measurement solutions for all things audio

Audio.TST Archive

Audio Precision sends out the Audio.TST newsletter once a month to approximately 13,000 audio engineers and other audio professionals. Each issue looks at current events in audio test, answers readers´ technical questions and announces any AP news.


audio.TST April 2007

Notes from the Test Bench
Output
Sound Advice
Test Results


Notes from the Test Bench

By Bruce Hofer, Chairman & Co-Founder, Audio Precision

Hello Reader

Well, we all made it back from NAB safely and some of us even won some money!

The show tracked with our expectations as we saw a significant increase in demand for multichannel compared to last year, as well as a general trend towards combined audio/ video and the near ubiquity of digital over analog (though we did get one request for the wow & flutter option to remain available for the 2700 Series).

Alongside NAB, we also saw the first order for an APx586 16 channel analyzer just 4 days after availability was announced. Congratulations to Crestron Electronics and AP partner Technology for Measurement.

-- Bruce Hofer

PS: If you have a minute, watch this video of AP's booth being set up in record time: Goes to show AP really is the fastest in audio test!

Output: Tech tips and new applications from AP

Q: I would like to make a frequency response measurement sweeping from 20 Hz to 20 kHz with 100 log spaced steps each lasting 1 second for a total sweep time of 100 seconds (1 min. 40 sec.). How can I change the step time to 1 second?

A: You can use the Delay property in the Settling panel to achieve this.

  1. Setup the Sweep panel as you normally would for this measurement.
    (See Figure 1)


  2. Open the Settling panel (select Panels > Settling) and set the Level A Algorithm to None.


  3. Set the Level A Delay to 1 second.

  1. Set the Level A Points to 1.
    (See Figure 2)

Figure 1


Figure 2

back to top >
Sound Advice: Audio Test Q&A

Q: I'm experiencing jitter in a broadcast environment, but I can't identify the source. Any suggestions?

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 >
Test Results: AP News & Events

122nd AES Convention Vienna -- May 5-8
Booth No. 1422 Visit the show website
AP will be sharing a booth with our German / Austrian Sales Partner, RTW. We look forward to seeing you there.

IBC Amsterdam -- September  7-11
Booth: 8.410 Visit the show website

123rd AES Convention New York -- October 5-8
Visit the show website

 

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.


back to top >

 

back to top >