mwilson
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So the truth is that you somehow are the absolute arbiter in terms of what's worth reading and what isn't?
dCS Ring DAC - A Technical Explanation
- Thread starter dCS James
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- dac dcs dcs bartok ring dac
So the truth is that you somehow are the absolute arbiter in terms of what's worth reading and what isn't?
manueljenkin
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+1. I am interested to hear more about DCS approach.
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sajunky
Headphoneus Supremus
Of course, no question about. After noticing repeatedly missing a truth, it is not worth reading.
mwilson
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LOL ok then.
You seem displeased. I for one am happy to know who to see how things.
manueljenkin
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If you guys want something more comprehensive about different types of DAC, maybe look into this blog as well : http://s-audio.systems/blog/da-conversion-methods
Now let DCS explain their approach. They are making an explanation in simple terms, not obliged to make a lecture on every single type of dac out there on the planet. Read the title of the Blog, it's not named "A list of DAC architectures". It is named "dCS Ring DAC - A Technical Explanation".
What if to them sign magnitude just didn't sound good hence they didn't feel worthy of making a mention, like to some of us here to whom, negative feedback doesn't sound good despite measuring well.
Also stop assuming everyone else wants the same thing you want. Audio perception is still an open area of exploration and the boundaries of perception is still open. No conclusion can be derived as of yet since the type and structure of noise is as important as the level of noise. The correlation to presently measurable/measured parameters is also open ended.
Now let DCS explain their approach. They are making an explanation in simple terms, not obliged to make a lecture on every single type of dac out there on the planet. Read the title of the Blog, it's not named "A list of DAC architectures". It is named "dCS Ring DAC - A Technical Explanation".
What if to them sign magnitude just didn't sound good hence they didn't feel worthy of making a mention, like to some of us here to whom, negative feedback doesn't sound good despite measuring well.
Also stop assuming everyone else wants the same thing you want. Audio perception is still an open area of exploration and the boundaries of perception is still open. No conclusion can be derived as of yet since the type and structure of noise is as important as the level of noise. The correlation to presently measurable/measured parameters is also open ended.
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mwilson
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Displeased with what? Someone telling me what is and isn't worth reading? Darn right I am displeased, that will be my decision thank you very much.
My masochistic wallet that seems to have forgotten the safe word agrees with you.
I have notoriously low standards for a dac. Mainly turn on when told to, stay on, don't bother my ears and we'll be friends. This being said, I still enjoy James' explainers.
John Massaria
Member of the Trade: JM Audio Editions/Headphone Modifications
Fellow lovers of music unite in our differences - you are of course entitled to your opinion - state it without worry without resorting to drama - be calm state your claim carry on- remember why we are here - we are members united under the love of music 
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I for one came here to understand the ring DAC. I’m unlikely to run out and buy one. There seems no reason to argue and call names, let the man talk.
New guy here. I am interested. Mechanical Engineer but I made A to D and D to A converters in an electronics lab for Mechanical Engineers in about 1978. 8 Bit!! This is the first I've seen Pulse Density Modulation. Serious question - Does Nyquist apply to Pulse Density Modulation? I am not being sarcastic and I am not trying to stir up fights. Thanks.Part 2 – Basics of Pulse Density Modulation (DSD)
As opposed to PCM audio where the ADC sampling process takes the absolute value of the analogue voltage coming in to it at any given point, Pulse Density Modulation (PDM) instead works based on the time between two samples dictating whether the wave is increasing or decreasing in amplitude. If the samples are closer together, the wave is increasing in amplitude. If they are further apart, the amplitude of the waveform is decreasing. The absolute value of the waveform is not known per se when looking at an individual sample (as it would be with PCM), but put together the samples produce a good representation of the original waveform.
The caveat with this method is that the ‘dynamic resolution’ (the amount of information about the amplitude which is stored in any one sample of the audio) is incredibly low, being 1 bit, so the samples need to be taken at a much higher rate than with PCM audio. Where PCM typically samples at 44,100 samples a second, DSD works at a minimum of 64 times this rate, around 2,800,000 samples per second.
This process of encoding digital audio creates a lot more noise. This is due to both the low bit depth (which at 1-bit creates more quantisation noise) and the higher sample rate (essentially turning things on and off at a much higher rate creates noise). In order to make the format usable, the data is noise-shaped to clear the quantisation noise out of the audio band into the ultrasonic region (above 20kHz), where it cannot be heard.
The result is near 24-bit performance in the audio band (0 – 20kHz) and a signal bandwidth that extends beyond 100kHz. The price for the 1-bit approach is a very large amount of noise in the ultrasonic region (20kHz – 1.4MHz), but this is not normally heard as a noticeable background noise. This method of digitally encoding music is what is used in the format Digital Stream Direct (DSD). This format of 1-bit conversion is the basis of Bitstream Sigma-Delta Digital to Analogue Converters (which will be covered in a later post).
