Jitter Does it Matter?

INTRO: Some have asked for more details on jitter. It's a controversial topic and there are lots of myths associated with it. Here’s what I know.

TYPICAL JITTER MYTH: I read in a well regarded high-end audiophile magazine that portable players have less jitter because they use flash memory compared to a PC using a rotating magnetic hard drive. The explanation given: Because a hard drive is spinning, it must have jitter, while solid state flash memory does not. Like so many things in high-end audiophile magazines, this has absolutely no basis in fact. Jitter doesn't come from how the digital audio file is stored. It comes from how it was created, how it is sent in real-time (files are not real-time) over interfaces, and how it is re-constructed.

THE FLAC JITTER MYTH: Some argue that FLAC sounds worse than a WAVE file because the compression/de-compression process and/or CPU loading introduces jitter. This is completely false. The DAC receives the identical bit stream for either file type.

THE AES3 VS COAX VS TOSLINK MYTH: While it’s true jitter can be induced by interconnecting digital equipment, all three of the most popular standards AES3 (also called AES/EBU), S/PDIF coaxial, and S/PDIF TOSLINK optical, use essentially the same bitstream format and all combine the clock with the data itself. One format isn’t necessarily better than the other two as it very much depends on other variables. 3 feet of optical TOSLINK might easily outperform 6 feet of coax or 12 feet of AES3. It depends on the cable quality, transmitting hardware, receiving hardware, potential sources of interference, and more. The 3 standards are all prone to similar kinds of problems with AES3 being a bit more robust. Contrary to what some believe, the extra pin in AES3 is not a dedicated clock.

A REAL LISTENING TEST: I participated in a listening test at an Audio Engineering Society conference where they played test tracks with varying known levels of different kinds of jitter. It wasn't a rigid ABX or double blind test, but it was very enlightening. We, the listeners, didn't know which tracks were which until they were explained later. The most obvious audible effect was high amounts of jitter at certain frequencies sounds much like the old analog "wow and flutter" that is produced by vinyl turntables and tape recorders. Both devices have tiny deviations in speed--they might be slow variations (wow) like a vinyl record with the hole punched off-center. Or they might be fast (flutter) caused by the motor not rotating smoothly or bad bearings. Digital jitter, interestingly, often produces a rather similar end result.

PIANOS AND CYMBALS ARE CHALLENGING: Acoustic piano music seems to be an especially sensitive indicator of jitter (and wow and flutter). As the jitter increases, the notes take on a more brittle quality--that expensive Steinway Grand now sounds more like a garage-sale upright piano. And when the jitter is really high at certain frequencies, you can hear a "warble" to sustained piano notes--just like with low quality tape players or turntables. Some also say cymbals are good at revealing jitter.

WHAT IS JITTER ITSELF? Imagine a ticking clock that keeps perfect time over the long haul, but the second hand might move every 0.9, 1.0, or 1.1 seconds. If the variations average out evenly, the clock still keeps perfect time. But if you try to measure just 1 second, it might be off by 10%. That's jitter. The 44.1 Khz clock used for CD audio averages out to the right rate but the individual “ticks” may not be so accurate. These variations introduce a form of time distortion into the music. Excessive jitter is typically caused by poor design, cost savings, poor interfaces, or other shortcuts.

JITTER IN SPECTRUM VIEW: If you look at the audio spectrum both wow & flutter and jitter manifest themselves as symmetrical sidebands to a fundamental pure frequency. And very low frequency jitter causes the base of the pure tone to “spread”. The height and extent of the spread indicates the magnitude of low frequency jitter. The height and number of the side bands indicates the magnitude of higher frequency jitter.

SAMPLING ACCURACY: To properly digitize and re-construct digital audio it's relatively important the sampling intervals be as accurate as possible between the recording process and when it's played back. Random or periodic variation to these intervals along the way is jitter. It gets complicated (and controversial) to describe all the possible sources, and interactions, of jitter but some things are fairly well understood and agreed upon.

