Do Hard-Disk Drives Sound Better Than CD?
Do Hard-Disk Drives Sound Better than CDs?
Robert Harley
A reader's post of an interesting and relevant question regarding the sound quality of hard-disk-based music servers versus CD
(http://www.avguide.com/forums/cd-player-vs-computerserver-system) prompted me to post my experience with the subject as this blog. This text originally appeared in The Absolute Sound Issue 177 as part of my review of the Sooloos and QSonix music servers.
There’s been much debate about the sound quality of CDs played back by a conventional transport versus the sound quality of files made from the same CDs, copied to and then played back from a hard-disk drive. Many contend that hard-disk drives sound better, much as CD-Rs sound better than the source CD from which the CD-Rs are made.
I tested this contention by comparing the sound of the Sooloos and Qsonix servers to a state-of-the-art CD transport (the Esoteric P-03), alternately feeding the music server under evaluation and the CD transport to an Esoteric D-03 digital-to-analog converter through the same digital interconnect.
Listening to the same CD from Sooloos and from the transport, I heard a subtle but noticeable improvement in sound quality from the music server. I heard more space, air, bloom, and soundstage depth when the audio data were read from the Sooloos hard drive rather than from the CD. The hard-drive-sourced sound had better resolution of low-level detail, particularly reverb decay, which is why the presentation sounded more airy and spacious. The presentation was gentler, a little more laid-back and relaxed, and had a greater sense of ease. The upper midrange and treble were smoother and more “organic” sounding when the data feeding the DAC was sourced from Sooloos’ hard-disk
drive. I repeated this comparison with the Qsonix server and heard a similar increase in resolution, improvement in spaciousness and depth, smoother textures, and more relaxed presentation. In fact, the two music servers sounded very much alike.
Much of the current speculation as to why this should be the case focuses on the fact that the Exact Audio importing software used in Sooloos and many other hard-drive-based systems can read sections of the disc multiple times if data errors are detected. The implication is that random bit errors affect sonic qualities such as soundstage depth and the reproduction of timbre.
I disagree. CD’s error-correction system can completely correct burst errors of up to 4000 successive bits. My experience performing bit-for-bit comparisons between source data and replicated CDs when I worked as a CD mastering engineer suggests that bit errors are virtually nonexistent. If CDs were rife with errors, computer software couldn’t reliably be delivered on CD (CD-ROM does, however, have an extra layer of error detection and correction but is invoked only when discs have been mishandled and become damaged).
Let’s perform a thought experiment. Suppose that uncorrected bit errors are common in CD playback. What are the sonic consequences of those errors? If a 16-bit word’s least significant bit (LSB) were changed from a zero to a one (or from a one to a zero), an amplitude error would be introduced that is one part in 65,536. (The number of amplitude steps in a quantization word is 2n, where n is the number of bits in the word; 216 is 65,536.) You would never hear an amplitude error that small in a single sample. But let’s say the bit error occurred in the word’s most significant bit (MSB). The value of the MSB is the value of the other 15 bits combined plus one (32,769). The MSB represents just over half the amplitude of a full-scale signal. Changing the MSB from a zero to a one, or from a one to a zero, would cause the signal level to suddenly increase or decrease by half the full-scale amplitude, producing a loud click at every error. Do you hear clicks when playing CDs?
Note also that bit errors would cause momentary jumps in signal amplitude, but would not affect factors such as soundstage depth or instrumental timbre. Bitstreams read from different media (CD, CD-R, hard disk) can sound different from one another, but not because of data errors. The most likely explanation is that hard drives deliver a bitstream with greater timing precision (lower jitter). If the bits are the same, and the sound is different, the only thing left is jitter.
