasonic noise to nearly twice as far from the audio band as it was with DSD64. Critical listeners - particularly those who record microphone feeds direct to DSD - are close to unanimous in their preference for DSD128 over DSD64. The additional benefits in going to DSD256 and above seem to be real enough, but definitely fall into the realms of diminishing returns. However, even though the remarkably low cost and huge capacity of hard disks today makes the storage of a substantial DSD library a practical possibility, if this library were to be DSD512 for example, this would start to represent a significant expense in both disk storage and download bandwidth costs. In any case, as a result of all these developments, DSD128 recordings are now beginning to be made available in larger and larger numbers, and very occasionally we get sample tracks made available for evaluation in DSD256 format. However, at the time of writing I don’t know where you can go to download samples of DSD512 or higher.
In the Apple World where BitPerfect users live, playback of DSD requires the use of the DoP (“DSD over PCM”) protocol. This dresses up a DSD bitstream in a faux PCM format, where a 24-bit PCM word comprises 16 bits of raw DSD data plus an 8-bit marker which identifies it as such. Windows users have the ability to use an ASIO driver which dispenses with the need for the 8-bit marker and transmits the raw DSD data directly to the DAC in its “native” format. ASIO for Mac, while possible, remains problematic.
As mentioned, DoP encoding transmits the data to the DAC using a faux PCM stream format. For DSD64 the DAC’s USB interface must provide 24-bit/176.4kHz support, which is generally not a particularly challenging requirement. For DSD128 the required PCM stream format is 24-bit/352.8kHz which is still not especially challenging, but is less commonly encountered. But if we go up to DSD256 we now have a requirement for a 24-bit/705.6kHz PCM stream format. The good news is that your Mac can handle it out of the box, but unfortunately, very few DACs offer this. Inside your DAC, if you prise off the cover, you will find that the USB subsystem is separate from the DAC chip itself. USB receiver chipsets are sourced from specialist suppliers, and if you want one that will support a 24/705.6 format it will cost you more. Additionally, if you are currently using a different receiver chipset, you may have a lot of time and effort invested in programming it, and you will have to return to GO if you move to a new design (do not collect $200). The situation gets progressively worse with higher rate DSD formats.
Thus it is that we see examples of DSD-compatible DACs such as the OPPO HA-1 which offers DSD256 support, but only in “native” mode. What this means is that if you have a Mac and are therefore constrained to using DoP, you need access to a 24/705.6 PCM stream format in order to deliver DSD256, and the HA-1 has apparently been designed with a USB receiver chipset that does not support it. It may not be as simple as that, and there may be other considerations at play, but if so I am not aware of them.
Interestingly, the DoP specification does offer a workaround for precisely this circumstance. It provides for an alternative to a 2-channel 24/705.6 PCM format using a 4-channel 24/352.8 PCM format. The 8-bit DoP marker specified is different, which enables the DAC to tell 4-channel DSD128 from 2-channel DSD256 (they would otherwise be indistinguishable). Very few DAC manufacturers currently support this variant format. Mytek is the only one I know of - as I understand it their 192-DSD DAC supports DSD128 using the standard 2-channel DoP over USB, but using the 4-channel variant DoP over FireWire.
Because of its negligible adoption rate, BitPerfect currently does not support the 4-channel DoP variant. If we did, it would require some additional configuration options in the DSD Support window. I worry that such options are bound to end up confusing people. For example, despite what our user manual says, you would not believe the number of customers who write to me because they have checked the “dCS DoP” checkbox and wonder why DSD playback isn’t working! Maybe they were hoping it would make their DACs sound like a dCS, I dunno. I can only imagine what they will make of a 2ch/4ch configurator!!!
As a final observation, some playback software will on-the-fly convert high-order DSD formats which are not supported by the user’s DAC to a lower-order DSD format which is. While this is a noble solution, it should be noted that format conversion in DSD is a fundamentally lossy process, and that all of the benefits of the higher-order DSD format - and more - will as a result be lost. In particular, the ultrasonic noise profile will be that of the output DSD format, not that of the source DSD format. Additionally, DSD bitstreams are created by Sigma-Delta Modulators. These are complex and very challenging algorithms which are seriously hard to design and implement successfully, particularly if you want anything beyond modest performance out of them. The FPGA-based implementation developed for the PS Audio DirectStream DAC is an example of a good one, but there are some less-praiseworthy efforts out there. In general, you can expect to obtain audibly superior results pre-converting instead to 24/176.4 (or even 24/352.8) PCM using DSD Master, which will retain both the extended frequency response and the lower ultrasonic noise floor of the DSD256 original.
Simply put, these formats are identical to regular DSD, except that the sample rate is increased. The benefit in doing so is twofold. First, you can reduce the magnitude of the noise floor in the audio band. Second, you can push the onset of undesirable ultrasonic noise further away from the audio band.
