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I just received my Korg DS-DAC-10R recorder, so haven't had a chance to actually USE it yet... but having just installed the software (Audiogate4), it appears that one can burn SACDs! Although the software is free, it is limited in function and you can't burn discs.
The main reason I bought the Korg was because it also does the EQ digitally which is an interesting proposition if one uses the Phono EQ function since in principle you should get improved HF resolution due to the ADC being done BEFORE the EQ hence the pre-emphasis will benefit the amplitude resolution at HF. There is a theoretical penalty at the low end of course. The Line In can be switched to a Phono input and you can choose the EQ curve from RIAA,NAB,Columbia, Decca and AES as well as the RIAA-IEC curve (but who wants to use that!). The input sensitivity is only designed for MM and set for 5mV with an input overload margin of 26dB which isn't bad!
I just did spectral analysis on the idle state and was blown away at how quiet the spectrum was. I was wondering if I was going to have to spend extra on something like the iFi iUSB3.0.
I'll post a review when I get a chance to use it, although I'm in the process of changing arms on my deck while I refurbish the arm base assembly...
So far, it looks like it is going to be a very useful product! Although it won't accept balanced inputs or a Word Clock input.
Regards Anthony
"Beauty is Truth, Truth Beauty.." Keats
Follow Ups:
Hi,
I've been tempted to get one myself. I have a Korg MR-1000 that I bought years ago. It records in DSD formats to an internal hard drive.
One small clarification: Audiogate software allows you to burn DSD discs, not SACD discs. DSD discs require a compatible player; few SACD players can play these discs. Sony's recent UHP-H1 can though.
J
Thanks for that! I hadn't realised there was a difference until this morning when my Oppo 103 didn't know what to do with the disc - I see that a "real" SACD can't be authored on a PC.
It seems to be well thought out. I was worried about the possible sensitivity to noise on the USB power supply, but the spectrum was incredibly clean. This could well be PC dependent - I'll have to test it out on different machines before deciding if I need an additional power supply device.
The lack of a word clock sort of limits it for me so I will continue to use my Tascam DA-3000 or Benchmark ADC1 USB with a Grimm CC1 master clock for my reference recordings, but for the target application, I think it's a well-priced product. I got mine direct from Japan off eBay at a slight discount over the US prices.
Regards Anthony
"Beauty is Truth, Truth Beauty.." Keats
Your Oppo BDP-103 will play DSD files from a USB drive. I bought them for my children so they could play all my DSD music files. Therefore, if you can transfer the DSD file from your disc to a USB flash drive, you should be able to play them in your BDP-103. Of course, this assumes they are DSD(64) and not higher resolution DSD.
Best regards,
John Elison
Ah, good to know! Thanks!
I was somewhat disappointed that it didn't recognise a DSD Disc though I guess it makes sense that the optical disc will be strictly for compliant discs.
Regards Anthony
"Beauty is Truth, Truth Beauty.." Keats
Hi Anthony,
I don't understand why it would be beneficial to accomplish RIAA EQ after recording versus before. I don't understand why it would provide improved HF resolution. Can you explain this in more detail?
In my way of thinking, there should be no difference other than the ability to switch to different EQ curves after the fact.
Thanks,
John Elison
John,
here is the paper from Rob Robinson (Channel D) on this topic.
Regards
13DoW
I've downloaded it so I can read it thoroughly.
Thanks again,
John Elison
As flood2 stated, the signal direct from disc (no RIAA) has a better SNR for the treble before digitization than the post-RIAA version. The other side of the coin is that because the treble is boosted it might cause limiting in the ADC. But, according to the paper, it is not really a problem based on a large survey of real recordings.
Regards
13DoW
Hi John
Two main reasons:
1) Noise introduced by the input buffer and the quantisation noise at the ADC stage are the issues and encoding the pre-emphasised input signal (RIAA or other EQ curve) has the benefit of lifting the signal level above the noise level by at least the amount of the emphasis. Depending on the oversampling rate and the noise-shaping characteristics of typical D-S ADCs, the noise level rises rapidly above at least 20kHz (64fs i.e 2.8MHz DSD) thus potentially swamping the HF components of interest in the absence of pre-emphasis. Given that the spectral energy at HF is generally very low, this is important.