The Esoteric P-03/D-03 combination’s separate clock link allowed me to test this hypothesis. This separate clock link slaves the transport to the processor’s clock, a feature that greatly reduces jitter in the D/A’s word clock. (The word clock controls precisely when the DAC converts the incoming samples to analog, which is the point where jitter matters.) The arrangement obviates the need to extract a clock from the S/PDIF interface, a primary source of jitter in separate transports and D/A converters. In addition to avoiding the jitter-inducing effects of the interface, another advantage of a separate clock line is that the D/A converter becomes the timing reference for the entire system. That is, the D/A converter generates the timing reference for the crucial digital-to-analog converter stage, and the transport must lock to this reference. In a conventional connection between a transport and D/A converter, the D/A converter must lock to the transport’s timing reference. The result is higher jitter in the clock that controls the actual digital-to-analog converter chips. The separate clock line eliminates these two shortcomings.
I compared the sound from the music server’s digital output to the Esoteric P-03 transport with and without the separate clock line (you can engage and disengage the clock with a front-panel button). Engaging the Esoteric’s clock (lowering the jitter) when listening to a CD rendered an improvement in sound quality virtually identical to what I heard when the music was being streamed from the music servers. This suggests that jitter is, indeed, the explanation for why hard-disk drives sound better than optical discs.
Note that in this experiment, the music server’s output was subjected to the jitter-inducing S/PDIF interface while the transport’s output was not. One might reasonably infer that jitter in the signal recovered from an optical disc is at least as significant a source of sonic degradation as the S/PDIF interface. The ideal solution is hard-disk-based storage with no S/PDIF interface. It’s not just the S/PDIF interface that’s a problem; any asynchronous connection between source and DAC is potentially problematic, including the popular USB connection for connecting a computer to an outboard DAC.
We will increasingly listen to music from hard-disk drives; it’s good news that their advantages don’t come with a sonic penalty. Indeed, hard-disk drives actually deliver better sound than optical disc.
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Comments
Thanks for the great technical article.
I am curious about jitter issues. If I connect a computer (say a mac mini) to a DAC via S/PDIF (optical) should I be worried about jitter issues?
Any time you pass a digital audio signal through the S/PDIF interface, jitter is introduced. The S/PDIF interface is asynchronous, meaning that the transmitting and receiving clocks are not locked. In the interface, the clock is embeded in the audio data (it's more correct technically to think of the audio data being embedded in the clock). This clock is then recovered and serves as the reference clock for digital-to-analog conversion. This system by its fundamental nature introduces timing errors, particularly if the interface isn't implemented as a perfect transmission line.
This phenomenon was first documented in the Audio Engineering Society Paper "Is The AES/EBU/SPDIF Interface Flawed?" by Malcolm Hawksford and Chris Dunn (1991 AES convention).
Hi Robert,
"...The ideal solution is hard-disk-based storage with no S/PDIF interface..."
Agreed. The best I've heard (by a good country mile) has been from the hard disk via my Firewire interface.
Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com
Barry,
What are you using for a Firewire interface?
Thanks,
Mike
Hi MikeD,
My current interface is an as yet unannounced, fully functioning prototype and I am not at liberty to divulge the source.
In the past, I have had wonderful success with a Metric Halo ULN-2, which I cannot recommend highly enough.
Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com
Hi MikeD,
My current interface is an as yet unannounced, fully functioning prototype and I am not at liberty to divulge the source.
In the past, I have had wonderful success with a Metric Halo ULN-2, which I cannot recommend highly enough.
Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com
That is simly NOT TRUE as a whole thing.
This article is nice example of manipulation. There is absolute true about what can bad bits do in sound. And then there is pure speculation about what can jitter do presented in simmilar way to the facts.
One thing: You have a computer with hard drive, CD drive and digital output (eg. S/PDIF). When you play the music directly from the CD (which is since Windows 2000 in fact software-based audio grabbing and sending those data to sound card, that sends that data to the output), or when you first grab that CD to hard drive (presumably without errors in data) and then play that sound from the hard drive, there does not exist anything that would change the sound. It simply must be the same, because sound output form is not up to the hard drive, nor up to the CD drive, but up to the sound card (audio chip).