DSD is a noise-shaped 1-bit PCM encoding format (Oh yes it is!). Because of that, the encoded analog signal can be reconstructed simply by passing the raw 1-bit data stream through a low-pass filter. One way of looking at this is that at any instant in time the analog signal is very close to being the average of a number of consecutive DSD bits which encode that exact moment. Consider this: the average of the sequence 1,0,0,0,1,1,0,1 is exactly 0.5 because it comprises four zeros and four ones. Obviously, any sequence of 8 bits comprising four zeros and four ones will have an average value of 0.5. So, if all we want is for our average to be 0.5, we have many choices as to how we can arrange the four zeros and four ones.
That simplistic illustration is a good example of how noise shaping works. In effect we have a choice as to how we can arrange the stream of ones and zeros such that passing it through a low pass filter recreates the original waveform. Some of those choices result in a lower noise floor in the audio band, but figuring out how to make those choices optimally is rather challenging from a mathematical standpoint. Theory, however, does tell us a few things. The first is that you cannot just take noise away from a certain frequency band. You can only move it into another frequency band (or spread it over a selection of other frequency bands). The second is that there are limits to both how low the noise floor can be depressed at the frequencies where you want to remove noise, and how high the noise floor can be raised at the frequencies you want to move it to.
Just like digging a hole in the ground, what you end up with is a low frequency area where you have removed as much of the noise as you can, and a high frequency area where all this removed noise has been piled up. If DSD is to work, the low frequency area must cover the complete audio band, and the noise floor there must be pushed down by a certain minimum amount. DSD was originally developed and specified to have a sample rate of 2,822,400 samples per second (2.8MHz) as this is the lowest convenient sample rate at which we can realize those key criteria. We call it DSD64 because 2.8224MHz is exactly 64 times the standard sample rate of CD audio (44.1kHz). The downside is that the removed noise starts to pile up uncomfortably close to the audio band, and it turns out that all the optimizing in the world does not make a significant dent in that problem.
This is the fundamental limitation of DSD64. If we want to move the ultrasonic noise further away from the audio band we have to increase either the bit depth or the sample rate. Of the two, there are, surprisingly enough, perhaps more reasons to want to increase the bit depth than the sample rate. However, these are trumped by the great advantages in implementing an accurate D/A converter if the ‘D’ part is 1-bit. Therefore we now have various new flavours of DSD with higher and higher sample rates. DSD128 has a sample rate of 128 times 44.1kHz, which works out to about 5.6MHz. Likewise we have DSD256, DSD512, and even DSD1024.
Of these, perhaps the biggest bang for the buck is obtained with DSD128. Already, it moves the rise in the ultrasonic noise to nearly twice as far from the audio band as it was with DSD64. Critical listeners - particularly those who record microphone feeds direct to DSD - are close to unanimous in their preference for DSD128 over DSD64. The additional benefits in going to DSD256 and above seem to be real enough, but definitely fall into the realms of diminishing returns. However, even though the remarkably low cost and huge capacity of hard disks today makes the storage of a substantial DSD library a practical possibility, if this library were to be DSD512 for example, this would start to represent a significant expense in both disk storage and download bandwidth costs. In any case, as a result of all these developments, DSD128 recordings are now beginning to be made available in larger and larger numbers, and very occasionally we get sample tracks made available for evaluation in DSD256 format. However, at the time of writing I don’t know where you can go to download samples of DSD512 or higher.
In the Apple World where BitPerfect users live, playback of DSD requires the use of the DoP (“DSD over PCM”) protocol. This dresses up a DSD bitstream in a faux PCM format, where a 24-bit PCM word comprises 16 bits of raw DSD data plus an 8-bit marker which identifies it as such. Windows users have the ability to use an ASIO driver which dispenses with the need for the 8-bit marker and transmits the raw DSD data directly to the DAC in its “native” format. ASIO for Mac, while possible, remains problematic.
As mentioned, DoP encoding transmits the data to the DAC using a faux PCM stream format. For DSD64 the DAC’s USB interface must provide 24-bit/176.4kHz support, which is generally not a particularly challenging requirement. For DSD128 the required PCM stream format is 24-bit/352.8kHz which is still not especially challenging, but is less commonly encountered. But if we go up to DSD256 we now have a requirement for a 24-bit/705.6kHz PCM stream format. The good news is that your Mac can handle it out of the box, but unfortunately, very few DACs offer this. Inside your DAC, if you prise off the cover, you will find that the USB subsystem is separate from the DAC chip itself. USB receiver chipsets are sourced from specialist suppliers, and if you want one that will support a 24/705.6 format it will cost you more. Additionally, if you are currently using a different receiver chipset, you may have a lot of time and effort invested in programming it, and you will have to return to GO if you move to a new design (do not collect $200). The situation gets progressively worse with higher rate DSD formats.
