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T.RayBullard
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12th October 2006
Old 12th October 2006
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Talking new Korg portable DSD recorders..

http://www.korg.com/gear/info.asp?a_...&category_id=3










Future-proof digital recording is here
With the pristine fidelity and ultimate flexibility of 1-bit technology, the new, surprisingly affordable MR-1000 and MR-1 ensure that your final mixes and location recordings are never obsolete.

Uncompromising, future-proof fidelity at an unprecedented price
Korg’s new, super-portable MR Professional Mobile Recorders break new ground by providing high fidelity 1-bit recording technology at unheard of low prices. The hand-held MR-1 provides high quality 1-bit/2.8 MHz recording and playback, while the mobile/tabletop MR-1000 delivers an astonishing 1-bit/5.6 MHz, doubling industry DSD recording quality standards.

Powerful new software bundled with each MR recorder enables the conversion of your 1-bit recordings and mixes into all of today’s audio formats without degradation. Best of all, by archiving your data in 1-bit format you’re ready for the future, as playback systems and standards change.

The MR Professional Mobile Recorders are ideal for source recording, podcasting, broadcast journalism including ENG/EFP, as well as archiving final mixes and master recordings.

1-bit technology equals pristine fidelity
1-bit recording is the latest advancement in audio, and has been adopted for use in the critically acclaimed SACD recording format. It offers a frequency response of DC to 100 kHz and dynamic range of 120 dB. This uncompromising fidelity, low noise floor, extended dynamic range, lifelike imaging and analog quality depth has been praised by top experts. But there are other important advantages to the format that are of benefit to all recording professionals, regardless of their tracking platform.

Future proof flexibility
Today’s state-of-the-art 24-bit converters use high-speed 1-bit conversion to capture audio, using real-time decimation and other processes to present the data in the desired bit depth/sample rate format. The beauty of the MR’s bitstream format is that it uses the original 1-bit data, without the need for the other processes. What comes in comes out, with no manipulation needed.

As files are converted and reconverted between various bit depths and sampling rates, there are possibly degrading effects, depending on the sample rate conversion algorithms. The critical issue is that files mixed and mastered in today’s state of the art resolution may be insufficient for tomorrow’s formats. Archiving your final mixes and masters in a 1-bit system allows you to bypass these issues, and preserve your music with both the highest fidelity and in a more “universal” format. That data then can more easily be converted at a later date to the bit depth/sample rate format of your choice without compromising the integrity or fidelity of the data. You can even convert to newer formats that will be adopted in the future, be they multi-bit or a further migration to the 1-bit format.

Free integrated software solution
Both MR recorders come with Korg’s innovative and powerful AudioGate™ software for Mac® and PC. AudioGate can convert 1-bit recordings into WAV and AIFF formats at various bit-rates (and vice versa) and offers real-time conversion and playback of 1-bit files. It also does essential functions like DC offset removal, gain control, and fade in/out. The combination of these recorders and this exclusive software makes the perfect system for both capturing and preserving your critical projects and source recordings. This archiving capability that had only been the domain of the major record companies and top studios is now available for everyone!

Real world features
Each recorder supports multiple recording formats including DSDIFF, DSF, and WSD 1-bit formats, as well as multi-bit PCM format (BWF) with resolutions up to 24-bit/192 kHz. They both feature internal 20 Gbyte hard drives for ample recording times, and USB 2.0 connection for fast and easy transfer of files between the recorder and your computer.

MR-1000
The MR-1000 is the perfect tool for the professional who wants to record and archive their final mixes in the studio, while its compact size and portability make it ideal for location recording. Whatever the use, you get the benefit of capturing it at the new high 5.6 MHz rate. After recording and editing all your tracks in the DAW/hardware system of your choice, you can mix directly, or via an analog summing mixer, to the MR-1000. This gives you the superb fidelity of high-rate1-bit technology, which when transferred to your computer can be converted to the mastering format of your choice thanks to AudioGate. And backing up in 1-bit format “future proofs” your mix for potential reuse in the future.


MR-1000 Back Panel

The table-top MR-1000 includes combination XLR/ 1/4" input connectors with top quality microphone preamps, phantom power and built-in limiting. Both XLR and RCA outputs are provided for maximum flexibility. High speed USB connectivity allows easy file transfer to and from a computer. The MR-1000 runs on AC power or via AA batteries for mobile freedom.

MR-1
The MR-1 is the ultimate portable 1-bit recorder, perfect for location recording, broadcast journalism, live music performances – even for rehearsals and song-writing sessions. By recording in 1-bit/2.8 MHz format you are assured that your most important, once-in-a-lifetime moments are captured in stunning detail, and ready for whatever the future brings you. And back in the studio it also provides superb final mix and archiving benefits.