I wonder, why NO ONE of such bats (those people that can hear unaudible things) wants to make "blind hearing test". No, in fact, I do not wonder. I have a friend who tried to make that with some of his "bat-enabled" friends. He tried also to make "fake not-blind" hearing test, where he sometimes told them that they are listening to "something" when they were in fact listening to "something else". Their conclusions were mostly according to what they have thought they were listening to. In blind hearing test they were statistically unable to tell what they are listening to.
The simple fact is that placebo is a big bastard.
@L from the heaven,
The deleterious effects of jitter have been extremely well documented in both measured, blind and sighted tests for a lot longer than CD has been around. In fact, one of the most important papers on the subject - Digital Sound Signals: subjective effect of timing jitter, by W.I. Manson (http://www.bbc.co.uk/rd/pubs/reports/1974-11.pdf) - just celebrated its 35th birthday, and the AES have papers on the subject that grandfather that one.
Manson's seminal paper was from the time when the BBC had a Research Department, and when it - along with NHK in Japan - was a significant force to be reckoned with in audio engineering. Sadly, no more; the BBC's legacy of R&D is just that... a legacy.
I must object to the statement that no one wants to make blind tests of these things. I am planning a blind listening test to determine whether there is a difference between the sound of CD (played through a CD player) vs the sound of the same CD ripped and stored on a hard disk and stored bit-for-bit. I hope to compare both level matched (should it be necessary) through the same DAC, using the USB connection from a computer and S/PDIF output from a CD transport into the same preamp, power amp and loudspeakers. The test will be run as a level-matched ABX, although the components to be used in the test are still to be confirmed. If it is logistically possible, the test will be conducted double-blind. Given this test will be conducted at an audio show here in the UK in September, it should be possible to have a large enough sample of listeners to determine whether any difference can be heard.
RH's comparison has covered the same basis, albeit not under blind conditions (possibly because he didn't have tens or hundreds of test subjects traipsing through his listening room at the time to make a valid blind test).
There seems to be a fixation on blind testing to evaluate products, neatly forgetting that it is entirely the wrong tool for the job in most cases. Such tests are perfect for evaluating whether there is difference between two things, not to attempt to quantify those differences. In other words, a blind test should have no 'is it better' component, just 'is it really different?' Play A. Play B. Play X and see if people can spot whether X is A or B. That's it. If you try to introduce any other test into the mix, the whole thing falls apart in one big bolus of unrobustiocity. If we determine that a difference exists through blind test, it then hands over to subjective testing to determine the 'shape' of those differences.
A blind test is there to determine whether people can tell whether there's a difference between vinegar and red wine. Assuming such a difference exists, it's then down to the wine lovers to tell the difference between a $10 bottle of 'plonk' and a $10,000 bottle of vintage Chateau Petrus.
Alan Sircom
Editor, Hi-Fi Plus Magazine
London, England
editor [at] hifiplus [dot] com
Unless these tests are performed SCIENTIFICALLY ie in a strict double blind fashion, any findings or conclusions drawn from them are UTTERLY MEANINGLESS.So much so called 'professional investigation' is just so much opinionated twoddle.Unless the listener is effectively 'blinded' and all playback and measuring equipment is standardised and only a single variable is compared at any one time, any conclusions are merely empirical anecdote and opinion - not fact - which is why 99.9% of hifi reviews and reports using different (and frequently changing) equipment in different listening rooms using different supports, cables,interconnects and power supplies under different listening conditions have absolutely no relevance to anyone. Whatever is or is not not heard by a particular reviewer on a particular day is a one off experience personal to him and him alone - all of it coloured by personal bias,prejudice,conditioning.Have you noticed, these experts usually hear what they expect to hear? What a surprise! As the contributor above rightly points out; PLACEBO IS ONE BIG BASTARD.
@RWnonymous,
Strangely, although I do not measure or sit through blind tests routinely in the evaluation of products any more, I have found my subjective tests have tallied both with measurement of products conducted by other reviewers and often with the conclusions drawn by other reviewers. If reviewer X, Y and Z independently report the same characteristic about a product irrespective of the systems they test it on, do you really hold that the characteristic reported is meaningless?
