Thanks for putting up the link to those measurements because it illustrates the graph I was referring to earlier. Here's the relevant section of the THD+N vs level plot. The top of the window shown is 0.01% or -80dB for this figure, vertical scale 5dB and across the screen heading to the right is the level increasing:
Attachments
Hi Julf,
the DAC inside the Tera-Player runs on the native sample-rate of the wav file.
Supported sample-rates are 44.1/48/88.2/96/176.4 and 192 kHz.
The DAC could run up to 384 kHz, however then I would need a twice as fast ARM chip, which would be much bigger and would need more power. Therefore I have limited the max sample-rate to 192kHz, I wanted the Tera-Player to be small and have a long playtime, and there is not much 384kHz material available anyway.
Two separate oscillators with a max. RMS phase jitter of 1ps (acc. to data-sheet, I cannot measure such a low jitter) are employed for the timing.
One for 44.1 /88.2 /176.4 kHz.
and the other for 48/96/192 kHz.
The on-the-fly switching of the oscillators is quite tricky, and I am very happy that I was able to make it work, LOL
Charles
Erin, I would be loath to recommend a particular company, or item, because in my experience it's more complicated than that. I come from the angle of whole system engineering, looking at an audio setup as being a single component, or circuit, starting from where the mains power is hooked into at the wall, right up to where the speaker driver is being driven. The fact that this typically is all separated into individual boxes just makes the whole business of getting a system to work right very messy, and introduces a high percentage of the problems, to boot.
So a particular DAC may be brilliant, or barely tolerable, for a whole range of issues, many of which really need to addressed by fiddling with the other boxes - the best way - or trying variations of the other boxes.
That said, I, like many others, would go for reasonably priced units that have got good reports by a wide variety of users as a starting point.
I have had good experiences with Burr Brown R2R, in older Yamaha gear ...
Frank
Hi, Charles,
Indeed, and while I can't see the Philips head office building from my window, I just need to go around the corner to see it.
No, I don't - and not many people do here in Amsterdam. Most of the Dutch R&D is in Eindhoven (where Philips originally started out from), but almost all real operational and manufacturing activities are outside the Netherlands these days.
[and to abraxalito - sorry for lingering on in your blogthread, but I wanted to reply to the direct question from Charles]
Indeed, and while I can't see the Philips head office building from my window, I just need to go around the corner to see it.
No, I don't - and not many people do here in Amsterdam. Most of the Dutch R&D is in Eindhoven (where Philips originally started out from), but almost all real operational and manufacturing activities are outside the Netherlands these days.
[and to abraxalito - sorry for lingering on in your blogthread, but I wanted to reply to the direct question from Charles]
OT noise removed. Stick to technical discussion.
Hello Richard,
The worlds most used music production software is Pro Tools, its hardware would definitely be using SD A/D, as most everybody else does.
Could you please give an example of a Company which manufactures a non SD A/D converter which is used in professional audio products which is used for audio capturing for recording purposes in studios, which is current production.
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the only one i'm aware of is the Grimm AD1 but its primarily a DSD encoder and tbh i'm only assuming its not SD its a discrete convertor + FPGA
but its hardly going to be used in the majority
also remember its only a small minority that seems to think its even a problem and its certainly not a problem compared to ridiculously low resolution below that attainable by early 90's mobile phone ring tones and video game machines. I do not experience a dirty window, quite the opposite... my point was anything claiming to produce clean distinct non SD sound is putting forward a red herring, because the music used to assess this 'superiority' must already be 'effected' by this curse; yet it sounds marvelous... its a miracle! i'm cured!
frankly its a bit absurd to even blame the dac at this level. use a volume control like that on any 16bit dac of any type or quality and the volume control will be the limiting factor
but its hardly going to be used in the majority
also remember its only a small minority that seems to think its even a problem and its certainly not a problem compared to ridiculously low resolution below that attainable by early 90's mobile phone ring tones and video game machines. I do not experience a dirty window, quite the opposite... my point was anything claiming to produce clean distinct non SD sound is putting forward a red herring, because the music used to assess this 'superiority' must already be 'effected' by this curse; yet it sounds marvelous... its a miracle! i'm cured!
frankly its a bit absurd to even blame the dac at this level. use a volume control like that on any 16bit dac of any type or quality and the volume control will be the limiting factor
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Ah ok I wasnt sure. well ha I was just trying to provide a single example, i'm stumped now...
I suppose Bruno was a bit involved in the paper? i'd be interested to read it.
A "bit" (pun intended?) involved, yes.
http://www.grimmaudio.com/whitepapers/discrete ad converter.pdf
Hi Pheonix,
out of curiosity I created such a thing and offer it since 2005:
The Altmann Creation ADC
Sounds very good, sold a couple but not (yet) to major recording studios.
Sorry for answering on behalf of Richard,
Charles
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Hi Pheonix,
out of curiosity I created such a thing and offer it since 2005:
The Altmann Creation ADC
Sounds very good, sold a couple but not (yet) to major recording studios.
Sorry for answering on behalf of Richard,
Charles
That sounds like an SAR converter chip from TI
No recording studio is going to buy a 16 bit A to D. All mastering is done in 24 bit these days.
No recording studio with a real engineer is going to buy an ADC which lacks anti-aliasing filters, and tries to make a virtue of this lack - thus demonstrating an apparent lack of understanding by the designer (or perhaps a hope that his customers lack understanding). Unless, of course, the microphones used by the studio have such poor HF response that they act as the anti-aliasing filter themselves. Any recording made with this ADC will have aliases appearing from any input components at more than half the sampling rate. I suppose it is possible that some people might confuse these with 'detail'?
The ripples seen on the trace are caused by trying to put a sharp transition through a brick-wall filter. They are unavoidable because the filter is unavoidable, however much wishful thinking goes on in leiu of engineering.
The ripples seen on the trace are caused by trying to put a sharp transition through a brick-wall filter. They are unavoidable because the filter is unavoidable, however much wishful thinking goes on in leiu of engineering.
Its one SAR-chip per channel from Analog Devices.
Made this ADC, because I wanted to actually hear alias distortion, I only knew it from books
The Creation ADC samples up to 192kHz, where you would have to feed it with something above 96kHz in order to expect some alias distortion.
I never managed, except in one case ...
Now my question to the experts: What did I have to connect to my ADC's input in order to get really nasty alias-distortion ?
Charles
PS: eager to hear your suggestions, solution will come tomorrow
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