MR-1 Top and Side views and
stereo electret condenser mic.

The ultra portable MR-1 includes dual balanced mini plug inputs and a stereo mini plug output. It includes a stereo electret condenser mic so the unit can be tucked away and the mic clipped or placed in a convenient location. High speed USB connectivity allows easily file transfer to and from a computer. The MR-1 runs on AC or long-life rechargeable lithium polymer battery.

Future Proof Recording Explained
For a complete overview of current PCM digital audio techniques and some of the principals and advantages that 1-bit recording provides, both in terms of fidelity and archiving for future use, read Future Proof Recording Explained.



Specifications subject to change without notice
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12th October 2006
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Looks cool. I wonder how good the preamps and converters sound.
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12th October 2006
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Hey i've just been to the korg forums reading what they said about this Machine , the idea behind it seems great ,Much more affordable DSD technology for the small guys, What are your thoughts T Ray, gonna get one?.
Will be for sale March next year, i think this may really open things out a lot.

Sinewave.
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12th October 2006
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Quote:
Originally Posted by Sinewave View Post
Hey i've just been to the korg forums reading what they said about this Machine , the idea behind it seems great ,Much more affordable DSD technology for the small guys, What are your thoughts T Ray, gonna get one?.
Will be for sale March next year, i think this may really open things out a lot.

Sinewave.
well, ive already got the Tascam DV-RA so I am set for now..I normally run a mixdown from my main multitrack recorder into the tascam in DSD mode. i might get one though..it has piqued my interest for sure..
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12th October 2006
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worth a listen I reckon..

PS: I love future proof objects..
My store room is FULL of them
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12th October 2006
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I think the software that comes with it really caught my eye, and even though the Tascam comes with the discwelder software, this new audiogate that will come with the Korg seems to make things easier and convinient. And it works i Both Mac and PC.
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12th October 2006
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I didn't even consider checking the Korg booth when I was at the AES in SF.

Apparently it's my loss. Damn!
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13th October 2006
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Checked it out at AES... A couple comments:

First of all, decent build quality, the big one looking considerably better than the little one. The little guy seeming more like a toy than a "real" piece of gear.

The sound on the demo box was quite good. Supposedly it was recorded using 2 mics directly into the mic pres that are onboard. If that is the case, than it is a pretty impressive sound... You never really know, though... Records native DSD at a reasonable price then you can transfer to a computer to resample (software provided) to PCM via USB.

My only gripes with the bigger box- no digital inputs for PCM recording. Common, a SPDIF input really shouldn't be too much to ask for (I've got the same gripe about the new Fostex flashcard recorder).

The internal disc is also pretty small- only 20 GB. Not bad for many gigs, but kind of precludes you bringing it out for long festival dates where you record a couple days in a row pretty much without break. At the end of those crazy days, the last thing you want to do is to grab a laptop and hard drive and start transfering tracks so you can record the next day.

Anyways, my thoughts on it based on what I saw at AES.

--Ben
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13th October 2006
Old 13th October 2006
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MR1000 potential

One thing that intrigues me, and I may be all wrong on understanding the theory, is whether or not it can convert audio at a higher, less-colored standard than the top-end converters, such as Lavry, Weiss and so on, for the following reasons:

According to Korg's white paper on the MR 1000, it uses "State of the art Burr Brown PCM4202's for the AD conversion and then eliminates the decimation filter process and the anti-aliasing filter step which PCM converters go through in the AD stage, and stores the data directly at 1 bit, 5.6448. The filtering of the original 1 bit data stream by PCM converters is what is responsible for the coloration or differences we perceive in converters due to considerations of phase, linearity, transient response and ripple, all subject to the math being done".

If one would go analog line-in, from a high-end signal path, (mic to mic-pre to the MR1000) or from a mix bus, wouldn't you get a truly uncolored, transparent recording with these PCM steps out to the picture, theoretically speaking? Transfer the files to your DAW for editing via USB2. Seems like it would be ideal for capturing stereo for critical acoustic, jazz or classical recording, just using the built-in converters.