It's somewhat difficult to draw this conclusion in the US right now, because there are only two main hi-fi titles currently published. In the UK, there are five. We often get the same new product roll out to four or five magazines at the same time. They all go to different reviewers with different systems and these reviewers seldom speak to one another about products. Some are purely subjective, some combined subjective/objective, some tested blind to a listening panel. The reviews are often published at the same time and there is a high degree of correllation between the findings of the reviews; I'd say it's less than one in 50 where the reviews do not draw very similar conclusions.
So... really sure it's just down to placebo?
Alan Sircom
Editor, Hi-Fi Plus Magazine
London, England
editor [at] hifiplus [dot] com
To my mind, this is not manipulation.
I agree about some imprecisions about some technical aspects ( for exemple : we you listen to a CD with ( not too much ) errors, you hear no click either ).
But I don't about the only placebo effect.
When I ( and many others ) listen directly from a CD via a computer, the sound is usually very bad. I done long double blinding tests on that.
But there is some technical reasons for that.
y.
I have my entire CD library ripped and stored as FLAC files on my home network, more for the convenience of being able to listen to my music anwhere in the house without needing to break out the CD. However, what would you suggest as an interface for a computer if both USB and S/PDIF are so prone to jitter themselves? AES/EBU (such as a PCIe card from Lynx Studio Tech), or Firewire as suggested earlier, or something else?
- Chris Heinonen
I don't want this thread to turn into yet another debate on blind listening tests. For those who are skeptical about the role of listening in judging reproduced sound, I suggest reading the recently updated and expanded version of my Audio Engineering Society paper "The Role of Critical Listening in Evaluating Audio Equipment Quality" available here:
http://www.avguide.com/news/2008/05/28/the-role-of-critical-listening-in-evaluating-audio-equipment-quality/
Comment on blind listening tests are best posted here:
http://www.avguide.com/forums/blind-listening-tests-are-flawed-editorial
Mr. Harley, et al.,
Could you comment on the DAC1 USB, which claims to also have solved the clock synchronization problem? As a low-budget guy with high-end tastes, I'm always in need of something that goes most of the distance of the state-of-the-art solution, at a fraction of the cost.
See: http://www.benchmarkmedia.com/system1/digital-analog-converter/dac1-usb
Also, I come across claims of "asynchronous USB mode" and similar anti-jitter USB modes, attached to some lower-end DACs now. Is this a workaround solution? If so, who offers a reasonable implementation of it at various price points?
Thanks so much.
Hi Art,
In my view, USB is a compromise for audio. It was originally designed as an inexpensive means for connecting a mouse and keyboard and unlike Firewire, was not intended to stream audio or video. It needs to poll the CPU to decide when to send its "packets", where Firewire operates peer-to-peer and was designed from the get-go to stream audio and video.
All that aside, what counts is performance. Just from a sonic standpoint alone, I would suggest auditioning a Metric Halo ULN-2 (if you have a Mac, which is required to set it up - after which it can operate standalone). To my ears, the ULN-2 will leave any USB DAC in the dust.
If, on the other hand, your listening is limited to 16/44 ("CD quality") files, a USB DAC will not be as stressed, though the ULN-2, with (to my ears) better conversion, analog stages and clocking will still shame it.
Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com
In the current issue of Hi-Fi Plus, we have tested the Benchmark DAC1 Pre (this is essentially the DAC1 USB, with - surprise, surprise - a preamp stage, but also with improved op-amps in the analogue stage). With regard to the Benchmark's jitter claims, there's some confusion about whether the process is a function of the Benchmark system or intrinsic to the Analog Devices AD1986 chip the Benchmark products use. However, the result is the same; the DAC uses a very good 'polyphase' PLL (phase lock loop) implementation and an über-stable clock. So, instead of thinking 'eliminating jitter', look at it as 'side-stepping the jitter issue' - but that still works. I don't think this is a 100% solution, but it it's an effective real-world fix.
