Should New Master
Recordings be Analog, DSD or 24-Bit PCM?
The case for modern analog master recordings
With razor blade or cutting block in hand for physical editing, many engineers prefer to record in analog despite it being more labor intensive, as they don't believe even the highest resolution digital can equal the final sonic results, or they just like analog's sound better. One point in their favor is an analog musical waveform is how music is experienced by human beings, thus all digital must be converted to analog to be heard as music. So many of them reason, "Why not just keep it analog from microphone to the music lovers' speakers?" This is the argument for new LPs and 2 Track 15 IPS reel to reel master tape copies.
Some who deliver their final results in a digital format still prefer to record in analog, using high resolution PCM or DSD as the delivery format.
Cookie Marenco of Blue Coast Records is one of DSDs biggest supporters; however she prefers to record in analog. In her thirty years of experience she claims no PCM of any size has sounded as accurate to the source as two inch analog tape. She said at Computer Audiophile, "Coloration exits in all formats, and once you start comparing AD/DA converters, built in filters, inherent noise floor, reduced stereo image, and the lack of frequency response at the high and low end, tape wins by far. Tape vs. PCM is like looking through a glass window, (tape) vs. looking through a screen in front of the window (PCM). DSD digital comes very close to tape, but as much as I love it, it is not tape."
For 10 years Cookie Marenco has mixed all of her analog projects to DSD and claims DSD mix-down provides a respectable way to offer a near studio experience to her customers. She doesn't, however, drag her analog tape machine to remote gigs, she records direct to DSD instead. Based on the natural and realistic sound of her DSD downloads, which are among the finest recordings in my collection, I believe she is correct in her observations.
Tony Faulkner of Green Room Productions was originally convinced of the overwhelming superiority of digital recording, especially 24-bit 192kHz PCM and DSD. However, about seven years ago he had a change of heart and is now recording in both digital, including DSD and analog, using a pair of Tim de Paravicini rebuilt Studer A-80s
One of my favorite audiophile labels is Opus 3, which also records in analog, and they have recently made several of their recordings available as 2.8MHz and 5.6MHz DSD downloads, see my other article in this issue of Positive Feedback. In addition Opus 3 offers many titles on pure analog 2 track 15 IPS 10 inch reel to reels in addition to LPs, SACDs and CDs.
There are many other labels that also still record in analog by choice because they believe it still offers the finest sound quality available today.
Mike Spitz owner of ATR Services and ATR Magnetics says, "Analog's attraction lies in its ultra-high resolution capability, DSD is capable of 2,822,400 transitions per track per second, but a high-quality mastering tape contains approximately 80 million transitions per track second. And that's just for 1/4-inch two-track tape running at 15 IPS, the resolution goes up substantially with wider tracks and higher tape speeds." http://variety.com/2011/more/news/analog-recording-makes-a-comeback-1118029668/
The case for 24-Bit PCM Master Recordings
PCM is a method used to digitally represent sampled analog signals. The first consumer product that played PCM was the CD player; our concern in this article is not with this lower 16 bit resolution presentation but in the 24-bit versions that succeeded it.
A PCM stream is a digital representation of an analog signal, in which the magnitude of the analog signal is sampled regularly at uniform intervals, with each sample being quantized to the nearest value within a range of digital steps. PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that each sample can take.
Many recording engineers believe that 24-bit PCM at 96kHz, or higher, is the finest fidelity currently available to capture their master recordings. Some of the most outspoken supporters of 24-bit PCM include Mark Waldrep of AIX records who records at 24/96, Prof. Keith O. Johnson who records at 24/176.4, Barry Diament of Soundkeeper Recordings who records at 24/192, and Morten Lindberg of 2L who records at 24/352.8. The sonic quality I've heard with recordings on these labels makes a great argument for 24-bit PCM.
Another argument for 24-bit PCM is that digital tools that can edit are easily available and manipulate the master with relative ease.
The case for DSD Master Recordings
DSD uses pulse-density modulation (PDM) encoding. The signal is stored as delta-sigma modulated digital audio, a sequence of single-bit values at a sampling rate of 2.8224 MHz.