Some have mentioned no digital inputs, but what about it's own conversion ability? If it did indeed, convert at a highly tranparent, uncoloured rate it could be very useful for more the ENG, foley and so on. It's got me interested for sure.
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13th October 2006
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Quote:
Originally Posted by RickCording View Post
One thing that intrigues me, and I may be all wrong on understanding the theory, is whether or not it can convert audio at a higher, less-colored standard than the top-end converters, such as Lavry, Weiss and so on, for the following reasons:

According to Korg's white paper on the MR 1000, it uses "State of the art Burr Brown PCM4202's for the AD conversion and then eliminates the decimation filter process and the anti-aliasing filter step which PCM converters go through in the AD stage, and stores the data directly at 1 bit, 5.6448. The filtering of the original 1 bit data stream by PCM converters is what is responsible for the coloration or differences we perceive in converters due to considerations of phase, linearity, transient response and ripple, all subject to the math being done".

If one would go analog line-in, from a high-end signal path, (mic to mic-pre to the MR1000) or from a mix bus, wouldn't you get a truly uncolored, transparent recording with these PCM steps out to the picture, theoretically speaking? Transfer the files to your DAW for editing via USB2. Seems like it would be ideal for capturing stereo for critical acoustic, jazz or classical recording, just using the built-in converters.

Some have mentioned no digital inputs, but what about it's own conversion ability? If it did indeed, convert at a highly tranparent, uncoloured rate it could be very useful for more the ENG, foley and so on. It's got me interested for sure.


Yup you may be right, it is after all a final mixdown , archiving and source recording device, i dont see the need for digital inputs for PCM when it has it's own DSD converters, and besides you can do your PCM mixes in your normal workstation of choice then convert it to DSD via the software or record your final mixes using some sort of analog summing directly into the inputs of the Korg.
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19th October 2006
Old 19th October 2006
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This thing have timecode for those remote shoots?
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19th October 2006
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Quote:
Originally Posted by Sinewave View Post
Yup you may be right, it is after all a final mixdown , archiving and source recording device, i dont see the need for digital inputs for PCM when it has it's own DSD converters, and besides you can do your PCM mixes in your normal workstation of choice then convert it to DSD via the software or record your final mixes using some sort of analog summing directly into the inputs of the Korg.
this is exactly what I do with my final mixes..through a meitner ADC into the tascam.

teddy
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19th October 2006
Old 19th October 2006
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I really like the idea of having a recorder like this or a tascam, but I am not confident in my panning/mixing abilities at all, and I wouldn't want to "commit" to a stereo recording. But I would love to have the portability of this rather then taking around my desktop. I guess I am just paranoid I will ruin recordings, and panning is alot easier then I think. If you planned on using 4 mics I guess you would also need to invest in a good mixer? Or does everyone pretty much only use these for doing stereo pair recordings where mixing isn't a factor?
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19th October 2006
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Quote:
Originally Posted by jakromm View Post
I really like the idea of having a recorder like this or a tascam, but I am not confident in my panning/mixing abilities at all, and I wouldn't want to "commit" to a stereo recording. But I would love to have the portability of this rather then taking around my desktop. I guess I am just paranoid I will ruin recordings, and panning is alot easier then I think. If you planned on using 4 mics I guess you would also need to invest in a good mixer? Or does everyone pretty much only use these for doing stereo pair recordings where mixing isn't a factor?
I did a multimike recording and mixed down to stereo to a CD recorder live the other day and it worked out really well, luckily it was friends i was recording. It can work, you just need to take your time and think about things, besides think of it as an oppurtunity to improve your engineering skills.
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19th October 2006
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Originally Posted by jakromm View Post
I really like the idea of having a recorder like this or a tascam, but I am not confident in my panning/mixing abilities at all, and I wouldn't want to "commit" to a stereo recording. But I would love to have the portability of this rather then taking around my desktop. I guess I am just paranoid I will ruin recordings, and panning is alot easier then I think. If you planned on using 4 mics I guess you would also need to invest in a good mixer? Or does everyone pretty much only use these for doing stereo pair recordings where mixing isn't a factor?
what I do is mix everything down in Samplitude/sequoia, then send that final mix to the tascam....I dont use it while the recording is going on..i Use afterwards for archival..
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19th October 2006
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Quote:
Originally Posted by SiliconAudioLab View Post
This thing have timecode for those remote shoots?
A timecode recorder under $1K? Nah... Only has analog inputs. My beef is that it doesn't even have digital ins.

--Ben
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19th October 2006
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Quote:
Originally Posted by fifthcircle View Post
A timecode recorder under $1K? Nah... Only has analog inputs. My beef is that it doesn't even have digital ins.

--Ben
For me, if it keeps the price down, I'm glad it's only analogue! The fact that it has USB means you can get stuff into it from most places...
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19th October 2006
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A timecode recorder under $1K? Nah... Only has analog inputs. My beef is that it doesn't even have digital ins.