Thus it is that we see examples of DSD-compatible DACs such as the OPPO HA-1 which offers DSD256 support, but only in “native” mode. What this means is that if you have a Mac and are therefore constrained to using DoP, you need access to a 24/705.6 PCM stream format in order to deliver DSD256, and the HA-1 has apparently been designed with a USB receiver chipset that does not support it. It may not be as simple as that, and there may be other considerations at play, but if so I am not aware of them.
Interestingly, the DoP specification does offer a workaround for precisely this circumstance. It provides for an alternative to a 2-channel 24/705.6 PCM format using a 4-channel 24/352.8 PCM format. The 8-bit DoP marker specified is different, which enables the DAC to tell 4-channel DSD128 from 2-channel DSD256 (they would otherwise be indistinguishable). Very few DAC manufacturers currently support this variant format. Mytek is the only one I know of - as I understand it their 192-DSD DAC supports DSD128 using the standard 2-channel DoP over USB, but using the 4-channel variant DoP over FireWire.
Because of its negligible adoption rate, BitPerfect currently does not support the 4-channel DoP variant. If we did, it would require some additional configuration options in the DSD Support window. I worry that such options are bound to end up confusing people. For example, despite what our user manual says, you would not believe the number of customers who write to me because they have checked the “dCS DoP” checkbox and wonder why DSD playback isn’t working! Maybe they were hoping it would make their DACs sound like a dCS, I dunno. I can only imagine what they will make of a 2ch/4ch configurator!!!
As a final observation, some playback software will on-the-fly convert high-order DSD formats which are not supported by the user’s DAC to a lower-order DSD format which is. While this is a noble solution, it should be noted that format conversion in DSD is a fundamentally lossy process, and that all of the benefits of the higher-order DSD format - and more - will as a result be lost. In particular, the ultrasonic noise profile will be that of the output DSD format, not that of the source DSD format. Additionally, DSD bitstreams are created by Sigma-Delta Modulators. These are complex and very challenging algorithms which are seriously hard to design and implement successfully, particularly if you want anything beyond modest performance out of them. The FPGA-based implementation developed for the PS Audio DirectStream DAC is an example of a good one, but there are some less-praiseworthy efforts out there. In general, you can expect to obtain audibly superior results pre-converting instead to 24/176.4 (or even 24/352.8) PCM using DSD Master, which will retain both the extended frequency response and the lower ultrasonic noise floor of the DSD256 original.
Thursday, 1 January 2015
A Convoluted Discussion
In mathematics, the word ‘convolution’ describes a very important class of manipulations. If you want to know more about it, a pretty good treatment is shown on its Wikipedia page. And even if you don’t, I am going to briefly summarize it here before going on to make my point :)
A convolution is an operation performed on two functions, or on two sets of data. Typically (but not always) one is the actual data that we are trying to manipulate, and the other is a weighting function, or set of weights. Convolution is massively important in the field of signal processing, and therefore is something that anybody who wants (or needs) to talk knowledgeably about digital audio needs to bone up on. The most prominent convolution processes that you may have heard of are Fourier Transforms (which are used to extract from a waveform its audio spectrum) and digital filtering. It is the latter of those that I want to focus on here.
In very simple terms, a filter (whether digital or analog) operates as a convolution between a waveform and an impulse response. You will have heard of impulse responses, and indeed you may have read about them in some of my previous posts. In digital audio, an impulse response is a graphical representation of the ‘weights’ or ‘coefficients’ which define a digital filter. Complicated mathematical relationships describe the way in which the impulse response relates to the key characteristics of the filter, and I have covered those in my earlier posts on ‘Pole Dancing’.
Impulse responses are therefore very useful. They are nice to look at, and easy to categorize and classify. Unfortunately, it has become commonplace to project the aesthetic properties of the impulse response onto the sonic properties arising from the filter which uses it. In simple language, we see a feature on the impulse response, and we imagine that such a feature is impressed onto the audio waveform itself after it comes out of the filter. It is an easy mistake to make, since the convolution process itself is exactly that - a mathematical impression of the impulse response onto the audio waveform. But the mathematical result of the convolution is really not as simple as that.
The one feature I see misrepresented most often is pre-ringing. In digital audio, an impulse is just one peak occurring in a valley of flat zeros. It is useful as a tool to characterize a filter because it contains components of every frequency that the music bit stream is capable of representing. Therefore if the filter does anything at all, the impulse is going to be disturbed as a result of passing through it. For example, if you read my posts on square waves, you will know that removing high frequency components from a square wave results in a waveform which is no longer square, and contains ripples. Those ripples decay away from the leading edge of the square wave. This is pleasing in a certain way, because the ripples appear to be caused by, and arise in response to, the abrupt leading edge of the square wave. In our nice ordered world we like to see effect preceded by cause, and are disturbed by suggestions of the opposite.