Regardless, we are very impressed with the Benchmark; it's strikingly neutral and stunningly detailed. It does the state-of-the-art thing well, but it's not going to out-Wadia a Wadia.
As to asynchronous USB... more investigation is needed. With Ayre, dCS and Wavelength making asynchronous-mode USB products, it's definitely worthy of checking out.
Alan Sircom
Editor, Hi-Fi Plus Magazine
London, England
editor [at] hifiplus [dot] com
Quoting directly from Benchmark website:
"Jitter is NOT a problem for the DAC1 and DAC1 USB, which achieve jitter immunity by utilizing the proprietary UltraLock™ clock-recovery system. With the UltraLock™ clock-recovery system, the digital-to-analog conversion-clock in the DAC1 and DAC1 USB is totally isolated from the clock of the digital audio input. This clock-recovery topology outperforms even the most well-designed two-stage PLL designs. Using state-of-the-art Audio Precision testing equipment, no jitter-induced artifacts can be detected with the DAC1 or the DAC1 USB."
Does this make sense? Is the testing protocol valid?
I'd love to hear others comments.
Best wishes,
Art
Hi Art,
I think audiophiles and reviewers tend to "lock on" to certain mechanisms once named and quantified.
The current buzzword seems to be jitter. Personally, I think there are other as yet unnamed parameters and mechanisms involved that degrade digital performance. (Remember that when the next "big problem" is identified and jitter fades away as a concern.)
In my experience, claims of "jitter immunity" are nonsense but based on what I said above, I don't pay much attention to them anyway.
The bottom line is that is a very nice unit. For a bit less expense, I find the ULN-2 to be a considerably better unit (that just happens to throw in a superb pair of mic preamps, A-D converters, top shelf clocking and analog stages and the best recorder I've ever heard, regardless of price or format). The ULN-2 does require a Mac for initial setup, after which it can run as a standalone DAC, with monitor level control.
Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com
Robert,
In your blog you note "The ideal solution is hard-disk-based storage with no S/PDIF interface." Wouldn´t such a solution introduce noise from the hard-disk-drive?
You rightly point out that the jitter problem remains with typical USB-connected DACs (it is my hypothesis that clock-recovery systems never seem to fully repair jitter that is already in the signal - not tested) Is this is different for the recently announced solution by dCS: http://www.dcsltd.co.uk/product/scarlatti-upsampler. From that webpage:
Asynchronous USB Mode
In USB, there are numerous modes for synchronising the audio between the PC (the host), and an audio device. The most popular of these, "adaptive", involves the audio device synchronising itself to the USB "frame" provided by the PC. This tends to give poor jitter performance. The Scarlatti Upsampler's USB interface operates in "asynchronous" mode (NOT to be confused with asynchronous rate conversion), where the Upsampler synchronises the audio by providing a feedback pipe to the PC. The PC then is effectively locked to the Upsampler, which can have a much more accurate clock and much lower jitter.
How is the use of the word "asynchronous" here different from yours?
Many thanks for your insights.
I do not understand what sometimes appears to be the obsession with USB DACs when the highest quality is desired.
They make fine cheap DACs for listening to mp3s but why would anyone want to add a USB DAC to a high end system when Firewire is free of USB's problems and limitations?
Playback from the hard disk via Firewire is by far the best reproduction I've heard.
In a studio, mastering room or any other mission critical situation where the highest quality is a must, why would I want to show up at the race with a Geo when there is a Ferrari available?
Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com
One reason for using USB is that Firewire (unfortunately) may be going the way of the dodo. Many computer manufacturers are abandoning Firewire.
Mr. Plus,
Thanks for the pointer to Ayre (whose technology is licensed from Wavelength). This is an interesting read on the subject:
http://www.ayre.com/PDF/Ayre_USB_DAC_White_Paper.pdf
This "asynchronous USB" solution seems similar to dCS´s.
The question remains wether these solutions result in better performance than from an appropriately master-clock controlled transport, or just the same/similar performance.
Anyone any experience with one of these dCS, Ayre or Wavelength products?
