MADI magic

Jan 13, 2012 10:42 AM, By Ned Soseman

    
This PCM-1610 system sold in 1980 for about $40,000. It added internal time code to the original PCM-1600. Photo courtesy of John Moran Mastering, Houston, TX.

This PCM-1610 system sold in 1980 for about $40,000. It added internal time code to the original PCM-1600. Photo courtesy of John Moran Mastering, Houston, TX.

When discussing technical aspects of television, audio is often taken for granted. However, if you ever worked in radio, you know that television is just radio with pictures, right? With digital television, we tend to think of video stream transports, bandwidths measured in MHz and MPEG encoding schemes. The audio is simply a part of the media transport stream that everyone expects to be perfect.

Digital audio technology is a decade or two ahead of digital video. Prototype digital audio CDs were created around 1980. Of all the major manufacturers at the time, such as 3M, Soundstream, JVC, Mitsubishi and others, only the Sony+Philips consortium was working on a digital home delivery system. Philips began it as a 500-day timeline project and brought in Sony for its electronics, while Philips worked on the laser optics. It’s still the Philips Red Book spec that is followed for audio CDs.

At about the same time, Sony introduced the PCM-1600, the world’s first professional digital audio recording system. The PCM-1600 was the first system used for mastering audio CDs. It used digital conversion and pulse code modulation (PCM) in conjunction with a tricked-out Sony Broadcast U-matic for transport. The PCM-1600 weighed about 150lbs, and the companion BVU-200B U-matic wasn’t much lighter. I know because I helped deliver one of the first PCM-1600 systems to Stevie Wonder’s studios in Los Angeles in 1979. The “B” version of the BVU-200 was modified to move the head switch to the vertical interval and had the drop out compensator and chroma switched off. Typically, the BVU-200Bs were sold in pairs for editing. The biggest problem with the early PCM/VCR systems was tape dropouts.

A PCM adaptor converted analog inputs into digital audio, encoded as pseudo-video so the signal could be recorded on an analog video tape recorder. The number of NTSC video lines, frame rate and bits per line dictated a sampling frequency of 44.1kHz. This number was used because the Nyquist Thoerum states the sampling frequency must be double the audio bandwidth. 44.1kHz should also sound familiar because it is not coincidently the sampling rate for audio CDs. The audio CD was specifically designed to have a running time of up to 74 minutes so it could accommodate Beethoven’s 5th uninterrupted, as per Dr. Toshi Doi of Sony.

It didn’t take long for the Audio Engineering Society (AES) and the European Broadcasting Union (EBU) to take note of this trend. The result was the development of an international standard for two-channel PCM digital audio, formally known as AES3, also called AES/EBU. Note that one AES3 channel contains two channels of audio, often a stereo pair.

AES/EBU was developed to carry the two-channel audio data with an embedded word clock. Originally, Sony carried the data over separate wires with separate word clock, which was a problem for wiring. AES/EBU ins and outs were balanced with transformers so it could be transmitted over longer distances than an unbalanced signal, which degraded quickly. The unbalanced equivalent is S/PDIF, which is essentially a lower voltage unbalanced AES/EBU signal for consumer use.




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