And so it is that with impulse responses we tend to be more comfortable seeing ripples decaying away after the impulse, and less comfortable when they precede the impulse, gathering in strength as they approach it. Our flawed interpretation is that the impulse is the cause and the ripples the effect, and if these don’t occur in the correct sequence then the result is bound to be unnatural. It is therefore common practice to dismiss filters whose impulse response contains what is termed “pre-ringing” because the result of such filters is bound to be somewhat “unnatural”. After all, in nature, effects don’t precede their cause, do they?
I would like you to take a short break, and head over to your kitchen sink for a moment. Turn on the tap (or faucet, if you prefer) and set the water flow to a very gentle stream. What we are looking for is a smooth flow with no turbulence at all. We call this ‘laminar’ flow. What usually happens, if the tap outlet is sufficiently far above the bottom of the sink is that the laminar flow is maintained for some distance and then breaks up into a turbulent flow. The chances are good that you will see this happening, but it is no problem if you don’t - so long as you can find a setting that gives you a stable laminar stream. Now, take your finger, and gently insert it into the water stream. Look closely. What you will see are ripples forming in the water stream **above** your finger. If you don’t, gradually move your finger up towards the tap and they should appear (YTMV/YFMV). What you will be looking at is an apparently perfect example of an effect (the ripples) occurring before, or upstream of, the cause (your finger).
What I have demonstrated here is not your comfortable world breaking down before your eyes. What is instead breaking down is the comfort zone of an over-simplistic interpretation of what you saw. Because the idea of the finger being the cause and the ripples being the effect is not an adequate description of what actually happened.
In the same way, the notion of pre-ringing in the impulse response of a filter resulting in sonic effects that precede their cause in the resultant audio waveform, is not an adequate description of what is happening. However, the misconception gains credence for an important, if inconvenient reason, which is that filters which exhibit pronounced pre-ringing do in fact tend to sound less preferable than those which don’t. These sort of things happen often in science - most notably in medical science - and when it does it opens a door to misinformation. In this case, the potential for misinformation lies in the reason given for why one filter sounds better than another - that the one with pre-ringing in its impulse response results in sounds that precede the things which caused them. By all means state your preference for filters with a certain type of impulse response, but please don’t justify your preference with flawed reasoning. It is OK to admit that you are unclear as to why.
I want to finish this with an audio example to make my point. The well-known Nyquist-Shannon theory states that a regularly sampled waveform can be perfectly recreated if (i) it is perfectly sampled; and (ii) it contains no frequency components at or above one half of the sample rate. The theory doesn’t just set forth its premise, it provides a solid proof. In essence, it does this by convolving the sampled waveform with a Sinc() function, in a process pretty much identical to the way a digital filter convolves the waveform with an Impulse Response. Nyquist-Shannon proves that this convolution results in a mathematically perfect reconstruction of the original waveform if - and only if - the two stipulations I mentioned are strictly adhered to. This is interesting in the context of this post because the Sinc() function which acts as the Impulse Response exhibits an infinitely long pre-ring at a significantly high amplitude. Neither Nyqvist or Shannon, nor the entire industry which their theory spawned, harbour any concerns about causality in reconstructed waveforms!
A convolution is an operation performed on two functions, or on two sets of data. Typically (but not always) one is the actual data that we are trying to manipulate, and the other is a weighting function, or set of weights. Convolution is massively important in the field of signal processing, and therefore is something that anybody who wants (or needs) to talk knowledgeably about digital audio needs to bone up on. The most prominent convolution processes that you may have heard of are Fourier Transforms (which are used to extract from a waveform its audio spectrum) and digital filtering. It is the latter of those that I want to focus on here.
In very simple terms, a filter (whether digital or analog) operates as a convolution between a waveform and an impulse response. You will have heard of impulse responses, and indeed you may have read about them in some of my previous posts. In digital audio, an impulse response is a graphical representation of the ‘weights’ or ‘coefficients’ which define a digital filter. Complicated mathematical relationships describe the way in which the impulse response relates to the key characteristics of the filter, and I have covered those in my earlier posts on ‘Pole Dancing’.
Impulse responses are therefore very useful. They are nice to look at, and easy to categorize and classify. Unfortunately, it has become commonplace to project the aesthetic properties of the impulse response onto the sonic properties arising from the filter which uses it. In simple language, we see a feature on the impulse response, and we imagine that such a feature is impressed onto the audio waveform itself after it comes out of the filter. It is an easy mistake to make, since the convolution process itself is exactly that - a mathematical impression of the impulse response onto the audio waveform. But the mathematical result of the convolution is really not as simple as that.
The one feature I see misrepresented most often is pre-ringing. In digital audio, an impulse is just one peak occurring in a valley of flat zeros. It is useful as a tool to characterize a filter because it contains components of every frequency that the music bit stream is capable of representing. Therefore if the filter does anything at all, the impulse is going to be disturbed as a result of passing through it. For example, if you read my posts on square waves, you will know that removing high frequency components from a square wave results in a waveform which is no longer square, and contains ripples. Those ripples decay away from the leading edge of the square wave. This is pleasing in a certain way, because the ripples appear to be caused by, and arise in response to, the abrupt leading edge of the square wave. In our nice ordered world we like to see effect preceded by cause, and are disturbed by suggestions of the opposite.
