As digital audio and the means of playing it back mature, there is an increasing divergence of perspectives to be found on the Internet. Some revel in the sonics of music heard at high resolution, while others argue that the CD standard is not to be audibly improved upon and still others want even higher resolution. All this while Joe and Jane Average download one song at a time at resolutions that throw away at least 75% of the information contained on a CD.
There are new efforts from some quarters to show Joe and Jane what they’re missing and to elevate what the download services offer. The idea is to, at the very least, deliver 100% of what the CD offers and at best, deliver true high resolution. Yet these efforts have spawned Internet “papers” and articles in effect, ridiculing the very idea of high resolution and arguing the supposed inaudibility of its benefits, or worse, suggesting that high resolution by definition will sound worse, not better.
I can’t speak for what others find but I can say that whatever these folks are reporting is quite the opposite of what I experience. I’m hearing fidelity such as I’ve dreamed about for years and when I read those stories, they strike me much as though the authors are trying to convince me there are no colors in a rainbow.
The arrival of high resolution digital has the potential to fulfill the promise digital audio first made more than a quarter century ago. Back then, astute listeners wondered at the marketing mantra “perfect sound forever” while cringing at the dry, bleached and airless sounds delivered by the first CD players. While a great deal of progress has been made during the intervening years, the inherent limitations of the format remain.
Looked at in the most rudimentary fashion, the specifications for CD would, on the surface, appear to be all that is needed to perfectly reproduce anything that can be heard. Human hearing is nominally sensitive to frequencies from 20 Hz through 20 kHz (i.e., 20 cycles per second through 20,000 cycles per second). As we age, the top end limit decreases and most adults would be lucky to hear 15 kHz. With CD, music is sampled 44,100 times per second. That is, the digital recorder “looks at” the sound 44,100 times every second and captures a sample. According to the theory, all frequencies below half the sample rate, (in this case, all frequencies below 22,050 cycles per second) will be captured accurately and since this is well beyond what most folks can hear, it all sounds quite neat.
These digital samples are each a series of digital bits, with each bit representing one of two binary states or values, often thought of as “ones and zeros”. Each sample is stored in a digital word. The CD standard uses 16-bit words, where each sample contains 16 values. The particular combination of ones and zeros represents the level (i.e., volume) of each sample. A series of 16 zeros (i.e., 0000000000000000) would be the lowest level that can be encoded and represents complete silence. A 16-bit word representing an intermediate level might look like this: 0111011110101110. The highest level would be 0111111111111111, a zero followed by 15 ones. (For technical reasons which are beyond the scope of this entry, the loudest value is not a series of 16 ones.)
A word length of 16-bits allows up to 65,536 different levels to be represented. The difference between the loudest sound that can be captured and the noise floor of the format is called the signal-to-noise ratio. Signal-to-noise ratio is measured in units of loudness called decibels (dB). For a 16-bit format like CD, the signal-to-noise ratio is approximately 96 dB, which means the noise floor (the inherent noise of the format) is 96 decibels below the loudest sound that can be captured. This is much quieter than vinyl or analog tape. Any hiss heard on a CD is captured from the source and is not inherent in the medium. Many folks confuse the signal-to-noise ratio specification with dynamic range (the difference in level between the loudest possible sound and the lowest sound). We’ll come back to this later and see why this is misguided.
The problems start when we move from the theoretical to the practical. (Someone, perhaps it was Yogi Berra, once said “In theory, there is no difference between theory and practice, but in practice, there is.”) When digital audio is recorded, any frequencies above half the sample rate can cause problems – they engender aliases or aliasing distortion, false frequencies that are not part of the program material. In order to avoid aliasing, when digital audio is encoded, as well as when it is played back, most digital processors use a filter to ensure that no frequencies above half the sample rate can pass. These anti-aliasing filters have audible side effects, manifesting in the time domain – the signal gets smeared in time. Some designers will use gentler filters to minimize the time smear but in doing so, they cause the higher frequencies to fall off prematurely. A number of modern playback devices have user-selectable filters where the listener can select between steep filtering and its associated time issues or gentler filtering and its associated frequency issues.
So, while CD can capture all the audible frequency range, the requisite filtering means the frequencies delivered to the listener are not all arriving on time or are not all arriving in the same proportion in which they were captured, or some combination of both of these. One great advantage of the higher sample rates is that the anti-aliasing filter is moved far above the audible range. This allows gentler filtering to be used without affecting the audible frequency range.
In recent years, thanks in no small part to formats like DVD and others, which are capable of storing more information than will fit on a CD, digital audio has grown up from the 16-bit words and 44.1 kHz sample rates by which sound is encoded for CD. We’ve had 24-bit audio with sample rates of 96 kHz, 176.4 kHz and 192 kHz. For reference, a 24/96 (24-bit, 96 kHz) version of a given recording contains more than three times the information contained in the same recording at 16/44 (16-bit, 44.1 kHz). A 24/192 version contains more than six times the information. And where a word length of 16-bits allows up to 65,536 different levels to be represented, going to 24-bits increases the dynamic resolution 256 times, allowing up to 16,777,216 different levels to be represented.