--Ben
the tascam HDp2 is around that price and has timecode..
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19th October 2006
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Quote:
Originally Posted by T.RayBullard928678
the tascam HDp2 is around that price and has timecode..
I was not impressed with this unit after working with it for a week. The firmware needs more work. The sound was adequate. I hope the Korg is better!
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19th October 2006
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What´s advantage on record on DSD? As I understood, you need to convert it to PCM to can edit it..
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20th October 2006
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Originally Posted by lagoausente View Post
What´s advantage on record on DSD? As I understood, you need to convert it to PCM to can edit it..
some desks(like the sonoma workstation) edit DSD as is..and Sadie, I believe..and Pyramix(or does that one downsample)

as for DSD..the benefit is the SOUND...just loads better..IMHO..of course.
======================================================
DSD Vs PCM from the Chief Engineer at Phillips



Studio people who have compared the live misc feed to DSD and PCM say that DSD is much better and they cannot tell the two apart. If, as you say, it is flawed, why are the studio engineers all for it. As someone on the forum said, it is analog without all the problems.

BP:
1) Sound of DSD:

The HF noise and the low-level nonlinearities of DSD do not get in the way of the sound as we hear it. However, it is impossible to build a production chain using the format.

DSD is in its place in one (1) application: Mastering from analogue. When the recording chain is completely analogue, you can feed the audio from the analogue mastering into a DSD A/D converter and cut that signal straight onto a disc without any further processing. It is in this application that DSD can be viewed as pretty transparent. When you convert the signal twice, however (such as when using a DSD recorder as the tracking medium), the second conversion is no longer transparent, due to the HF noise present in the source signal hitting a second analogue deltasigma modulator.

In this I have a serious gripe with AudioQuest. Before DSD, they tracked onto a 15ips 1/2" two track. These tapes were first used to master to ordinary CD, later again to SACD for a reissue, which indeed was a sonic forward step. However, they then switched to using a DSD recorder for tracking as well. Grundman then mastered it as usual (ie through an analogue mastering studio) onto the DSD master recorder.
The recording guys may well have found the DSD recorder better than their analogue two-track in a dry shoot-out, so you can't blame them for having made that choice. The resulting SACD releases however, are below anything. What's worse, there's no way of salvaging the recordings for a better-sounding reissue (unless they had the analogue two-track running as a backup). As an example, get hold of the classic "BluesQuest" sacd, which was made from analogue tapes. Then compare Doug McLeod's "Whose truth whose lies", where a DSD machine was used. If you're a bit of a sensitive person you'll run away screaming.

I think we could say that DSD is analog with a few extra problems. Serious ones.

2) Total transparency:

Apart from that I consider any claim of "not being able to discern between a live feed and DSD" as something of a hyperbole. While flicking a switch during the music will indeed not reveal any serious deficiencies, more controlled listening (e.g. listen - rewind - switch - listen) will certainly get you hearing a lot more. I can't even insert a unity gain stage (low-noise low-distortion etc) in the signal path without hearing degradation, let alone a pair of converters.

3) Jitter:

When a signal is fed straight from an ADC into a DAC, they share the clock. When you sample a signal with some jitter, and you reproduce it with the *same* jitter, the jitter has NO impact on the sound (jitter which impacts the sound most, incidentally is of the LF kind, which does not affect noise specs, and which is typically not attenuated by PLLs). With DSD this works - the delay between ADC and DAC is nearly zero. With PCM it doesn't, because there is a delay of several milliseconds between the two, meaning sampling and reconstruction see different jitter. This means that a live vs DSD will always sound more transparent than when you take the digital signal to tape and replay it.

4) PCM implementation issues:

On www.nanophon.com you can read a number of the late Julian Dunn's excellent papers on how compromised implementation of digital filters account for many of the deficiencies noted with PCM. These can be readily solved with due care for details and at the expense of only mildly increased computational burden.


TH:
How do you feel about the DSD workstations? The existing ones, I believe, are all using DSD-wide, or PCM-narrow.

BP:
The workstation from Merging is in itself OK in that the data is kept in 352.8kHz/32bit floating until it is written to an sacd master file. Currently the incoming data is DSD, causing a "two DSD conversions" problem (although apparently in the digital domain the sonic degradation is less than when it happens in the analogue domain). If you can somehow get the data in through a non-DSD converter (preferably at 352.8kHz/24bit or so), it would become a digital equivalent of the "ideal analogue chain" in that only one deltasigma coder is in the signal path (when writing the master). That would be fine.

