Old Formats Never Die Part 1of2

By Roger Nichols

Samples and Bits

It seems like I am getting more and more material from the early days of digital that needs to be re-mastered or re-mixed. The formats I am dealing with are Sony PCM-10 stereo, PCM-F1 stereo, PCM 1610 stereo, PCM 3324 24-track, Mitsubishi X-80 stereo, Mitsubishi X-800 32-track, 3M 32-track, 3M 4-track, Sound Stream 8-track, and DAT stereo. There are some quirks in these early formats that you should know about and be able to correct for. Sample rates, bit depth, DC content and pre-emphasis were basically ignored in the ‘80s, so we have to deal with it now if we want to salvage the recordings for archiving or re-release 20+ years later.

Sample rates were 44.056kHz, 44.1kHz, 50.4kHz, 50kHz, and 48kHz, bit depth were 14bit or 16bit, pre-emphasis was usually turned on, and DC ran rampant in digital recordings. In dealing with these tapes, we have to watch out for tapes with some or all of these characteristics and correct for them when updating the format for current use. Here is some background on these parameters.

Sony’s early stereo PCM digital recorders were based on videotape machines for digital audio storage. The data was recorded as black or white dots the videotape scan lines. The same data was duplicated on the odd and even fields of the video image. Since video ran at a frame rate of 29.97 frames per second, the data rate ended up being 44.056kHz. If you ran the video machine from a 30-frame source, the data rate was 44.1kHz. 44.1kHz was adopted for the CD sample rate because it was the common denominator between PAL video at 25 frames per second and the pulled up NTSC video at 30 frames per second. Early machines like the Sony PCM-10 used 29.97 as the clock, so those machines recorded audio at 44.056kHz. The PCM-10 was a 14bit recorder.

When the Sony PCM-F1 was introduced for consumer digital recording around 1982, the NTSC versions recorded at 44.056 and the PAL versions recorded at 44.1kHz. Many companies jumped on the consumer digital bandwagon and built videotape based recorders. The standard was for 14bit recording, although Sony processors (and Nakamichi, made by Sony) allowed you to switch to 16bit recording with a loss in error correction. There was only so much room on the videotape for those black and white spots, so you had to make a choice between error correction and resolution.

When professional machines started showing up on the market it looked as if there was going to be a sample rate war. 3M offered a 32-track and 4-track machine that recorded at 50kHz, 16bit. Mitsubishi introduced the X-80 2-track machine at a sample rate of 50.4kHz, 16bit. Mitsubishi also introduced a 32-track machine at 48kHz, 16bit. Soon after that, the industry decided to standardize the professional sample rate at 48kHz, with the consumer standard set at 44.1kHz to match the CD sample rate. Until 1989 there was no such thing as a sample rate converter, so sample rates had to be the same in order to transfer audio from one format to another digitally.

Remember that it is possible to play back a 44.056kHz tape at 44.1kHz, it will just be a little sharp and fast. You can play back a 50kHz tape at 48kHz, it will just sound flat and slow. That is, if you have a reference. It is possible that a song was sped up or slowed down during the original production, but it is doubtful that all of the songs of an album were speed changed by exactly the same amount. Play back the source at another sample rate and compare it with a live piano or other instrument to see which sample rate is most plausible as the intended original sample rate. On many of these early projects the sample rate was not written on the box, because there was no other choice and other sample rates had not been invented yet. You never saw an analog tape labeled “NON DOLBY” before Dolby noise reduction was invented.

The same thing is true regarding bit depth. All consumer digital tapes were 14bit except Sony. Nobody ever labeled a tape 14bit. Use a meter that shows actual bit activity to properly determine the bit depth of a digital signal.

Emphasis

Emphasis came about because of early converter design. The entire sampling process was new, and A to D converters exhibited low level noise because of bad linearity in the conversion process. This process added some high frequency broadband noise to the digital signal. Manufacturers overcame this byproduct by boosting (emphasis) the high frequencies during the conversion from analog to digital, and then rolling off (de-emphasis) the high frequencies by the same amount after the conversion back fro digital to analog. This process was optional and there was a switch to select emphasis on each track during record. A flag was set in the digital bit-stream, which automatically activated de-emphasis during playback. All CD players, DVD players, and DAT machines detect this flag and turn on a high frequency roll-off in the analog domain during playback. If the digital signal contains emphasis and the flag is missing or turned off, then the roll-off does not occur and the audio will be brighter than normal.