Practical DSD converter implementations were pioneered by Andreas Koch and Ed Meitner, the original founders of EMM Labs. Andreas Koch later left EMM Labs, and along with Jonathan Tinn founded Playback Designs, who have pioneered the transfer of DSD files over USB connections.
Computer language uses the binary numeral system and has two states: 1s and 0s. However, DSD has three states, and here is how it accomplishes that feat: DSD compares the previous slice of the musical waveform in the pattern buffer to the current one to determine if it goes up, down, or stays the same. "1" corresponds to a pulse of positive polarity, "0" corresponds to a pulse of negative polarity. A run consisting of all 1s would correspond to the maximum positive amplitude value, all 0s would correspond to the minimum negative amplitude value, and alternating 1s and 0s would correspond to a zero amplitude value. Thus, there must be two 1s in a row for the waveform to go up, or two 0s in a row for it to go down. If the DAC sees 10 or 01 the output stays the same. The continuous amplitude waveform is recovered by low-pass filtering the bipolar PDM bitstream.
One bit DSD is very noisy, so the benefit of such wide frequency response is the ability to push the noise into the higher ultrasonic regions, making DSD quieter than PCM in the audible range, this is called "noise shaping". After doing many plot spectrums on my SACDs I have discovered that DSD noise begins rising at 30-35kHz on some recordings, as low as 25kHz on others, and at 40kHz the noise is louder than any ultrasonic signal. A 50kHz filter is usually used on SACD players to keep this noise from adversely affecting up-line equipment, so in reality the maximum ultrasonic response to be heard above DSD noise for 2.8MHz DSD is around 35kHz. With 5.6MHz DSD, which is double SACD's resolution, it moves ultrasonic noise beyond 60kHz. The moving of ultrasonic noise further away from the audible range is what many feel gives 5.6MHz its superior sound quality.
It is true a lot of editing cannot be done in the DSD domain. However, there is what is called "DSD wide" which is 8 bit 2.8224 MHz. And editing facilities available without leaving the DSD domain are improving all the time, however due to the way DSD reads the musical waveform and transforms it into digital code, some editing and post production work may never be available while keeping the music file pure DSD. Some companies edit in high resolution PCM, some use high resolution analog tools, and a very few companies strive to keep the music file pure DSD and do as little editing as possible.
DMP "Direct to DSD" SACDs have no editing within a song or movement, as Tom Jung felt it destroyed the spontaneity and adversely effected the flow of the music. If an error or wrong note occurred they recorded the whole song all over. This is about as pure DSD one can get, and they sound smooth, extremely realistic, and thrilling.
Telarc and Heads Up "Pure DSD" recordings do preproduction engineering rather than postproduction, the level is set before recording, and that is what it stays at and what we hear with their finished product. They don't manipulate the signal in any manner, what we get is exactly what the microphones captured. In addition they do as little editing as possible. Michael Bishop explained how, only for the duration of the edit cross-fade (as short as 2ms) does the DSD signal go through the Sony E-Chip for editing processing. The music before the edit cross-fade and the music after is pure DSD.
I feel that many DSD recordings sound excellent, many not so good, and I think it has a lot to do with the way DSD is edited. Hopefully, in time, all the editing issues will be solved so DSD recordings never have to be converted to PCM or analog in postproduction.
The highest resolution 24-bit recordings speak well of PCM but in my system they don't quite equal the best DSD and analog recordings.
In my other article in this issue on Opus 3's DSD downloads I found the sonic differences in the "analog to 2.8MHz DSD" and "analog to 5.6MHz DSD" Opus 3 samplers large enough to prove to me that well made analog masters are likely superior to even highest resolution digital masters, and I wonder if 11.2MHz DSD would reveal even more resolution from the analog masters. Thus, in my opinion, in order to future-proof master recordings I believe they should be made on analog tape at 15 or 30 IPS per second.
I agree with Cookie Marenco that analog tape is still superior to high resolution digital including DSD. It's getting closer, and digital formats, especially computer music files, offer a convenience that analog never could. So my current ideal is well made analog recordings transferred to 5.6MHz DSD computer music files.