If you do have to edit a recording which is already in DSD, it can be done in a sonically transparent way by using the "transport" mode of the workstation. In this mode, the DSD data is preserved exactly except during the edits (crossfades). The two conversions problem would be restricted to those edits only. Of course, any form of processing, such as level change, EQ or mixing, is not possible in this mode. It's really just cutting and splicing.

The workstations from the Sony camp (I don't know names) use 2.8224MHz, 8 bits, still noise shaped. This still has the same problem as reconverting to DSD every time, but reduced by 48dB. Still I wouldn't recommend it. The Merging approach is the most suitable one.


TH:
You mention the use of a non-DSD converter (preferably at 352.8kHz/24bit or so). EE Dan Lavry argues that the bigger the numbers the less accurate the conversion is. He says that there really is no need to go beyond 24/96 (48k bandwidth) and anything more will not result in a more accurate signal but rather more noise and distortion.

BP:
Dan is correct in that as the sampling frequency is increased, the available signal to noise ratio inside the nyquist band decreases. However, when you keep the bandwidth across which you are measuring SNR constant (e.g. you measure noise across 20kHz) and when noise shaping is used, this trend is often reversed. DSD is precisely a case in point. I have one converter here (homegrown discrete circuit) that puts out 1 bit at 2.8224MHz. Measured across its nyquist bandwidth (1.4112MHz), its SNR is useless, well below 6dB. However, taken across 20kHz, it delivers the full 120dB. When you push the sampling rate too high, performance will again resume a downward trend.

Dan's converters are multibit, non-noise shaping. Such converters will not even hold their precision at constant bandwidth when sampling rate is increased. Since his converters have ultra-low noise as their hallmark, he's quite right to maintain a reasonable sampling rate.

When I propose to use a 352.8kHz converter, in practice that would be a noise shaped converter designed to offer maximum SNR performance over as wide as possible a band, but not up to 176.4kHz (not feasible). It would still have a "noise shaping tail" in the nyquist band, but much less so than DSD. At the final DSD conversion stage, the noise of the ADC would be negligible compared to the DSD noise, so the output spectrum would be pretty much the same as that of an analog signal converted to DSD in one go.

A design which is in the work at home uses 64fs, 16level PWM with 7th order noise shaping. This would offer up to 129dB SNR (limited by amplifier noise) over 80kHz, which is 4 times as wide as DSD. This would insure all the flexibility of PCM while a later conversion to DSD is not compromised.


TH:
How do you feel about DSD as an archiving format? Is the transparency good enough for Sony's precious analog master tapes? And what makes DSD ideal for archiving as opposed to PCM? Or does it matter? I read Neil Young chose hi-res PCM format for his masters.

BP:
When sony came up with DSD as an archiving system there was hardly even 96kHz PCM around. If at that time they had some old tapes to archive before they fell apart, DSD was the best available. However, since DSD is a liability in terms of processability, archiving to DSD now is no longer a good idea and use of 192/24 is warranted instead. Since SACD is probably here to stay we should view DSD as strictly a release format, in the same way as we didn't produce on vinyl, but music was brought to the home on it.


TH:
What do you say when you hear audiophiles make comments that DSD has all the "air" and "smoothness" of analog?

BP:
In my own experience, high speed PCM also produces the air DSD has, while the "digital glare" of some PCM can even be solved at low speeds. It is caused by the narrow alias-band that is present between 20 and 24.1kHz. Removing this band prior to playback restores naturalness and focus.

I don't suspect DSD of any "euphonic" effects, although, who knows, the HF noise?

Admittedly, the DSD camp has been able to mobilise more audiophile designers (folks like Ed Meitner), resulting in the analog circuitry of the best DSD converters sounding better than that of most available PCM converters. Doing a straight shoot-out is actually quite challenging technically, as the two formats normally use different converters, necessarily producing a different sound. This is another reason for me to do this 352.8kHz converter, because its output can be converted to either DSD or 192/24 while compromising the performance of neither. This would finally allow a direct comparison.


Gardo:
Are the production chain problems insoluble, or not yet solved?