Once the data has been encoded, the signal can be manipulated by the required amount which reduces the ADC-introduced noise by the equivalent amount of the pre-emphasis.
2) We eliminate the analogue pre-amp doing the EQ which in itself adds noise
A properly dithered 16 bit, 44.1kHz signal can still resolve a signal down to close to -120dB with noise shaping so you will decode the original signal more faithfully compared to EQ applied PRIOR to the ADC.
Note that I was careful to write "in principle" in my original post. Given that the overall rms signal level of vinyl is typically well above the noise floor except at the top end of the audible range it is likely that the benefit is inaudible. However, from a theoretical point of view it is preferable to decode POST ADC. Not to mention that the digital EQ is not subject to component drift and tolerance introduced errors!
Of course the other issue (which PSAudio consider) is the phase characteristic of the EQ filter compared to an analogue filter - this is why PS Audio are at pains to implement a separate analogue filter in the PerfectWave ADC as they believe that is the best way to provide equivalency.
In the interests of a discussion topic, what would your reasoning be that there would be no difference?
Regards Anthony
"Beauty is Truth, Truth Beauty.." Keats
After reading you explanation, I'm sure there will be a difference. I suppose it depends on individual preference like everything else in vinyl reproduction. I've heard digital equalization of vinyl and it sounded okay to me, but it sounded slightly different that what I'm used to. Therefore, when I record vinyl, I attempt to accurately reproduce the sound of my analog system, which includes the sound of its analog RIAA filter.
Thanks again for your explanation!
John Elison
I take your point and it is certainly a concern of mine too since there is precious little information about the filter design (not surprisingly!). Surprisingly though, I couldn't find any specification on input capacitance which is not terribly useful given that it is designed ONLY for MM cartridges and the input "impedance" shifts from 47k when configured for Line IN to 50k for phono. I have emailed Korg, but am not hopeful of getting that level of detail so will probably have to work it out for myself.
Where were you able to get the digital EQ? Was it the Apple specific vinyl recording software (with a name that escapes me now!). I think the differences you hear will likely be down to the design of the digital filter simulating the time constants required and the effect on group delay. Linear phase is achieved with a finite impulse response FIR filter which is a feed forward design. This is why you get pre-ringing which is commonly attributed to the "non-analogue" characteristic of digital.
Regards Anthony
"Beauty is Truth, Truth Beauty.." Keats
One of the people at Muzikmike's Ribfest had a digital phono stage. I don't know what brand it was but he was emphasizing to us that it applied RIAA equalization in the digital domain. I thought it sounded a bit different than most analog phono stages. It's been several years ago and I don't actually remember too much about it. I just recall my reaction at the time was that it sounded good but somewhat different.Best regards,
John Elison
Edits: 01/15/17
Digital filters can potentially be more ideal in that they don't have the various non linearity distortion problems real world filter components all have.Typical analog phono filters work by attenuating high frequencies and gaining the low frequencies. Feeding the raw uncompensated signal to the ADC means there is more sampled bits per volt of high frequency and less bits per volt of low frequency. You may be able to hear the difference due to the type of error differences introduced by doing it one way as opposed to the other.
Edit: Damn, forgot to mention noise. All the thermal, Brownian, emi interference, etc. noises that are added from all the extra circuitry in the RIAA filters removed from the equation when you go digital /edit
I've wondered about possible quality enhancements which may be open to those doing compensation in the digital domain:
I've heard some people think that using minimum phase filters on the high frequencies sounds better than using the linear phase filters that analog phono filters stages use. Since minimum phase filters are finite impulse response, I believe the only practical way to accomplish and apply them is in software.
A hybrid system where low frequencies are filtered in the analog domain and the high frequencies are compensated digitally might potentially minimize the trade offs a person looking to run minimum phase filters on the high frequencies might experience. Though, a dsp expert may have something to say about the downside of using a minimum phase filter to compensate for what was presumably a typical linear phase filter used in the original encoding during the record manufacturing. I'm not a digital signal theory expert.
Edits: 01/15/17 01/15/17 01/15/17 01/15/17 01/15/17
Good explanation! I hadn't thought about the resolution and distortion advantage in the high-frequencies or the noise advantage.
Thanks again,
John Elison
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