I wouldn't presume to discuss this on any kind of esoteric technical level, but this topic is very interesting to me:
I recently ripped all of my CDs to a MacMini in Apple Lossless format, and am now playing them throught my home audio system WIRELESSLY using the MacMini's 802.11 G connection paired with an Apple AirPort Extreme. The AirPort is connected to the amplifier with a TOSlink cable.
My first impressions were that the music somehow sounded better than it had when I played it directly from CD, also connected via TOSlink. Further listening has reinforced those initial impressions, however unlikely it seemed at the time.
So, my CD players are in storage, and my music is more accessible and better-sounding than ever. Who'd a thunk it???
My understanding of how Benchmark achieves immunity to interface jitter is that it employs an asynchronous sample rate converter at the input. This chip, made by Analog Devices, resamples the datastream under control of its own clock. It doesn't recover the clock from the interface. This device can take in a wide range of sampling rates and the output will be a precise 44.1kHz.
A drawback of this approach is that timing errors at the input are converted into amplitude errors because of the resampling. Yes, the output clock is stable, but the sample values have been changed.
I attended the presentation of a paper by the chip's designer, Robert Adams of Analog Devices, at an Audio Engineering Society convention in the mid-1990s. At the time, the idea of changing the sample values seemed anathema to high-end values. I had always wondered about the sonic effects of removing interface jitter at the expense of altering the sample values.
My question was answered at the most recent CES, where PS Audio demonstrated their Perfect Wave and DAC. The DAC has a sample-rate converter that can be switched out of the circuit. I listened to the same music going through, adn then bypassing, the sample rate converter. My experience was that the sample-rate converter added hardness and glare, reduced resolution of low-level detail, and truncated the sense of space.
It is possible that the Benchmark doesn't suffer from these problems, but I have not had the opportunity to hear one for myself.
Hi Robert,
This is entirely in line with my experience testing (what by now must be dozens of) sample rate conversion algorithms.
I have yet to hear real-time conversion that does not add spurious harmonics that result in a hardening and brightening of the sound, obscuring low level detail and shrinking the space (not to mention "defocusing" it).
My method for testing is to compare against the unconverted original. This can be merciless at revealing just how bad most SRC is. And it partially explains why some folks record at 88.2 or 176.4 as many SRC algorithms have an easier time dealing with integer conversion. The best, most transparent SRC I've heard performs at non-integer rates (e.g. 192 to 44.1) with none of these artifacts. It also takes quite a while, perhaps 5-10 minutes per minute of program material. (Some of this may be a function of my computer's CPU, which is now a few years old and not as fast as the latest models.)
(While I would agree that upsampling has its benefits, I would much rather perform this outside of real-time. Great as their marketing value might be, the players that upsample as the disk plays all exact to my ears, a similar sonic price to what you have described.)
Similar things can be said for most dither/noise shaping algorithms I've tested. Most tend to defocus the space and alter instrumental timbres. The best are quite faithful to the undithered original file.
Additionally, the software used to make the comparisons (i.e. to play the files) makes a difference. I would not have expected this, though as an experienced engineer and audiophile, I can't say why. I have four different applications I use for editing/mastering (each has strengths the others don't) and each sounds quite different from the others. Taking an original, minimally mic'd recording and bringing it into each of these, making a gain change and saving the results provided an interesting and educational test. The gain change was not as demanding on the "engine" of each application as say, applying equalization would be. Still, bringing the four files into each application in turn showed that some were better at revealing the differences than others. I was not surprised to find the app that created the most transparent sounding file was also the best at revealing the differences between all four files.
Best regards,
Barry
www.soundkeeperrecordings.com
www.barrydiamentaudio.com
So it sounds like we won't have to spend $15,000 on an atomic master clock - a quality hardisk should do it....
It's interesting that different sofware applications affect the sound differently.
Thanks for sharing your insights on this and other topics with everyone on AVguide. Audiophiles should know what goes on behind the scenes in creating the music that we play on our systems.