BP:
The "production chain problem" ie. the fact that signal quality quickly and irreversibly deteriorates as it passes through subsequent processing stages, is in itself not solvable. The root cause lies in the high HF noise level. This noise is indistinguishable from the wanted signal, so it cannot be stopped from accumulating every time a signal is converted to 1-bit. This is not to say there are no workarounds. However, the mere fact that such workarounds are necessary shows that the DSD format itself was misconceived.
1. "Production DSD". To use a 1-bit signal at 128fs or 256fs instead of 64fs. Especially in the case of 256fs, signal and noise no longer overlap. The signal bandwidth is specified, as before, at 70...80kHz. The quantisation noise only comes out of the analogue noise floor at 80kHz as opposed to 20kHz in the case of 64fs. Now, signal and noise can be separated, quite simply by filtering. It follows that every processing step still involves filtering to remove the HF noise from previous conversions and deltasigma modulation to recode the processed signal into 1-bit. This constitutes a considerable processing overhead.
2. PCM-narrow. To use 352.8kHz, 32bit floating point as the intermediate format. This is PCM by all means, but according to the listening tests carried out by Philips and Merging, the decimation and upsampling filters needed to convert to and from 2.8224MHz do produce audible artefacts. The advantage here is that this format is practical enough to maintain throughout production, store on hard disk etc. Converting from DSD to PCM-narrow is done only once, namely as the data comes into thr production chain. Conversion to DSD is only done once, namely at mastering. Moreover, the 352.9kHz data can be derived straight from a better-than-DSD AD converter (which means most of all present day converters), burdening the production chain with no or greatly reduced HF noise.
3. DSD-wide. 2.8224MHz, 8 bit. This format still requires all processed data to be noise-shaped back, but to 8 bits instead of 1. This can be done many times before noticeable headroom reduction or other adverse effects set in.
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#22
20th October 2006
Old 20th October 2006
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Gardo:
If the problems with PCM are easily solved, why haven't they been? That's a real question. If I understand DP, he's saying that removing aliasing effects in the 20khz-24.1khz band takes care of digital "glare." How would this be accomplished? Better brick-wall filters? Aren't those aggressive digital filters part of the problem?

BP:
In order to "solve" the PCM problems, it is required that a certain amount of care is given to all steps in the production chain. It is currently no problem to build a set of AD/DA converters that avoid all of the pitfalls in PCM. However, until everyone takes the same actions to get their digital filters right, these issues can reappear at any time.
Here's a concise list of the problems:
1. Half-band filtering: The upsampling (interpolation) and downsampling (decimation) filters are often specified such that the frequency response is symmetric around fsh/4;0.5 (on a linear amplitude and frequency scale. fsh=2fs). The effect of this symmetry is that every second coefficient is 0, except for the middle one, which is unity. This halves the number of computations necessary to implement a filter of the same steepness. However, the symmetry means that the response at 0.5fs is -6dB, a far cry from the total rejection called for by the nyquist theorem. Most if not all of PCM implementations actually do not conform to the nyquist criterium.
Now here's some funny psychology. Looking at the data sheets of converters you'll find that the passband goes to 0.4535fs, so the stopband starts at (1-passband) ie. 0.5465fs. At 44.1kHz, this translates into twentythousand point zero Hz. Apparently the belief must have been that as long as everything is well until 20.0k, all is cool and nobody will care about aliasing between 20.0k and 24.1k.
2. Equiripple filtering: These days chips are fast enough to run longer filters, so the half-band filter has gone somewhat out of fashion. One thing in which half-band filters are seen to be something of an overkill is in passband ripple. Due to the symmetry, a HB filter that has -100dB stop-band rejection has +/-0.0001 dB flatness in the passband. Not requiring a halfband characteristic but a looser passband spec (such as 0.01dB) will make the filter shorter in terms of time. All coefficients are now nonzero, but the total number of coefficients can be shrunk considerably, resulting in a reduction of group delay. Group delay is a serious problem in multitracking where a musician needs to hear herself play in the mix while tracking.
The result is that the filters are now commonly spec'ed as 0.01dB (or worse) flat until 20.000kHz (re 44.1kHz), stopband from 24.100k, because that was the accepted practice till now, no? Requiring a nonaliasing response would either mean shrinking back the passband to about 18kHz (a commercial no-no) or increasing the filter length (where the exercise was all about shortening it). The alias band thus remains status quo at 0.4535fs to 0.5465fs.
Why am I making such an issue of this flatness affair? Well, this periodic ripple corresponds to two small secondary spikes (echos) in the impulse response. One at the start, one at the end. For a filter that's flat to +/-0.01dB, these spikes are -66dB, and are a serious threat to stereo imaging and produce pretty obvious time smear (well, obvious to us perfectionists - aren't we a flea in the fur of the industry?)
3. Sharp filtering
A sharp filter has the effect of psychoacoustically enhancing the corner frequency, resulting in a kind of unnatural brightness or roughness. This effect is to my knowledge the most innocuous of effects associated with PCM. Also, making the filter steeper from what it is to begin with has no additional "brightening" effect.