The widespread use of computers (and computing devices) for audio playback has enabled the proliferation of high resolution audio and emancipated music from the confines of silver discs and the limitations imposed by the process of retrieving music from these in real time. (Separating the processing “overhead” from the playback will provide higher quality playback.) Good as the best disc players and transports can be, my experience has been that there is invariably a loss of focus and fine detail, often subtle, sometimes not so subtle. It is only via proper computer playback that I’ve heard results that I find indistinguishable from listening to the master used to create those silver discs.
This is good news, even for music at CD resolution, because the listener at home can now hear what is effectively the CD master itself. However, while the limitations of playback from molded disc have been removed, the limitations of the format remain. In addition to the frequency and time-related issues brought about by having the anti-aliasing filter just above the audible range, there are the consequences of inadequate word length. Although the noise floor with a 16-bit medium like CD is 96 dB below the loudest possible sound that can be captured by the format, many often confuse this signal-to-noise ratio with dynamic range. The assumption is that if the noise floor is 96 dB below the loudest sound, sounds just above the noise floor will be captured with the same fidelity, providing a range of dynamics as wide as the signal-to-noise ratio. In fact, with a 16-bit medium, the fidelity plummets at lower levels.
The full resolution, in this case 16-bits, is only realized for sounds near the top of the volume range. Each bit captures about 6 dB of the dynamic range (about 6.02 dB to be more precise but let’s use 6 in this example to keep things simple), so in a 16-bit system, sounds lower in level than 6 dB below the maximum will effectively be captured at less than 16-bit resolution. To wit, if this lower level information is say, 12 dB lower in level, it will be encoded at what is effectively approximately 2 bits less than the full resolution of the format (i.e., 14 bits in a 16-bit recording, 22 bits in a 24-bit recording). If it is say, 36 dB lower in level, it will be encoded at what is effectively approximately 6 bits less resolution (i.e., 10 bits in a 16-bit recording, 18 bits in a 24-bit recording).
Some information, such as the trailing end of reverb as it fades away, or the higher harmonics of musical instruments, can be well more than that 36 dB lower in level than the loudest sounds and will be encoded with resolutions corresponding to fewer bits. This results in the thinned, bleached and coarsened instrumental harmonics in even the best 16-bit recordings, as compared to a good 24-bit recording (or of course, the original sound in real life). It also results in the defocusing of the spatial information and in the relative airlessness in the 16-bit recording compared to a good 24-bit recording (and real life).
While the level meter may show a peak on that 16-bit recording that is within the top 6 dB, this, like the waveform views shown by some computer software, is only a view of the “top” part of the musical waveform — the loudest part. Sounds and components of sounds that are underneath the top part (i.e., in the background) are not captured as faithfully. Accordingly, when considering the dynamic range of the format, it is a good idea to take into account the relative distortion at different levels within the range. If increasing distortion is not desirable, the real dynamic range potential is going to be considerably less than what the spec sheet might suggest (or is often echoed in the audio press and in some places on the Internet). Note that even with low level information as in the examples above, a 24-bit recording still delivers more resolution than a 16-bit recording at its best.
Why then, would someone publish a “white paper” against higher resolution or declare that resolutions like 24/192 are “pointless” or worse? A few possible reasons come to mind:
- The higher sample rates place significantly increased demands on the gear used to record and play them back. For example, digital gear contains an internal clock to control the timing as the device encodes or decodes the stream of digital samples. Spacing between the samples must be kept accurate or the reconstructed analog waveform that we hear will not have the correct shape and hence, will not provide the correct sound. Irregularities in timing are referred to as jitter. Higher sample rates also mean the analog stages of the gear must be able to perform at the wider bandwidths. Perhaps the folks complaining about high resolution are using gear that does not have clocking that is up to the task and analog stages that can perform at high bandwidth. Such will either not reveal any benefits or will actually sound worse than they do at the easier, lower rates like 24/96. (This is true of a number of “professional” units as well as those sold to audio enthusiasts. A built-in, $250 “soundcard” simply won’t do it, regardless of what the specs claim. In today’s market, it may cost 10 times this amount for a device truly capable of revealing the potential of these sample rates. Maybe it is no wonder these folks hear little or no difference.)
- It could be possible that the rest of the system these folks are using isn’t up to resolving a wide band recording. Or it could be that these folks are just not sensitive to these particular differences. I’ve always found that different folks have different sensitivities to different aspects of sound.
- Perhaps they believe CDs (or 24/96) already sound identical to the input signal. If that is the case, I can understand that anything more would seem wasteful.
