View Single Post
  #20  
Old 2007-09-23, 12:15 AM
Tubular
 
Re: Reconsider ISO posting policy?

I'm not a taper but there are lots of 24 bit recorders that will provide real 48 volt phantom power, with no need for a separate preamp, including the SD 722 and 744, Tascam HD-P2, some Edirol models I think. I think the Sound Devices are regarded as the best.

http://en.wikipedia.org/wiki/Super_Audio_CD
Quote:
The process of creating a DSD signal is conceptually similar to taking a 1-bit delta-sigma analog-to-digital (A/D) converter and removing the decimator which converts the 1-bit bitstream into multibit PCM. Instead, the 1-bit signal is recorded directly and in theory only requires a lowpass filter to reconstruct the original analog waveform. In reality it is a little more complex, and the analogy is incomplete in that 1-bit sigma-delta converters are these days rather unusual, one reason being that a 1-bit signal cannot be dithered properly: most modern sigma-delta converters are multibit.

Because of the nature of sigma-delta converters, one cannot make a direct comparison between DSD and PCM. An approximation is possible, though, and would place DSD in some aspects comparable to a PCM format that has a bit depth of 20 bits and a sampling frequency of 192 kHz. PCM sampled at 24 bits provides a (theoretical) additional 24 dB of dynamic range. Due to the effects of quantization noise, the usable bandwidth of the SACD format is approximately 100 kHz, which is similar to 192 kHz PCM.

Because it has been extremely difficult to carry out DSP operations (for example performing EQ, balance, panning and other changes in the digital domain) in a 1-bit environment, and because of the prevalence of studio equipment such as Pro Tools, which is solely PCM-based, the vast majority of SACDs especially rock and contemporary music which relies on multitrack techniques are in fact mixed in PCM (or mixed analog and recorded on PCM recorders) and then converted to DSD for SACD mastering.

To address some of these issues, a new studio format has been developed, usually referred to as "DSD-wide", which retains standard DSD's high sample rate but uses an 8-bit, rather than single-bit digital word length, but still relies heavily on the noise shaping principle. It becomes almost the same as PCM (it's sometimes disparagingly referred to as "PCM-narrow") but has the added benefit of making DSP operations in the studio a great deal more practical. The main difference is that "DSD-wide" still retains 2.8224 MHz (64Fs) sampling frequency while the highest frequency in which PCM is being edited is 352.8 kHz (8Fs). The "DSD-wide" signal is down-converted to regular DSD for SACD mastering. As a result of this technique and other developments there are now a few digital audio workstations (DAWs) which operate, or can operate, in the DSD domain, notably Pyramix and some SADiE systems.
It says that a lot of modern AD converters start off as multibit, unlike a lot of older converters that were 1 bit/high sample rate.

From the Swedish article:
Quote:
One-bit converters for CD-players often use sampling rates between 11 and 50 MHz. The best one-bit converter probably is JVC's PEM-DD and it is much better than DSD. This said with reservation, I might have missed some even better one-bit technology than PEM-DD. But as far as I know this is the technology that comes closest to true multi bit technology in resolution.

The resolution/ information doubles when you double the sampling frequency (it is possible to be more specific, but for this example it is enough). But to double the resolution using PCM, you only have to add one more bit. If you go from 1 to 16 bits (adding 15 bits which use approximately 15 times more storage space), the resolution increases 65,536 times (from one step to 65,536 steps).

There is also another essential difference; the increase in resolution you achieve from raising the sampling frequency will be frequency dependant. A one-bit system will therefore have high resolution at low frequencies (where the information theoretically is low) and have low resolution at high frequencies (where the information theoretically is high).

By the use of noise shaping of high order, it is possible to increase the resolution at "quite high frequencies" at the expense of resolution at very high frequencies, but only for static, non transient signals. Transient signals will have poor resolution in a one-bit system. If the signal does not endure for a long enough time, the error will not be minimised by the noise shaper of the one bit system.

That's why you can read in documents from Burr Brown (who manufactures both one-bit and multi-bit converters) that you should use multi-bit converters for "waveform synthesis applications requiring very low distortion and noise". They have not written this for nothing.

A one-bit converter (i.e. the DSD system) cannot regenerate a short pulse with stringent form. It will change form from moment to moment. Every identical recorded pulse will show up with a new form.
Reply With Quote Reply with Nested Quotes