How to attack the problems:
1. The aliasing problem can be solved at once, anywhere in the audio chain, using a single lowpass filter that enters stop-band before the alias band ie before 0.4535fs. A good place to do this is at reproduction or before final dithering. This means that halfband filters as in 1 may be used throughout without deleterious effects. I find that running a CD through an ultrasteep filter (pb to 18.5kHz, sb from 20kHz, eliminating all aliases that were created anywhere in production) results in an improvement in contrast, depth and precision of the stereo image.
2. The echos due to equiripple filters cannot be removed except by employing surgical precision on a case-by-case basis (like I'm doing to solve this problem on a TI SRC4192 - putting a DSP before and after the chip to compensate for the echos in the int and dec filters. In short, equiripple filters must be avoided at all cost. Simply specifying a higher sampling rate and a looser filter has the same effect of shortening group delay, but this implies that chips specified in this way are no longer suitable for the low sampling rates of 44.1k and 48k.
3. As we're already increasing sampling rate in view of the group delay problem, we can push it further to obtain a more natural roll-off after 20kHz.

Recipe for perfect PCM:
Specify a sample rate well above twice the audio band, e.g. 192kHz (lower is arguably acceptable too but since we're slowly standardising on 192 who cares)
Specify all interpolation/decimation filters as halfband, 0.4fs to 0.6fs transition.
Put exactly one non-halfband lowpass filter in the chain (e.g. at replay or before final dithering) that enters stopband at 0.4fs. Specifying its passband at 20kHz will allow for a very smooth roll-off and hence very short and practically ringing-free impulse response.

A more detailed paper concerning the aliasing and equiripple problems can be found at www.nanophon.com (read them all - excellent stuff).
A description of a method to calculate the "final lowpass filter" is AES preprint 5822, by Peter Craven.


Gardo:
Every recording medium has fundamental flaws. CD tried to address some of the fundamental flaws of vinyl and introduced flaws of its own. I'm sure this will be true no matter how much net improvement is realized in recording and playback. The question for me is whether SACD (generally speaking) sounds better than CD (generally speaking). On my modest system, it certainly does.

BP:
On my less than modest system it does too. I'm only saying that of all the possible solutions that could be formulated for the deficiencies of CD, DSD is the least intelligent one.


Gardo:
BP says that "SACD is here to stay." Do we think that's a step forward or a step back?

BP:
Sonically it's a step forward. In terms of practical usefulness it's more like putting the world's population on a spaceship to colonise a new planet without first checking if there's water and oxygen on it, and if the conditions on earth were really so bad we needed to leave it.


Gardo:
If I'm following your recommendations correctly, it seems that the second sentence should end "do not produce audible artefacts." Is that right?

BP:
Indeed: "do not". I've made the same typo in a different discussion not so long ago and it made things a bit confusing.


Gardo:
Also, I'm gathering from your remarks that you do not believe there is any reason to have a passband greater than 20khz, so I take it you do not agree with those who say there's extra information up there that we need to be able to reproduce. Is that correct?

BP:
I was a bit unclear there...
The passband only needs to be flat up to 20kHz, but it shouldn't drop off after that - there's quite a lot of sonically relevant information above 20k. In the case of the 192kHz system I proposed, the stopband (-80dB or better) would start at about 77kHz. The -3dB point would lie at 36kHz, -6dB at 43kHz, -10 at 49kHz. This means a lot gets through and it's a nearly perfect compromise between impulse response and bandwidth.
Also keep in mind that this filter should be applied only once over the entire signal chain. If it's done in the customer's player, he can even chose between this and a flatter (but "ringier") response. My hunch is that the slow rolloff option is the most sonically transparent.


I have also included a couple of other questions that some of you might find of interest:

TH:
With regards to SACD playback, I notice that some multiformat players convert DSD to PCM before output. Does this conversion degrade the SACD sound in any audible way?

BP:
The conversion to PCM in these players is done by chips by NPC (2 versions available). They can downsample to 8, 4, 2 and even 1fs. How badly the sound is affected depends very much on which chip/setting is used. Someone here has ordered samples so we can measure the filters. The spec says nothing about them. A lot depends on how well the filters are implemented (see last mail). Probably the 8fs version is relatively innocuous. Funnily enough this is done in order to allow the DSD signal to be reproduced by Burr-Brown multibit DAC chips.

SACD players exist in an enormous variety, up to and including devices (usually the cheapest ones) that downsample to 44.1kHz. Also their DVD signals are reduced to 48kHz first. This is why indeed you have to be careful about buying a cheaper player.