There are further developments into DSD audio, whereby higher and higher rates are used. The original rate, referred to as DSD/64 or Single Speed DSD, runs at 64x the rate of CD audio. DSD/128 or Double Speed DSD runs at 128x CD audio rates, and so on for DSD/256 and DSD/512.
DSD files, even at the standard DSD/64 rate are large. The data rate is 5644.8 kbps for 2-channel stereo.
This post is on the shorter side, but with the basic formats covered we can get on to the fun stuff. The next post will be on the basics of digital to analogue conversion, starting with Ladder DACs.
Part 3: Introduction to D/A Conversion
sajunky
Headphoneus Supremus
Then you know from a school that Pulse Density Modulation is not the same as DSD encoding. It is confusing, like anything in this tutorial. A reason some people in this thread talk about MQA folding across a Nyquist barrier.
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The original author is on a journey. I’d like to hear him explain it without so many angry-sounding arguments. I feel competent enough to follow without gatekeepers telling me every objection to his basic points. I promise I won’t run out and waste money on an expensive DAC without your consensus! I’ll present my credential: I’ve purchased two DACs that didn’t ride along free in a larger device. A $40 gaming sound card and a $100-ish Schiit DAC. So if the end game of this thread is to trick me into buying a DAC that costs $mega I’m fairly safe.
manueljenkin
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So when are we getting the next article of the series?
One point to make here is that dynamic range does not equal fidelity. The argument has been raised on the thread that a 75dB dynamic range means no other equipment should need to exceed this figure. This figure does not account for the amount of granularity the human ear/bran can hear within that 75dB range. It describes a ratio of sound pressure levels that can be detected by a microphone, but not how small of a change between two pressures within that range can be detected & recorded.
At the recording end, the ADC used by the studio should be attempting to extract as much information on the exact amplitude of the analogue wave output from a microphone as possible – the fine detail of the sound source. The actual levels of a mic’s output will obviously be adjusted during mixing, so the goal from the conversion standpoint is to take samples as accurately as possible from an amplitude perspective.
The playback side of the signal chain should be at least as capable as the converters used at the recording end, as if nothing else it allows for one less area of potential error between the studio and the listener.
Considering the potential implications of, for example, digital volume controls and DSP within a DAC – particularly given the sensitivity of IEMS – any attempts to achieve as low a noise floor as possible should definitely be taken.
As the most recent post states, it is meant to be an introduction to the topic of D/A conversion and doesn’t attempt to cover every possible DAC architecture – instead, the aim is to explore the fundamental limitations of common D/A approaches, and next will go into how the Ring DAC improves on them.
Taking the case of a sign magnitude DAC, this approach will mitigate the zero crossing point error effectively (it does that quite well), but it does not eliminate similar issues with other multibit changes. With a 16-bit R-2R (sign magnitude or not), the following are a few examples of crossings which are still problematic:
On resistor value tolerances, with an R-2R approach the crucial point to consider is that any error manifests as an error which is correlated to the audio, which perceptually is just about the worst case scenario – distortion.
Failure rates of components is a separate conversation, unrelated to dynamic resolution or converter types.
At the recording end, the ADC used by the studio should be attempting to extract as much information on the exact amplitude of the analogue wave output from a microphone as possible – the fine detail of the sound source. The actual levels of a mic’s output will obviously be adjusted during mixing, so the goal from the conversion standpoint is to take samples as accurately as possible from an amplitude perspective.
The playback side of the signal chain should be at least as capable as the converters used at the recording end, as if nothing else it allows for one less area of potential error between the studio and the listener.
Considering the potential implications of, for example, digital volume controls and DSP within a DAC – particularly given the sensitivity of IEMS – any attempts to achieve as low a noise floor as possible should definitely be taken.
As the most recent post states, it is meant to be an introduction to the topic of D/A conversion and doesn’t attempt to cover every possible DAC architecture – instead, the aim is to explore the fundamental limitations of common D/A approaches, and next will go into how the Ring DAC improves on them.
Taking the case of a sign magnitude DAC, this approach will mitigate the zero crossing point error effectively (it does that quite well), but it does not eliminate similar issues with other multibit changes. With a 16-bit R-2R (sign magnitude or not), the following are a few examples of crossings which are still problematic:
- 16383 – 16384 / 0011111111111111 – 0100000000000000
- 8191 – 8192 / 0001111111111111 – 0010000000000000
- 4095 – 4096 / 0000111111111111 – 0001000000000000
On resistor value tolerances, with an R-2R approach the crucial point to consider is that any error manifests as an error which is correlated to the audio, which perceptually is just about the worst case scenario – distortion.
Failure rates of components is a separate conversation, unrelated to dynamic resolution or converter types.
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