This emphasis “feature was the biggest reason why different CD players sounded different when playing back the same CD, or DAT machines differed playing back the same DAT tape. The digital part and the conversion to analog were basically the same in all of the machines. The de-emphasis circuit was implemented in the analog domain using the least expensive circuit to perform the operation. There was high-end EQ on the output of every digital playback device, and there was no standard or calibration for how it was performed. If you played back a CD without emphasis, then all of the CD players sounded pretty much the same. If you played a CD with emphasis, then each playback device sounded very different from every other player.

Producers and engineers started turning off the emphasis switches. Converters were getting better so there was less converter noise, and the use of de-emphasis circuits was eliminated.

DC

PCM (Pulse Code Modulation) audio stores data as a 16 bits (or higher) of information. 15 of the bits are used to indicate the level of the sample above zero. The 16^th bit is used to indicate whether the sample is positive (above zero) or negative (below zero). The smallest positive amount was encoded as 10000000 00000000. The smallest negative value was encoded as 01111111 11111111. So, right at the quietest part of the signal, as the signal is crossing zero, the left bit that signifies the biggest bit is changing from 1 to 0 at the same time as all of the other bits are changing from 0 to 1. If the converter is not perfect, there will be a click at the zero crossing point known as zero crossing distortion. You can hear this easily and it is not good sounding.

In order to get rid of this problem, manufacturers added in a DC signal so that the zero crossing point in the converter is not at the same place as the zero crossing point in the recorded waveform. Now any distortion is happening when the signal is louder, and is masked by the audio. In DAT machines and modern converters and DC offset used is removed from the signal automatically in the digital domain so you don’t have to worry about it. In early recorders, this was not the case. Sometimes the DC level was cranked up so high that you had to turn down the record level. The positive waveforms would clip as much as 18dB before the negative side of the waveform.

This DC level was not removed in early CD production. Have you ever played back an old CD and heard a pop in the speakers when you skip from track to track? Try it, especially if you have an old Telarc CD from the ‘80s. Don’t turn it up too loud, the DC pop can blow out your speakers.

DC and Emphasis and DAW

DC is easy. You can see it in the waveform display. It looks like the waveform is shifted above zero. If you make an edit, there is a pop when you go in or out of the region. Pro Tools has an Audio Suite plug-in specifically for DC removal. The same tool is available for other DAW systems. Just run the audio through the plug-in and the problem is cured.

As far as emphasis goes, I have not found a de-emphasis plug-in, but there are some hardware and software “cheats” available.

Most of the Digital Audio Workstations ignore emphasis flags in digital audio streams. That means that if you transfer digital audio that indicated emphasis into a Pro Tools system, the flag will be stripped, but the signal will not be de-emphasized. It played back correctly from the DAT machine, CD player, PCM-1630, or Mitsubishi X-86, but when you play it back in Pro Tools it sounds bright enough to cut the top of your head off.

The emphasis problem should be corrected during the transfer into your DAW. At this point the playback machine will have an LED indication that says EMPHASIS. The signal is de-emphasized at the analog outputs, but not at the digital output. There are three solutions.

1)    Record the audio into your DAW using an analog connection. (Don’t tell anyone I suggested this, as I will deny it.) You do run into the problem we were talking about earlier. Different machines will sound different playing back the same material. If you are just dumping in some sound effects or something of low importance, then just do it.

2)    Run the digital audio signal through a box that will de-emphasize in the digital domain. One such box is the Roland SRC-2. This box is no longer made, but I have seen them on e-Bay and in some Hollywood pawn shops. It automatically detects the incoming emphasis flag. If you select EMPHASIS-OFF for the output, the de-emphasis will take place in the digital domain and the flag cleared. The box will also remove DC, perform sample rate conversion, and add in another 2-channel digital input to your original signal if you want.

3)    Cheat. Label the track as having EMPHASIS when you import into your DAW. Now, use a plug-in EQ on the track. I have not taken the time to figure out the settings with all EE, but Waves Q4 is what I usually use. Set the highest band as HI SHELF, set the frequency at 5000, and set the level for that band to –9.0. I checked it with an Audio Precision and it exactly eliminates the emphasis curve.

Conclusion

As you can see, it is not always a “piece of cake” to re-purpose old recordings. Just because the original is digital doesn’t mean that it will play back the way it was meant to. Remember, “Murphy is my co-pilot.”