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It helps on producing low distortion low level signals. Miska, as regards non-harmfulness of DSD noise, I would like to quote an excerpt from a document which can be found here http://www.cirrus.com/en/pubs/whitePaper/1103_ABCs_of_DACs.pdf. From it follows that even DSD developers themselves advise to filter quantification noise, which, according to you, is even useful. Therefore I wanted to hear what a representative of dCS says on this subject. "Requirements for Amplifier and Loudspeaker Protection" By their nature, analog-to-digital converters that utilize single-bit delta-sigma techniques generate a significant amount of quantization noise. These converters operate at high oversampling ratios, 2.8224 MHz for SACD, which distributes this noise over a wide bandwidth. In addition to the extended bandwidth, the quanti-zation noise is shaped by the modulator in such a way as to minimize the noise that exists within what is generally considered to be the audio bandwidth. When operating in a PCM mode, the digital decimation filter in the A/D removes the noise that is beyond the audio bandwidth. However, for DSD this noise is transferred to the disc, and there are potential systems issues that can arise when this high-frequency noise is converted into the analog domain and applied to an audio amplifier and speaker. These issues can include slew rate limiting in the amplifier stages as well as inter-modulation distortion and damage to the high-frequency drivers in loudspeakers. Philips Electronics and Sony Corp. recognize and address this issue in a document titled "Super Audio CD System Description." This document, commonly referred to in the audio industry as the Scarlet Book, is the defining document for Super Audio CD standards and systems requirements. Part 2, Annex E of this document recommends that, for pro-tection of analog loudspeakers and amplifiers, a SACD player employ an analog low-pass filter with a cut-off frequency of 50 kHz and a minimum 30 dB per octave slope (though this bandwidth recommendation is relaxed for wide-band audio gear). Unfortunately, this recommended analog filter has several potential pitfalls that include the frequency and phase response variations that result from standard component tolerances, additional circuit complexity, potential performance degradation, and increased cost. The systems designer must also determine if this analog filter is to remain within the signal path when the system is operating in a high-sample rate PCM mode, such DVD-A at 192 kHz sample rate. If the decision is to remove the filter, an additional cost is incurred to provide the required switching circuitry and control logic, as well as a risk of degrading sonic quality with this additional circuitry in the signal path. The overwhelming majority of digital-to-analog converters that are capable of converting DSD require an external analog filter to meet the Scarlet Book recommendations. Fortunately there is another option. Since this noise originates in the DSD recording process and is present in the digital domain, it is feasible that the 50 kHz filter can be implemented digitally to meet the goal of protecting analog amplifiers and speakers.

I have noticed that many people still are convinced that SACD provides 100kHz band. At the same time viewing spectrums for PCM files which are converted from DSD, it can be seen that the upper part of the spectrum is filled with noise. Sometimes this noise can be observed as low as at below 30kHz. Once I even wrote to the HDtracks do they know that half of the product they have sold to me is simply quantisation noise? As a representative of dCS has joined this discussion, I’d like to ask him if he can in simple text tell us about this particularity of DSD – quantisation noise. Where from it comes? How dangerous it is? Can it affect the sound in the range audible for an ear?

Thank you Chris, with your involvement all became clear so fast! The first time I tried out these files on MC15 system quite loudly – the noise was awful. In future I’ll be more careful with new content

I tried out Bravura Records files with MC14 on Vista and MC15 on Windows 7. Results are identical. Can it be a bug in Media Center? I also tried to convert one of files to wav and look at the waveform and spectrum. Never have seen anything like yet…

I advise to be very careful with these Christmas gifts, as spectres of audio files contain some additional components. E.g., playing them with J. River Media Center, there can be heard very strong noise which repeats regularly. On the other hand, with Foobar they sound normal. In turn to Chris I would advise to take a look at file spectrums before recommending them for download.

I'm surprised that nobody in this discussion has mentioned such a producer as MSB Technology. When I did choose DAC I had alike questions. This is what MSB Technology representative told me: „The unique feature of our DAC is our complete reclocking. We put the audio into a buffer on our DSP and clock it out using high-precisions oscillators (TCXOs). So with the complete ground isolation that our USB input has or that Toslink provides, combined with the reclocking system in our DACs, you can make a computer as good, if not better, than any other transport.”. Of course, the question arose for me about possible buffer overflow or its emptying in a case if I’m listening to the disc without pauses between tracks. Here is the answer: „When we designed the buffer, we planned on CDs without pauses such as Pink Floyd. We also have other albums that are without pauses. If it does reach the end of the buffer, it would reset itself and there would be a small tick in the audio. None of our customers have experienced this problem and we’ve been shipping Platinums with the reclocking system for years.” In my view the best solution (better than asynchronous transmission in which DAC plays a role of a host) is right like that used by MSB Technology. Of course, much depends on quality of implementation of the principle, but I do not doubt that in the case of MSB Technology.

http://jn.physiology.org/cgi/content/full/83/6/3548

Hi Chris,<br /> <br /> What would be your advice to those who want to enjoy multi-channel records, e.g., from iTrax.com? To connect three Berkeley Audio Design Alpha DAC to the sound card? Or maybe simply to connect Lynx Aurora 8/FW to the computer?

More 24/192 content from 2L:<br /> http://hirezfiles.com/blog/?p=5

It would be very interesting to know your opinion about Bencmark Media Systems DAC1 PRE, as an alternative DAC in reference music server system. <br /> <br /> And what is your opinion about USB interface between DAC and music server. I have seen how Bencmark Media Systems DAC1 USB on connection to USB itself automatically adjusts necessary settings in Windows XP operating system for transparent bit-to-bit transfer. <br />

<a href="http://www.stereophile.com/news/010508ref/>Stereophile, January 5, 2008</a><br /> <br /> <cite>The <strong>Alpha DAC</strong>, which does not infringe on the HDCD patents that Microsoft now owns, will debut at CES this week. Used with a PC equipped with an expensive <strong>Lynx sound card</strong> commonly capable of outputting digital data sampled at 176.4kHz, the DAC will be used to demonstrate RR's HRx files. I asked what program will be used to play the files.</cite><br /> <br /> <cite>"We're only working with <strong>Windows XP</strong>, because Windows Vista is currently a disaster for audio," says Martin. "Vista tries to control and play files its way, which is not what we want. . . .Meanwhile, one of the programs we've been using is a free one called <strong>Media Monkey</strong>. It goes up to 24/176.4 without messing up and changing the files and bits." </cite>