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£1k-2k DACs with remote control?

I'm looking for a mid-to-high end DAC for my hifi which has a remote control for switching inputs.The rest of the system is SBT with TT3 and custom linear P/S --> DAC --> Naim Nac202 with DIY regulated Hicap clone with Mr Tibbs' earthing mod --> Nap200 --> B&W 705. Mains via Isol-8 Mini Sub.
At 202/200 level, nDAC is the answer for a coherent system. While there may be fierce competion in the 2k£+ category, it will always undermine this brand cohesion and signature. Chag -
 
Mark from Item leant me a Wired 4 sound DAC for a month and was vey impressed.
John Westlake has modded my MDAC and when I get it will be a good comparison
 
Mark from Item leant me a Wired 4 sound DAC for a month and was vey impressed.
John Westlake has modded my MDAC and when I get it will be a good comparison

I'd be very interested to hear your thoughts on the comparison...!

Loki
 
Well I bite the bullet today and bought the Leema Elements DAC. It has not put a foot wrong in the time I've had it and pulls more detail out than any other of the DACs I've listened too. It also seems to add nothing, if that makes sense, you don't know that it is there, you just listen to the music. A short listening session turns into hours, you don't play selected tracks, but get lost in albums.

I'm using the balanced outputs straight into Active Speakers.

The Wryrd4Sound DAC definitely is a great DAC but it emphasizes bass, appearing to have great drive, but its a bit like having a loudness button on all the time. I didn't like the MDAC, but all these things are personal.
 
I have been considering the Leema as it has an analogue input, how does that work out compared to the digital inputs.
 
I don't know haven't used it yet, all my current sources are digital, though I could wire the SBT or Oppo to it, to see (hear). I would expect that it would work like a 'normal' preamp, the Leema being effectively a two input pre-amp, one the output from the DACs and the other from the external analogue input, followed by the volume control and analogue output stage.

It is DAC that should be on everyone's shopping list, if they want a sub 2K DAC with a volume control, and the fact that the volume control is in the analogue domain, means that no digital data will be lost with low volumes (Important if there is no other pre or attentuation further down the chain)
 
Digital attenuation doesn't need to lose bits if it is properly dithered, there is a white paper on the subject by Daniel Weiss I will post a link.
Keith.
 
Keith. Dan’s design does loose bits he evens says so in the paper and the manual supplied with the DAC202. It starts chopping off LSB's (least significant bits) from -20 all the way down to -60. Which means only the upper 3rd section of the volume control is usable as/for bit perfect audio. It is in Dan’s concept that the loss of these bits isn’t audible or significant to warrant an analog volume control..... He's not said that properly dithered attenuation is lossless. By design digital volume is exactly lossy; it's how it works!!
 
Keith

I agree, but many of this 'mid' price DACs, do seem to have an issue at low volumes, the Leema obviously won't have this issue. Weiss products were outside my budget (so I delibrately didn't audition, as my flexible friend is already being a bit too flexible!)
 
The other dac you might have looked at is the Lynx HILO, two analogue inputs, analogue and digital attenuation, completely seperate headphone amp, it is incredibly versatile, about £2k.
Keith.
 
Keith, that explains the concept not the implementation. However, Daniel has said himself that anything below -20db is indeed lossy. Digital volume control/attenuation/quantization is by default lossy. But there are workarounds to a extent, after that there are laws of physics ;)
 
Another nice feature of the Leema DAC ( I haven't got one so I'm not hyping it) is that it is made in the UK so service and support should be reliable - and a couple of ex-BBC guys are behind the company, which may or may not be a good thing.
 
Keith, that explains the concept not the implementation. However, Daniel has said himself that anything below -20db is indeed lossy. Digital volume control/attenuation/quantization is by default lossy. But there are workarounds to a extent, after that there are laws of physics ;)

It is all there in the white paper Raj, perhaps. re read it?
Where has Daniel 'said' that below -20db is indeed lossy?
Keith.
 
I did my research......... I'll dig it out Keith, IIRC he said so on a thread on Computer Audiophile. The paper is only a concept. The idea is based on what humans can detect as lossy or lossless. It's why the bit perfect test the DAC202 performs, can only be done with the files supplied with the DAC. These test files contain signals beyond our hearing and when used are played at -0db. It’s the only time the DAC202 is actually truly bit perfect!
 
I shall look forward to reading it Raj, bits are indeed lost unless you use dithering, it is a widely used technique in properly designed and implemented equipment.
Keith.
 
bits are indeed lost unless you use dithering
Keith.

That's absolutely right Keith, but only to an extent. Beyond a certain threshold dithering becomes lossy. The point is that we are supposed to not detect the loss when the db's are so low...

The implementation could easily be improved if the course volume setting (optional fixed output voltages) in the DAC202 also stepped down in tandem with the digital attenuation when the LSB's start to become truncated. Others have done exactly this and don't lose more than -4db along the whole volume range.
 