The audiophile brands each (or at least many of them) have their own converter philosophy, which warrants attention to detail but not necessarily knowledgeability :-)

I personally use a first generation SACD player (Marantz SA-1) which happened to be lying around here. I modified it to deliver the DSD at three BNCs at the back and have my own DAC to convert it to analogue. The same DAC will take 192/24 (but it does not yet have all the fancy filters present - next version) so by the time I can knock off a DVDA player somewhere I can use that too.


TH:
You seem to be an audiophile as well as an engineer. What is you opinion of vinyl as a delivery format for analog master tapes? The re-issuing labels like the late DCC, Classic Records, Sundance, etc. seems to put out some really good sounding releases that, unlike the old days, do not seem to be compromised (compressed or limited) in any way?


In terms of sound quality I believe both SACD and DVDA to be good formats for releasing remasters on. As said before, my problem with SACD lies in the practical side of affairs.
#23
20th October 2006
Old 20th October 2006
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Quote:
Originally Posted by T.RayBullard View Post
the tascam HDp2 is around that price and has timecode..
I don't consider a recorder that can only sync to an incoming timecode signal to be a true timecode recorder. It must be able to output as much as input- when on set, more often than not, the set's code is generated from the sound mixer.

The HD-P2 is about $1K (so a couple hundred bucks more), but I'm not a fan of it either... Sound is mediocre (for mediocre sound on analog, I'd just as soon stick to my microtrack and save a few hundred bucks), layout is questionable, software and menus are cumbersome... It'll do the job, but I don't really like it.

-Ben
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20th October 2006
Old 20th October 2006
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Thanks for the info Teddy! A fantastic post!

FWIW -
I read somewhere that a DG engineer was asked about the difference between 384 (yes Pyramix does downsample, but only for the duration of the edit) and DSD. He replied that moving a microphone an inch (in an orchestra recording) made a bigger difference than comparing the 2 formats.
#25
20th October 2006
Old 20th October 2006
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Korg?! Output format?

am i the only one blown away that Korg is putting out these units?

as far as i'm concerned they've got a fine reputation... with guitar tuners, keyboards, synth modules, and prosumer recording gear. considering this, it's understandable some are questioning it's analog front end.

that said, if this little guy (well, the larger of the two, anyway) can record DSD and do it well, sign me up! thumbsup

one question, though: which of these are options exporting the audio using the USB 2.0 port?
~DSD 5.6MHz
~DSD 2.8MHz
~DXD (384k PCM)
~downsampled PCM (24/96, etc.)
any clarity would be appreciated. would love the ability to record, transfer and edit in Pyramix while staying in DSD (or as close to it as per the above)...

cheers,
-c
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20th October 2006
Old 20th October 2006
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Quote:
When you convert the signal twice, however (such as when using a DSD recorder as the tracking medium), the second conversion is no longer transparent, due to the HF noise present in the source signal hitting a second analogue deltasigma modulator.
Now this seems to make sense ,since the recorded DSD material will have HF info, but some SACD projects by Telarc was mixed on an analogue board after being recorded in the sonoma daw and then mixed down into the sonoma workstation (since the sonoma can record while playing back ) I wonder if there was a lost of transparency problem ?.
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20th October 2006
Old 20th October 2006
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Quote:
Originally Posted by T.RayBullard View Post
this is exactly what I do with my final mixes..through a meitner ADC into the tascam.

teddy
Teddy,I am slightly confused about this.Are you using a meitner ADC or DAC during mixdown.

I had an opportunity to listen to some recordings made by Todd Garfinkle (MA recording ) on a MEitner/Philips SACD player.The Meitner converters are just wonderful.He said that many of the commercially available SACD players are actually not DSD to analog but DSD to PCM to analog and most folks dont even know about it.
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20th October 2006
Old 20th October 2006
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sorry..DAC. too many last night..

Quote:
Originally Posted by bit mangler View Post
Teddy,I am slightly confused about this.Are you using a meitner ADC or DAC during mixdown.

I had an opportunity to listen to some recordings made by Todd Garfinkle (MA recording ) on a MEitner/Philips SACD player.The Meitner converters are just wonderful.He said that many of the commercially available SACD players are actually not DSD to analog but DSD to PCM to analog and most folks dont even know about it.
#29
20th October 2006
Old 20th October 2006
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Quote:
Originally Posted by Don S View Post
Thanks for the info Teddy! A fantastic post!
thumbsup !!
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20th October 2006
Old 20th October 2006
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Does the Korg really have no SDIF inputs/outputs?

The Tascam DV-RA1000 has SDIF, so you can use outboard DSD converters like the Meitner.
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