And so it is that with impulse responses we tend to be more comfortable seeing ripples decaying away after the impulse, and less comfortable when they precede the impulse, gathering in strength as they approach it. Our flawed interpretation is that the impulse is the cause and the ripples the effect, and if these don’t occur in the correct sequence then the result is bound to be unnatural. It is therefore common practice to dismiss filters whose impulse response contains what is termed “pre-ringing” because the result of such filters is bound to be somewhat “unnatural”. After all, in nature, effects don’t precede their cause, do they?
I would like you to take a short break, and head over to your kitchen sink for a moment. Turn on the tap (or faucet, if you prefer) and set the water flow to a very gentle stream. What we are looking for is a smooth flow with no turbulence at all. We call this ‘laminar’ flow. What usually happens, if the tap outlet is sufficiently far above the bottom of the sink is that the laminar flow is maintained for some distance and then breaks up into a turbulent flow. The chances are good that you will see this happening, but it is no problem if you don’t - so long as you can find a setting that gives you a stable laminar stream. Now, take your finger, and gently insert it into the water stream. Look closely. What you will see are ripples forming in the water stream **above** your finger. If you don’t, gradually move your finger up towards the tap and they should appear (YTMV/YFMV). What you will be looking at is an apparently perfect example of an effect (the ripples) occurring before, or upstream of, the cause (your finger).
What I have demonstrated here is not your comfortable world breaking down before your eyes. What is instead breaking down is the comfort zone of an over-simplistic interpretation of what you saw. Because the idea of the finger being the cause and the ripples being the effect is not an adequate description of what actually happened.
In the same way, the notion of pre-ringing in the impulse response of a filter resulting in sonic effects that precede their cause in the resultant audio waveform, is not an adequate description of what is happening. However, the misconception gains credence for an important, if inconvenient reason, which is that filters which exhibit pronounced pre-ringing do in fact tend to sound less preferable than those which don’t. These sort of things happen often in science - most notably in medical science - and when it does it opens a door to misinformation. In this case, the potential for misinformation lies in the reason given for why one filter sounds better than another - that the one with pre-ringing in its impulse response results in sounds that precede the things which caused them. By all means state your preference for filters with a certain type of impulse response, but please don’t justify your preference with flawed reasoning. It is OK to admit that you are unclear as to why.
I want to finish this with an audio example to make my point. The well-known Nyquist-Shannon theory states that a regularly sampled waveform can be perfectly recreated if (i) it is perfectly sampled; and (ii) it contains no frequency components at or above one half of the sample rate. The theory doesn’t just set forth its premise, it provides a solid proof. In essence, it does this by convolving the sampled waveform with a Sinc() function, in a process pretty much identical to the way a digital filter convolves the waveform with an Impulse Response. Nyquist-Shannon proves that this convolution results in a mathematically perfect reconstruction of the original waveform if - and only if - the two stipulations I mentioned are strictly adhered to. This is interesting in the context of this post because the Sinc() function which acts as the Impulse Response exhibits an infinitely long pre-ring at a significantly high amplitude. Neither Nyqvist or Shannon, nor the entire industry which their theory spawned, harbour any concerns about causality in reconstructed waveforms!
Wednesday, 31 December 2014
Why are $20k loudspeakers so expensive?
Here is a short video that should give pause to those who have asked
that question with the confident skepticism of someone who has never
tried to actually make a pair themselves. This person has made
his own pair of B&W 800 Diamond loudspeakers. Has he succeeded?
We will never know, but it sure looks most impressive.
In practice, he has restricted himself to making his own set of elaborate cabinets, as it looks as though he has bought all the drive units from B&W. But even so, the overwhelming impression is of the expensive resources he has had to bring to bear to realize the project. OK, he has done the grunt work himself, but the project has clearly taken a HUGE amount of time and effort. Aside from some initial consternation, I imagine that the executives at B&W are having a good chuckle over it.
Presumably his motivation was purely the satisfaction of creating his own work of art. Think about it. How much money can he possibly have saved by doing it himself? Do you think you could do it yourself for less, without sacrificing at least some of the core design objectives?
Whatever, as I contemplate my own B&W 802 Diamonds, I am sure glad I bought mine!
https://www.youtube.com/watch?v=fHgdNQkiNVk
In practice, he has restricted himself to making his own set of elaborate cabinets, as it looks as though he has bought all the drive units from B&W. But even so, the overwhelming impression is of the expensive resources he has had to bring to bear to realize the project. OK, he has done the grunt work himself, but the project has clearly taken a HUGE amount of time and effort. Aside from some initial consternation, I imagine that the executives at B&W are having a good chuckle over it.
