First, I want to make it clear that I’m making many assumptions about Music V that are likely wrong. I have a mixed bag of fossilized Music V bones, and I’m doing my best Emily Deschanel impression to reconstruct them. If you are knowledgable of such things and catch an error, or would like to expand, please post a comment. With that said…
I had to read the Music V manual, but I finally found the bit-depth used for the examples in Mathews’ The Technology of Computer. 12-bit, just as I suspected. According the manual, the digital-analog-converters (DAC) at Bell Telephone Laboratories (BTL) operated with 12-bit samples. (pg. 168)
I don’t recall running across a 12-bit audio-file, ever. They just don’t make them like they used. I don’t even think Csound natively supports outputting a 12-bit file. However, it does allow us to easily simulate it with the 0dbfs opcode.
I rewrote part the CSD file from part 1 to better preserve the the original structure of the Music V example. Instead of rescaling the audio output signal of instr 1 by a factor of 16, I elminated this multiplier, and set the global dynamic range to 12-bit with 0dbfs = 2048. Original Music V amplitudes can now be used without modyifing the values or making unnecessaring changes to instrument translations. Here is the new CSD: Csound_Vs_MusicV_pt2.csd
Csound makes changing the dynamic range easy. In fact, almost all modern software allows users to choose from a list of several bit depths. Music V, not so much. When Music V was installed at a new installation, parts of the program would have to be coded, in assembly and/or Fortran, specifically for that particular system. Including the dynamic range. In theory, other computer facilities might have used a different dynamic range other than 12-bit, based on the hardware they had on site.
One interesting bit of information I discovered is something that could be considered a precursor to the modern audio buffer.
The General Electric GE645 computer at BTL used a 36-bit word length machine. In BTL’s particular setup, Music V would have to create 3 samples and collect them into a single 36-bit word before sending this word to the digital reel-to-reel. This isn’t too unlike the latency buffer found in programs such as Ableton Live, Max, Reaktor, etc., where n amount of samples are collected into a temporary buffer before sending the buffer to the DAC.
Hi,
Nice article.
You can also make sure that the output sounds 12-bit by making all output samples integers (removing the fractional part) using abs(). Since this will scale directly to 16 or 24 bit files you will be listening to the sound like it was meant to…
Cheers,
Andrés
Ahh, yes. Good catch. That’s going in the next revision.