JITTER SOURCES: Clock circuits are everywhere. Pretty much anything with a CPU—right down to the little keyfob that unlocks your car’s doors—has a clock in it. For some applications, like a wristwatch, the important spec is how accurate the clock is when it’s manufactured, over time, and with changes in temperature. If it averages above or below the designed frequency the watch won’t keep accurate time. But jitter in this application virtually doesn’t matter. It could be relatively huge and the watch will still work exactly as expected.

CLOCK PHASE NOISE: Another name for clock jitter is “phase noise”. The clock for digital audio equipment has to be both accurate in frequency (otherwise the audio will play fast or slow) and have low jitter (to avoid significant sampling error). The lower the jitter the more expensive the clock circuitry becomes. All clocks have some inherent jitter even if they’re operated in a perfect isolated environment, from a perfect power supply, etc. And when you put them in a less than perfect environment—typical consumer electronics—the jitter will usually get worse because there’s noise on the power supply, ground noise, and likely some EMI (electromagnetic noise) from other parts of the circuitry. Just changing the PC board layout can have a significant impact on jitter performance even using the exact same parts.

JITTER REDUCTION: Some audiophile manufactures making jitter reduction claims are likely just touting the features built into whatever chip(s) they’re using and going off the datasheet for the part(s). Some, I know for a fact, don’t even have equipment to even properly measure jitter. So it’s not surprising some of the products from these manufactures have lousy jitter performance despite the high-end parts used, and their marketing claims. You can’t take jitter performance for granted, design only by ear, etc. You have to measure it.

JITTER REDUCTION METHODS: Various methods are used for jitter reduction. Some use a PLL, a double PLL, Asynchronous Sample Rate Conversion (ASRC) techniques (as Benchmark does), and now some USB products use a different form of the USB Audio interface specification known as Asynchronous. Each company will often argue their way is the best way. But, in reality, each has its own strengths and weaknesses. The best way I know to compare them is to evaluate the jitter spectrum using a J-Test.

CABLE JITTER: Cables can “smear” digital signals by attenuating the highest frequencies. The diagram to the right is an example. The top blue waveform illustrates a perfect digital S/PDIF or AES3 bitstream. The bottom red waveform is what you might get out the other end of a long cable. The hardware receiving the signal uses the “zero crossings”—where the signal transitions an imaginary line drawn horizontally through the middle of the red waveform—to extract the clock. As you can see by the red arrows, that isn’t always correct due to the waveform distortion. The gaps between the red arrows are jitter. The amount of “smearing” depends on the bitstream itself so as the audio signal changes so does the clock timing creating jitter related to the audio itself.

RECORDING JITTER IS A MORE SERIOUS PROBLEM: If the device doing the A/D conversion during recording has a poor clock, or there are other clocking issues (such as a multi-track with multiple A/D devices recording simultaneously), the music will not be reliably sampled with low jitter. And, contrary to what many manufactures and the audiophile media would like you to believe, you cannot correct for recorded jitter during playback. Even those uber-expensive "jitter correction" devices like the Genesis Digital Lens can't fix recorded jitter. This is partly why I invested nearly $2000 in the Benchmark ADC1. I wanted an A/D that had reference quality jitter performance as you're stuck with whatever the A/D hardware gives you.

FURTHER INFORMATION: Here are 3 links including Benchmark's excellent description of jitter and how they deal with both measuring it and eliminating it, a generic description of jitter in general (not just audio jitter) on Wikipedia, and a rather wordy and technical PDF of an Audio Critic issue which had Jitter as their feature article:

Jitter Effects & Measurement (by John Siau, Benchmark Media)

Sound on Sound Benchmark Review With Jitter Discussion (an independent verification of Benchmark’s method)

Jitter on Wikipedia

Audio Critic Jitter Issue (pdf)

MEASURING JITTER FROM AN ANALOG SIGNAL: There are various ways to measure or evaluate jitter depending on what you're testing. If you're forced to test in the analog domain (i.e. gear with only an analog output), you generally use a particular FFT spectrum analysis of certain test signals and visually interpret the results. The question is what signal to use?