Presumably his motivation was purely the satisfaction of creating his own work of art. Think about it. How much money can he possibly have saved by doing it himself? Do you think you could do it yourself for less, without sacrificing at least some of the core design objectives?
Whatever, as I contemplate my own B&W 802 Diamonds, I am sure glad I bought mine!
https://www.youtube.com/watch?v=fHgdNQkiNVk
Monday, 15 December 2014
DSD - Is DST Compression Really Lossless?
The SACD format was built around DSD right from the start. Since DSD takes up about four times the amount of disk space of a 16/44.1 equivalent this meant that a new physical disc format with more capacity than a CD was going to be required. Additionally, SACD was specified to deliver multi-channel content, which increases the storage requirement by another factor of 3 or more, depending on how many channels you want to support. The only high-capacity disc format that was on the horizon at the time was the one eventually used for DVD, and even this was going to be inadequate for the full multi-channel capability envisaged for SACD.
The solution was to adopt a lossless data compression protocol to reduce the size of a multi-channel DSD master file so that it would fit. The protocol chosen is called DST, and is an elaborate DSP-based method derived from the way MP3 works. Essentially, you store a bunch of numbers that represent the actual data as a mathematical function which you can later use to try to re-create the original data. You then store a bunch of additional numbers which represent the differences between the actual data and the attempted recreation. If you do this properly, the mathematical function numbers, plus the difference data, takes up less space than the original data. On a SACD the compression achieved is about 50%, which is pretty good, and permits a lot of content to be stored.
Given that DST compression is lossless, it is interesting that the SACD format allows discs to be mastered with your choice of compressed or non-compressed data. And, taking a good look at a significant sample of SACDs, it appears that a substantial proportion of those discs do not use compression. Additionally, if you look closely, you will see that almost all of the serious audiophile remasters released on SACD are all uncompressed. So the question I have been asking is - is there any reason to believe that DST-compressed SACDs might sound worse than uncompressed ones?
First of all, let me be clear on one thing. The DST compression algorithm is lossless. This means that the reconstructed bit stream is bit-for-bit identical to the original uncompressed bit stream. This is not at issue here. Nor is the notion that compressing and decompressing the bits somehow stresses them so that they don’t sound so relaxed on playback. I don’t buy mumbo jumbo. The real answer is both simpler than you would imagine (although technically quite complicated), and at the same time typical of an industry which has been known to upsample CD content and sell it for twice the price on a SACD disc.
To understand this, we need to take a closer look at how the DSD format works. I have written at length about how DSD makes use of a combination of massive oversampling and noise shaping to encode a complex waveform in a 1-bit format. In a Sigma-Delta Modulator (SDM) the quantization noise is pushed out of the audio band and up into the vast reaches of the ultrasonic bandwidth which dominates the DSD encoding space. The audio signal only occupies the frequency space below 20kHz (to choose a number that most people will agree on). But DSD is sampled at 2,822kHz, so there is a vast amount of bandwidth between 20kHz and 2,822kHz available, into which the quantization noise can be swept.
One of the key attributes of a good clean audio signal is that it have low noise in the audio band. In general, the higher quality the audio signal, the lower the noise it will exhibit. The best microphones can capture sounds that cannot be fully encoded using 16-bit PCM. However, 24-bit PCM can capture anything that the best microphones will put out. Therefore if DSD is to deliver the very highest in audio performance standards it needs to be able to sustain a noise floor better than that of 16-bit audio, and approaching that of 24-bit audio.
The term “Noise Shaping” is a good one. Because quantization noise cannot be eliminated, all you can hope to do is to take it from one frequency band where you don’t want it, and move it into another where you don’t mind it - and in the 1-bit world of DSD there is an awful lot of quantization noise. This is the job of an SDM. The design of the SDM determines how much noise is removed from the audio frequency band, and where it gets put. Mathematically, DSD is capable of encoding a staggeringly low noise floor in the audio band. Something down in the region of -180dB to -200dB has been demonstrated. What good DSD recordings achieve is nearer to -120dB, and the difference is partly due to that fact that practical real-world SDM designs seriously underperform their theoretical capabilities. But it also arises because better performance requires a higher-order SDM design, and beyond a certain limit high-order SDMs are simply unstable. A workmanlike SDM would be a 5th-order design, but the best performance today is achieved with 8th or 9th order SDMs. Higher than that, and they cannot be made to work.
So how does a higher-order SDM achieve superior performance? The answer is that it packs more and more of the quantization noise into the upper reaches of the ultrasonic frequency space. So a higher-performance higher-order SDM will tend to encode progressively more high-frequency noise into the bitstream. A theoretically perfect SDM will create a Bit Stream whose high frequency content is virtually indistinguishable from full-scale white noise.