Found this so far, its the artical Dan refers to in his white paper: http://www.digido.com/dither.html have a look at 'Part II',

C&P from the link above:

Dither


How to Dither Let's look at that long sample word. Whether it's 24 bits or 32 bits, we have to find some way to move the important information contained in the lower (least significant) bits into the upper 16 bits for recording to the CD standard. Truncation is very bad. What about rounding? In our digital dollar example, we ended up with an extra 1/2 cent. In grammar school, they taught us to round the numbers up or down according to a rule (we learned "even numbers...roundup, odd...round down"). But when we're dealing with more numerical precision and small numbers that are significant, it gets a little more complicated.

It turns out the best solution for maintaining the resolution of digital audio is to calculate random numbers and add a different random number to every sample. Then, cut it off at 16 bits. The random numbers must also be different for left and right samples, or else stereo separation will be compromised.

For example:

Starting with a 24-bit word (each bit is either a 1 or a 0 in binary notation):

Upper 16 bits Lower 8

Original 24-bit Word MXXX XXXX XXXX XXXW YYYY YYYY

Add random number ZZZZ ZZZZ

The result of the addition of the Z's with the Y's gets carried over into the new least significant bit of the 16-bit word (LSB, letter W above), and possibly higher bits if you have to carry. In essence, the random number sequence combines with the original lower bit information, modulating the LSB. Therefore, the LSB, from moment to moment, turns on and off at the rate of the original low level musical information. The random number is called dither; the process is called redithering, to distinguish from the original dithering process used to during the original recording.

Random numbers such as these translate to random noise (hiss) when converted to analog. The amplitude of this noise is around 1 LSB, which for 16 bit lies at about 96 dB below full scale. By using dither, ambience and decay in a musical recording can be heard down to about -115 dB, even with a 16-bit wordlength. Thus, although the quantization steps of a 16-bit word can only theoretically encode 96 dB of range, with dither, there is an audible dynamic range of up to 115 dB! The maximum signal-to-noise ratio of a dithered 16-bit recording is about 96 dB. But the dynamic range is far greater, as much as 115 dB, because we can hear music below the noise. Usually, manufacturer's spec sheets don't reflect these important specifications, often mixing up dynamic range and signal-to-noise ratio. Signal-to-noise ratio (of a linear PCM system) is the RMS level of the noise with no signal applied expressed in dB below maximum level (without getting into fancy details such as noise modulation). It should be, ideally, the level of the dither noise. Dynamic range is a subjective judgment more than a measurement--you can compare the dynamic range of two systems empirically with identical listening tests. Apply a 1 kHz tone, and see low you can make it before it is undetectable. You can actually measure the dynamic range of an A/D converter without an FFT analyzer. All you need is an accurate test tone generator and your ears, and a low-noise headphone amplifier with sufficient gain. Listen to the analog output and see when it disappears (use a real good 16 bit D/A for this test). Another important test is to attenuate music in your workstation (about 40 dB) and listen to the output of the system with headphones. Listen for ambience and reverberation; a good system will still reveal ambience, even at that low level. Also listen to the character of the noise--it's a very educating experience.

Some Tests for Linearity
You can verify whether your digital audio workstation truncates digital words or does other nasty things, without any measurement instruments except your ears. Obtain the disc Best of Chesky Classics and Jazz and Audiophile Test Disc, Vol. III, Chesky JD111.* Track 42 is a fade to noise without dither, demonstrating quantization distortion and loss of resolution. Track 43 is a fade to noise with white noise dither, and track 44 uses noise-shaped dither (to be explained). Use Track 43 as your test source; you should be able to hear smooth and distortion-free signal down to about -115 dB. Then listen to track 44 to see how much better it can sound. Try processing track 43 with digital equalization or level changes (both gain and attenuation, with and without dither, if it's available in your workstation) to see what they do to the sound. If your workstation is not up to par, you'll be shocked. Use a quiet, high-gain headphone amplifier to help reveal the low level problems.

*available at major record chains or through Chesky Records, Box 1268, Radio City Station, New York, NY 10101; 212-586-7799. The hard-to-find CBS CD-1, track 20, also contains a fade to noise test.

So Little Noise, So Much Effect
-96 dB seems like so little noise. But strangely, engineers have been able to hear the effect of the dither noise, even at normal listening levels. Dither noise helps us recover ambience, but conversely it also obscures the same ambience we've been trying to recover! Dither noise adds a slight veil to the sound. That's why I say, dither, you can't live with it, and you can't live without it.

Improved Dithering Techniques
Where there's a will, there's a way. Although the required amplitude of the dither is about -96 dB, it's possible to shape (equalize) the dither to minimize its audibility. Noise-shaping techniques re-equalize the spectrum of the dither while retaining its average power, moving the noise away from the areas where the ear is most sensitive (circa 3 KHz), and into the high frequency region (10-22 KHz).
 
Keith, it also says in the DAC202 manual, that if the volume control isn't at 0.0db then it's not bit perfect (lobbing of LSB's) and the transparency check will fail.
 


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