Sample rates like 176.4k and 192k don’t, as some have erroneously suggested “have more jitter”. Sample rates don’t have jitter. As stated above, higher sampling rates do place greater demands on clocking accuracy (just one reason why buying a DAC
(digital-to-analog converter) “by the chip” is at best a foolish enterprise). They also place greater demands on the analog stages surrounding the digital stage.
Why some would see these characteristics as “flaws” (and write papers or articles on the subject), I don’t understand. I’ve always gone with empirical evidence over theoretical analysis; that is, when “theory” and direct experience are at odds with each other, I’ll tend to seek a new theory. (As I see it, theory should explain the experience, not the other way around.)
All this to say, when a firmware upgrade enabled 192k capability in the converters I use for my work, I approached it conservatively — even continuing to do a few recording sessions at 96k because I was familiar with it and could be confident in the results. But then I started running tests at 192k and quite quickly found I had to get my jaw up off the floor: for the very first time in my experience, I was hearing (with this device anyway) a recording device “disappear”. I had never heard that before, even with the best analog recorders and most certainly nothing close with the best digital recorders, even with this very device when used at 96k.
Now I felt a threshold had been crossed (I’ve read similar words since then from one of my favorite audio engineers, Keith Johnson). The results no longer sounded like “great digital”; they no longer sounded “digital” at all. They didn’t sound like “great analog” either. The jump from 24/96 to 24/192, when done well, is to my ears a much more significant jump than the one from 16/44 to 24/96. It’s all about that threshold; this is the promise digital made in 1983, finally and for real. (While it certainly sounds more faithful to the input signal than 16/44 does, 24/96 doesn’t yet, to my ears, “get out of the way”. Having the anti-aliasing filter moved well up and away from the audible range definitely helps but it is the rates like 176.4k and 192k where I find the threshold is crossed. Interestingly, to me, while many speak of the treble response, I find some of the greatest benefits to be in how much more lifelike I find the bass.) 24/192 pointless?!? Only if real progress in music reproduction is pointless.
While some decry the higher sample rates, either declaring them to offer no audible sonic differences from the lower ones or to offer inferior sonics (!) to the lower ones, other voices are talking even higher numbers. We are seeing marketers talking 32-bits and sample rates like 384 kHz. As long as there are customers who are taken in by sheer numbers, there will be those who see an opportunity for commerce, who will accommodate them with sheer numbers.
I get a chuckle out of these things because my experience has been that in reality (that is, with the hardware or software one can go out and buy today, as well as the recordings one can purchase to play on these), I see gear that isn’t particularly clean at 24-bits. I see other gear which, when presented with 4x sample rates (176.4k or 192k), performs worse than it does at 2x rates (88.2k or 96k). Yet the spec sheets and ads say “24-bit” (or more) and they say “192k” (or more). And the reviewers simply echo the numbers.
In the here and now, if it is a minority that can achieve the performance potential of 24/192, I take claims of higher numbers as a joke at best and cynical marketing at worst. Just my opinion of course but with so few showing they can design for 4x rates, why would anyone think those same few could deliver 8x rates (or more)? I find it interesting that those claims are not coming from the designers of gear that can achieve the potential of 4x rates. (We have the equivalent of makers of 2-cylinder subcompacts claiming to make cars that an outrun a Lotus!)
The numbers game isn’t limited to hardware. I’ve seen one company release CDs they claim were made with “32-bit mastering” and another claim “100 kHz resolution”. (Do they have 100k gear or are they rounding up from 96k?) Does anyone think those CDs are anything other than 16-bit, 44.1k? If there are such folks, I have a fine bridge in New York City to sell. The tools I use to create a CD master have 80-bit data paths and I’m working at 192k. The higher quality tools do result in a higher quality CD but should I then say they are made with “80-bit mastering”? Or that they exhibit “192 kHz [or 200 kHz] resolution”? I’d rather make records than sell bridges.
The finest 24/192 I’ve heard to date has given me back recordings I have not yet been able to discern from the direct input from my microphones. (To be clear, I am referring to gear that actually seems to achieve the potential of these numbers and not merely gear that sports them on a spec sheet.) Would 32/384 sound better? I suppose I’d have to hear the flaw(s) in properly done 24/192 first. And second, if the first condition was met (and in my experience, it has not even been challenged yet), that 32/384 gear would have to actually achieve the potential of that resolution and not merely claim it on a spec sheet. For me, right now, it is just marketing. Someday perhaps, we’ll have the audible evidence. Perhaps. Right now, I’m trying to imagine how it might be better than what is (so far) indistinguishable from the input signal.
24/192? 32/384? 64/768? Or should I wait for the 128/1536 version?
Is the best of today’s 24/192 too much? Is it not enough? I think it is just right.