This is where DST compression comes in. Recall that DST compression works by storing a set of numbers that enable you to reconstruct a close approximation of the original data, plus all of the differences between the reconstructed bit stream and the original bit stream. Obviously the size of the compressed (DST-encoded) file will be governed to a large degree by how much data is needed to store the difference signal. It turns out that the set of numbers that reconstruct the ‘close approximation’ do a relatively good job of encoding the low frequency data, but a relatively poor job of encoding the high frequency data. Therefore, the more high frequency data is present, the more additional data will be needed to encode the difference signal. And the larger the difference signal, the larger the compressed file will be. In the extreme, the difference signal can be so large that you will not be able to achieve much compression at all.
This is the situation we are in with today’s technology. We can produce the highest quality DSD signal and be unable to compress it effectively, or we can accept a reduction in quality and achieve a useful degree of (lossless) compression.
So what happens when we have a nice high-resolution DSD recording all ready to be sent to the SACD mastering plant? What happens if the DSD content is too large to fit onto a SACD, and cannot be compressed enough so that it does? The answer will disappoint you. What happens is that the high quality DSD master tape is remodulated using a modest 5th-order SDM, in the process producing a new DSD version which can now be efficiently compressed using DST compression. Most listeners agree that a 5th order SDM produces audibly inferior sound to a good 8th order SDM, but with real music recordings it is essentially impossible to inspect a DSD data file and determine unambiguously what order of SDM was used to encode it. So it is easy enough to get away with.
How do you tell if a SACD is compressed or not? Well, if you have the underground tools necessary, you can rip it and analyze it definitively. For the rest of us there is no sure method except for one. You simply add up the total duration of the music on the disc, and calculate 2,822,400 bits of data per second, per channel. If the answer amounts to more than 4.7GB then the data must be compressed. If it adds up to less, there is no guarantee that it won’t be DST-compressed, but the chances are pretty good that it is not. After all, if the record company wants to compress it, they’d have to pay someone to do that, and that probably ain’t gonna happen. The other simple guideline is that if it is multi-channel it is probably compressed, but if it is stereo it probably is not.
Of course, none of this need apply to downloaded DSD files. If produced by reputable studios these will have been produced using the best quality modulators they can afford, and since DST encoding is not used on commercial DSF and DFF* files this whole issue need not arise. However, if the downloaded files are derived from a SACD (as many files are which are not distributed by the original producers), then the possibility does exist that you are receiving an inferior 5th-order remodulated version. The take-away is that not all DSD is created equal. Yet another thing for us to have to bear in mind!
[* Actually, the DFF file format does allow for the DSD content to be DST compressed, because the DFF file format is what is used by the mastering house to provide the final distribution-ready content to the SACD disc manufacturing plant. However, for consumer DSD file distribution, I don’t think anybody employs DST compression.]
The solution was to adopt a lossless data compression protocol to reduce the size of a multi-channel DSD master file so that it would fit. The protocol chosen is called DST, and is an elaborate DSP-based method derived from the way MP3 works. Essentially, you store a bunch of numbers that represent the actual data as a mathematical function which you can later use to try to re-create the original data. You then store a bunch of additional numbers which represent the differences between the actual data and the attempted recreation. If you do this properly, the mathematical function numbers, plus the difference data, takes up less space than the original data. On a SACD the compression achieved is about 50%, which is pretty good, and permits a lot of content to be stored.
Given that DST compression is lossless, it is interesting that the SACD format allows discs to be mastered with your choice of compressed or non-compressed data. And, taking a good look at a significant sample of SACDs, it appears that a substantial proportion of those discs do not use compression. Additionally, if you look closely, you will see that almost all of the serious audiophile remasters released on SACD are all uncompressed. So the question I have been asking is - is there any reason to believe that DST-compressed SACDs might sound worse than uncompressed ones?
First of all, let me be clear on one thing. The DST compression algorithm is lossless. This means that the reconstructed bit stream is bit-for-bit identical to the original uncompressed bit stream. This is not at issue here. Nor is the notion that compressing and decompressing the bits somehow stresses them so that they don’t sound so relaxed on playback. I don’t buy mumbo jumbo. The real answer is both simpler than you would imagine (although technically quite complicated), and at the same time typical of an industry which has been known to upsample CD content and sell it for twice the price on a SACD disc.
To understand this, we need to take a closer look at how the DSD format works. I have written at length about how DSD makes use of a combination of massive oversampling and noise shaping to encode a complex waveform in a 1-bit format. In a Sigma-Delta Modulator (SDM) the quantization noise is pushed out of the audio band and up into the vast reaches of the ultrasonic bandwidth which dominates the DSD encoding space. The audio signal only occupies the frequency space below 20kHz (to choose a number that most people will agree on). But DSD is sampled at 2,822kHz, so there is a vast amount of bandwidth between 20kHz and 2,822kHz available, into which the quantization noise can be swept.
