John Siau, the head designer and principal at Benchmark, writes white papers and posts them to their site. As many of you know, I regard John to be one of the industry’s real experts and I defer to his expertise more times than I care to admit. His latest post is about sample rates and the Nyquist Theorem. It’s well worth the read. He manages to explain the essence of digital sampling in a clear and simple to understand way. You can click here to read it.
The bottom line on traditional PC sampling is that 96 kHz or 88.2 kHz is more than enough plenty of samples. The filters necessary to remove any frequencies higher than the sampler rate divided by 2 (the Nyquist) are well within the design abilities of current designers.
However, if you’ve been following Robert Stuart and the MQA initiative and other experts on the topic, the discussion turns to timing. There is lots of talk about the inability of lower sample rates to capture the “timing” ability of human hearing, which is pegged at between 5-10 microseconds. The claim is that unless we increase the sample rate to at least 192 kHz or 384 kHz or higher that music recordings are not delivering everything that our brains need to reconstruct a live audio event.
The need for these higher sample rates isn’t about making sure we include ultrasonics in our tracks but to subdivide time so fine that we don’t have any inter ear timing errors.
The arrival of audio signals to our ears is tremendously important for several reasons. But we have to distinguish between micro and milli second timing to understand theses reasons. The millisecond timing differences are used by our ear for directionality. Our stereophonic hearing is very good at being able to identify where a sound is coming from.
The microsecond timing stuff is the domain of neuroscientists and according to the Meridian guys there is recent research regarding the importance of getting this area of timing right.
Then there’s getting the “transients” right and the related “pre and post” ringing out of our PCM digital filters. The higher the sample rate the higher the point at which the ringing occurs. For traditional CDs, this means that the “pre-ringing” happens at just past our hearing range,. Maybe this type of error is audible or affects the sound of CD or maybe not. But if we raise the sample rate to 96 kHz it ceases being a factor.
This is the gist of what John wrote when I asked him about the benefits of higher sampling rates on the timing aspect of digital audio. Here’s his response:
“Timing accuracy is not a function of sample rate. I tried to illustrate this in my paper using the examples of the runner and the rotating wheel. Nanosecond timing variations are easily resolved with a 44.1 kHz sample rate. This is easily demonstrated with a jitter signal of a few nanoseconds. Jitter would not be an issue if we could not resolve nanosecond timing variations. It would also be impossible to make binaural recordings, which are very dependent upon phase accuracy.
Pre-ringing occurs near the cut-off frequency (Nyquist Frequency). If the cut-off frequency is 48 kHz (96 kHz sampling rate), the ringing is not an issue. At 44.1 kHz, the 22 kHz pre-ringing may be an issue if the listener has any ability to hear 22 kHz. It is more likely that the 22 kHz ringing causes IMD in the tweeter producing audible distortion at frequencies below 20 kHz. It should be noted that the duration of the pre and post ring decreases as the sample rate increases. So a 2X sample rate will shorter ring by a factor of 2 while increasing the ring frequency by a factor of 2. The combination of these two factors, give 96 kHz a significant advantage over 44.1 kHz (assuming that there is an audible defect at 44.1 kHz). The bottom line is that 44.1 kHz is so close to the limits of our hearing that is will cause audible problems unless the stars align perfectly. One small defect in a 44.1 kHz system can cause audible errors. In contrast, a 2X system has a much larger margin for error.
We’ll talk about IMD or intermodulation distortion soon.