J-TEST: Julian Dunn at Prism Sound (makers of the dScope) developed the Jtest as a way to provoke “worst case” jitter for the purposes of measurements. His work was published by the Audio Engineering Society and has become something of a benchmark for Jitter measurements. The Jtest signal creates a jitter “torture test” at exactly 1/4 the sampling rate combined with toggling the lowest bit in a way that exposes jitter. I use the same Jtest signal with a similar spectrum analysis and my results should compare well those in Stereophile and elsewhere.

SIDEBANDS: As mentioned earlier, periodic higher frequency jitter is revealed as symmetrical pairs of “sidebands” around the main signal—this is analogous to the old mechanically induced flutter. The “height” or peak level of these sidebands indicates the relative amplitude (in picoseconds or nanoseconds) of the jitter components. While “spread” at the base of the main 11,205 hz signal indicates random low frequency jitter—this is analogous to the old mechanically induced wow. The greater the spread, and higher in amplitude it reaches, the greater the low frequency jitter amplitude.

MEASUREMENT VALIDATION: Variations of this method has been well documented and verified by many. It relies on the fact that nearby symmetrical sidebands to the high frequency signal are not likely to be from anything but jitter. The difference in frequency is much less than a single sample period of the digital signal. So “distortion” components this close, in matched pairs, are extremely likely to be jitter related. Here's a description that's less technical than most and also serves to help validate the Benchmark link above: Measuring Jitter

PC SOUND OUTPUT: I've used the above technique and can say it works. Some devices have minimal jitter, and some have fairly obvious jitter. Here, for example, is the Jtest jitter spectrum of a typical PC’s audio output. The high amount of noise on the motherboard likely corrupts the digital audio clock and/or bitstreams creating more sources for high frequency periodic jitter than usual. There’s also significant “spread” indicating low frequency jitter:

 

M-AUDIO TRANSIT USB INTERFACE: The M-Audio shows much more “spread” all the way up to about –88 dB and fewer sidebands slightly lower in level than above:

 

FiiO E7 USB DAC: This is an improvement over both of the above showing less spread and a rather different distribution of jitter sidebands:

 

SO HOW MUCH IS TOO MUCH? Based on all the research I’ve done, if the total sum of all the jitter components is less than –100 dBFS within the audio band, the jitter will be inaudible. I’ve further tried to clarify this by developing a “limit guide line” with individual components being under –110 dBFS with a bit more leniency close to the fundamental test signal but still under the –100 dBFS limit. Here’s an example of the DAC1 with the limit line shown in green. You can see the $1600 DAC performs very well on jitter:

 

MEASURING A DIGITAL SIGNAL: In the digital domain, if you're say looking at a digital signal coming out of an A/D converter, CD/DVD/Media player with a digital output, etc. you can directly measure the jitter with the right instrument. The Prism Sound dScope has a low jitter internal time base clock. This allows the dScope to compare the clock in the digital signal to its own quality time base. From the timing differences it can calculate an actual jitter value in the time domain.

OSCILLOSCOPE MEASUREMENTS: Some think if you look at a digital signal on a scope with an infinite persistence feature the “smear” you get is jitter. But it’s not. What you’re really seeing is variations in the scope’s trigger point. Because the scope is entirely triggering on the incoming waveform, it’s automatically trying to filter out any jitter. The only way to measure jitter is to compare the signal in question to a known highly stable source. And scopes are not, by themselves, generally set up to do that.

SUMMARY: My thoughts on jitter are pretty simple:

• It doesn't matter how a digital audio file is stored or if lossless compression is used. A file played back on a PC, network file server, portable flash-based player, WAVE, FLAC, etc. are all the same with respect to jitter at the file level. It's what happens later in the playback process that matters.
• Many devices exist which effectively reduce playback induced jitter to very likely inaudible levels. If you're worried about jitter it's easy enough to choose one of these devices--especially ones that back up their marketing claims with some actual valid test results (like Benchmark, Anedio, etc.).
• If you’re worried about jitter, choose a device that has been independently tested for jitter with similar jitter spectrums to those shown in this article. Look for sidebands and “spread” under –110 dBFS.
• Jitter present on recordings, from the A/D process, is a more serious problem and cannot be removed by any playback hardware.