One of the key attributes of a good clean audio signal is that it have low noise in the audio band. In general, the higher quality the audio signal, the lower the noise it will exhibit. The best microphones can capture sounds that cannot be fully encoded using 16-bit PCM. However, 24-bit PCM can capture anything that the best microphones will put out. Therefore if DSD is to deliver the very highest in audio performance standards it needs to be able to sustain a noise floor better than that of 16-bit audio, and approaching that of 24-bit audio.
The term “Noise Shaping” is a good one. Because quantization noise cannot be eliminated, all you can hope to do is to take it from one frequency band where you don’t want it, and move it into another where you don’t mind it - and in the 1-bit world of DSD there is an awful lot of quantization noise. This is the job of an SDM. The design of the SDM determines how much noise is removed from the audio frequency band, and where it gets put. Mathematically, DSD is capable of encoding a staggeringly low noise floor in the audio band. Something down in the region of -180dB to -200dB has been demonstrated. What good DSD recordings achieve is nearer to -120dB, and the difference is partly due to that fact that practical real-world SDM designs seriously underperform their theoretical capabilities. But it also arises because better performance requires a higher-order SDM design, and beyond a certain limit high-order SDMs are simply unstable. A workmanlike SDM would be a 5th-order design, but the best performance today is achieved with 8th or 9th order SDMs. Higher than that, and they cannot be made to work.
So how does a higher-order SDM achieve superior performance? The answer is that it packs more and more of the quantization noise into the upper reaches of the ultrasonic frequency space. So a higher-performance higher-order SDM will tend to encode progressively more high-frequency noise into the bitstream. A theoretically perfect SDM will create a Bit Stream whose high frequency content is virtually indistinguishable from full-scale white noise.
This is where DST compression comes in. Recall that DST compression works by storing a set of numbers that enable you to reconstruct a close approximation of the original data, plus all of the differences between the reconstructed bit stream and the original bit stream. Obviously the size of the compressed (DST-encoded) file will be governed to a large degree by how much data is needed to store the difference signal. It turns out that the set of numbers that reconstruct the ‘close approximation’ do a relatively good job of encoding the low frequency data, but a relatively poor job of encoding the high frequency data. Therefore, the more high frequency data is present, the more additional data will be needed to encode the difference signal. And the larger the difference signal, the larger the compressed file will be. In the extreme, the difference signal can be so large that you will not be able to achieve much compression at all.
This is the situation we are in with today’s technology. We can produce the highest quality DSD signal and be unable to compress it effectively, or we can accept a reduction in quality and achieve a useful degree of (lossless) compression.
So what happens when we have a nice high-resolution DSD recording all ready to be sent to the SACD mastering plant? What happens if the DSD content is too large to fit onto a SACD, and cannot be compressed enough so that it does? The answer will disappoint you. What happens is that the high quality DSD master tape is remodulated using a modest 5th-order SDM, in the process producing a new DSD version which can now be efficiently compressed using DST compression. Most listeners agree that a 5th order SDM produces audibly inferior sound to a good 8th order SDM, but with real music recordings it is essentially impossible to inspect a DSD data file and determine unambiguously what order of SDM was used to encode it. So it is easy enough to get away with.
How do you tell if a SACD is compressed or not? Well, if you have the underground tools necessary, you can rip it and analyze it definitively. For the rest of us there is no sure method except for one. You simply add up the total duration of the music on the disc, and calculate 2,822,400 bits of data per second, per channel. If the answer amounts to more than 4.7GB then the data must be compressed. If it adds up to less, there is no guarantee that it won’t be DST-compressed, but the chances are pretty good that it is not. After all, if the record company wants to compress it, they’d have to pay someone to do that, and that probably ain’t gonna happen. The other simple guideline is that if it is multi-channel it is probably compressed, but if it is stereo it probably is not.
Of course, none of this need apply to downloaded DSD files. If produced by reputable studios these will have been produced using the best quality modulators they can afford, and since DST encoding is not used on commercial DSF and DFF* files this whole issue need not arise. However, if the downloaded files are derived from a SACD (as many files are which are not distributed by the original producers), then the possibility does exist that you are receiving an inferior 5th-order remodulated version. The take-away is that not all DSD is created equal. Yet another thing for us to have to bear in mind!
[* Actually, the DFF file format does allow for the DSD content to be DST compressed, because the DFF file format is what is used by the mastering house to provide the final distribution-ready content to the SACD disc manufacturing plant. However, for consumer DSD file distribution, I don’t think anybody employs DST compression.]
Saturday, 13 December 2014
BitPerfect 2.0.2 Released
Today, BitPerfect 2.0.2 has been released to the App Store. It may take up to 48hrs before it shows up in all regions. V2.0.2 contains several minor bug fixes, plus some minor enhancements to the audio engine to improve stability.
BitPerfect 2.0.2 is a free upgrade for existing BitPerfect users.
BitPerfect 2.0.2 is a free upgrade for existing BitPerfect users.
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