I recently came across your blog and have been reading along. I thought I would leave my first comment. I don't know what to say except that I have enjoyed reading. Nice blog. I will keep visiting this blog very often.
Joannah
http://2gbmemory.net
Robert,
I'm considering which options would be best for upgrading? The Marantz top of the line SACD; the new Denon Universal player, the Audio Research CD8; or chunk them all for the PS Audio Perfect Wave or one the competing Media Server systems? The Denon and the Perfect Wave seem to be the best options, but I wonder about the sound quality.
How does/should you future proof one's system and be economical ?
Maybe the answer is buy a cheap universal blu-ray player and save for the Spectral 4000 unit ?
:-)
Robert,
I'm considering which options would be best for upgrading? The Marantz top of the line SACD; the new Denon Universal player, the Audio Research CD8; or chunk them all for the PS Audio Perfect Wave or one the competing Media Server systems? The Denon and the Perfect Wave seem to be the best options, but I wonder about the sound quality.
How does/should you future proof one's system and be economical ?
Maybe the answer is buy a cheap universal blu-ray player and save for the Spectral 4000 unit ?
:-)
"Many contend that hard-disk drives sound better, much as CD-Rs sound better than the source CD from which the CD-Rs are made."
Mr Harley (or anyone): Why might CD-Rs (copies of orig. pressed/stamped commercial CDs) sound better?
thank you
Nice article, and clever thinking. However, I must say I disagree with your conclusion because there is something incorrect with your argument.
Let's state some things. First of all, the word jitter has several meanings in audio. Here you are using it as imprecision in the clock at the DAC, which is a continuous phenomenon, different from other kinds of jitter that are better understood discretely. And you are right: the kind of jitter you are speaking of has a detrimental and very audible effect on audio quality. It is good to have your experience here, and that of other forum members to describe this problem so well from a subjective point of view.
Nevertheless, this problem has nothing to do with whether the source is optical or a magnetic surface, or any other kind of source as long as they are digital. There is no relationship between the source and the timing of audio samples at the DAC (which is the only timing imprecision that can affect sound the way you described as you correctly said). No matter what kind of media the bits are stored in at the source, they all finish in a memory buffer before they are converted to analog signals. So, in a sense, the source media is always a silicon chip, and the information it sources has all about the signal and nothing about the original kind of media or how timely it delivers digital samples; it makes no difference.
What actually makes the difference you heard is how precise is the clock that times when each digital sample is sent to the DAC for conversion into a voltage analog signal. There is, however, a special case where your reasoning is right, and it is in poorly implemented SPDIF or similar reception: when the DAC relies in PCM timing instead of a memory buffer and a proper clock to time the digital to analog conversion.
Personally, I solve all this problems in my audio setup using a computer (a Mac Mini in my case, but that's irrelevant) connected to an audio interface for output with decent DAC and good conversion timing (I use the Saffire LE firewire interface for that). Never mind any timing horrors that may occur in the middle (if any) because all "bad timing random information" that may be generated is lost at the output. Simply put, all that matters is timing in sending samples from the buffer to the DAC. No reasonable system has the capability to "remember" bad timing in the source or middle of the chain. It would be expensive to implement.
In conclusion. You are completely right in that output clock jitter is detrimental to audio quality and it explains what you have found. Nevertheless, you are wrong in thinking that hard disk vs optical has anything to do with it. When it's digital, it is the output DAC clock precision what matters (and its linearity), regardless of the source media.
I have a related question as regards hard drives and data transfers. I work in a high end A/V studio and am running Pro Tools and transferring data from computer to computer using portable hard drives. These are wave files at 96k 24bit. I did a recent set of tests where the same file was transferred using different hard drives, using flash drives, and using different USB and firewire cables. Each transfer was loaded to a Pro Tools session on the receiving computer so that they could be easily called up and compared. We're talking about file transfers, not streaming. Theoretically, you would think they should all sound the same. However, they definitely did not. Some were duller than others, some had a difference in the low bass. This has me a bit baffled as to what could account for the differences. Can anyone shed some light on this?
luigilew
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