STONES SOUND STUDIO
By Bruce Hofer
Courtesy of Audio Precision
Sound Advice: Audio Test Q&A
Q: I'm experiencing jitter in a broadcast or home audio environment, but I can't identify the source.
Digital audio offers a lot: lower distortion, better dynamic range and immunity from hum and noise. It does have its own set of challenges though, chief among them jitter. For digital audio, jitter is the variation in time of the derived clock signal from nominal. Jitter can be introduced into a digital audio signal in two ways: in the sampling process, and in the digital interface.
Assuming your source material is good, the jitter must be coming from your interface. One very common source of jitter in broadcast is the result of a nonideal interconnection; typically, improper cables or very long cable runs.
Reactance in the cable or improper impedance can cause high frequency losses which result in a smearing of the pulse transitions. This would not be a serious problem if the effect were the same on every transition. That would just result in a small static delay to the signal that could be ignored. However, that would only be the case if the pulse stream were perfectly regular—a string of embedded ones or zeros, for example. But real pulse streams consist of bit patterns which are changing from moment to moment, and in the presence of cable losses these give rise to inter-symbol interference.
The proximity and width of data pulses effectively shift the baseline for their neighbors, and with the longer rise and fall times in the cable, the transitions are moved from their ideal zero crossings. The result is jitter.
As the AES3 interface uses the same signal to carry both clock and data, it is possible to induce jitter on the clock as a result of the data modulation. This means that care should be taken about mechanisms for interference between the data and the timing of the clock. The smearing of the waveform as a result of cable losses is one such mechanism.
Some suggestions when you face a long cable run:
1. Use one of the professional interfaces (the balanced AES3 or the unbalanced SMPTE276M). The consumer interface (S/PDIF) has a lower voltage and is not recommended for even moderately long runs.
2. Use the correct cable for the interface: quality balanced cable (shielded twisted-pair, 100 ohm) for AES3, or quality 75 ohm coax for SMPTE756M. Remember, the digital interface signal is in the megahertz range, even at moderate sampling rates. Using cables of the proper impedance, and terminating correctly are essential.
3. Consider an alternate transmission path, such as a microwave link or embedding the audio in SDI video.back to top >
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Bruce Hofer in EDN
Executive Editor Ron Wilson talks with Bruce Hofer about testing audio IC's. Read more at EDN's website: http://www.edn.com/article/CA6418209.html.