Do you mean that this "Advanced Segment DAC Modulator" combined with the "Current Segment DAC" inside PCM1794 is actualy a 6bit R2R converter? I realy don't know.In some audio places, the only acceptable modern DAC chip is PCM1794.
Which is, coincidentally, 6-bit R2R......
The only i know is that the DAC chip WM8741 of Wolfson is more flexible and offers simillar performance. The PCM1794 offers a better DNR of 127dB (stereo - A weighted) but does not includes a I/V converter like the WM8741 that directly provides voltage outputs. The DNR of WM8741 is 125dB (stereo - A weighted). If you include and the necessary external I/V converter to achieve voltage output from PCM1794 i am sure that the total DNR will falls down to 125dB like in Wolfson DAC.
Let me turn the discussion a bit.
I have the suspicion that many people prefers PCM1794 just because is easily managed as all of its control lines are parallel and so it can be configured still with the use of simple analog switches. It is a big aid for people unrelated with microcontrollers which are the vast majority of diyAudio forum members. On the other hand WM8741, although offers as bonus the DSD PLUS functionality, it can be configured only through I2C or SPI interface when is working in software mode and where it offers a big variety of internal filters selection. But I2C or SPI interface presupposes the use of microcontroller, and hence the difficulty for most diyers.
In my current project (integrated amplifier offering 7 SPDIF inputs plus 1 analog) DAC section, i use the pair WM8805 - WM8741. To make its management as simple as possible for the end user, i use only the SPDIF receiver WM8805 in software mode, managed by a micro through SPI interface just to benefit from two of its functions: a) its internal 8:1 multiplexer which offers the possibility of selection among the 7 spdif inputs b) WM8805 recognizes the sampling rate of incoming spdif signal. This information is transfered into micro and accordingly it selects the proper oversampling rate (or set of filters) for the DAC WM8741 which is working in hardware mode. IMHO the automatic oversampling rate selection of DAC, is the most painful story around DACs, all other are mostly conjectures, Hi-End stories, exotic plugs - sockets - cables etc. I have seen it enough times into this forum in other instances and of course i gave any possible help as for the software part of micro.
Almost all configuration settings of WM8741 are accesible through hardware, yet the OSR selection. But it is not convenient each time the sampling rate of incoming spdif signal is changed, to you move in the DAC device to change the position of a switch. To not say that you could not have information of the sampling rate each time. This function to be executed automatically needs the use of microcontroller. That is.
Thanks
The fact that DWA (data-weighted averaging) is shown tends to indicate its a 'thermometer DAC' rather than R2R. Having such an architecture lends itself to turning mis-matches in individual resistors in the 'thermoneter' into noise.
it's a st3ll4r dac
Do you mean that this "Advanced Segment DAC Modulator" combined with the "Current Segment DAC" inside PCM1794 is actualy a 6bit R2R converter?
That's what I was told, however it seems to be an error.
The only i know is that the DAC chip WM8741 of Wolfson is more flexible and offers simillar performance.
The PCM1794 offers a better DNR of 127dB (stereo - A weighted) but does not includes a I/V converter like the WM8741 that directly provides voltage outputs. The DNR of WM8741 is 125dB (stereo - A weighted). If you include and the necessary external I/V converter to achieve voltage output from PCM1794 i am sure that the total DNR will falls down to 125dB like in Wolfson DAC.
Let me turn the discussion a bit.
I have the suspicion that many people prefers PCM1794 just because is easily managed as all of its control lines are parallel and so it can be configured still with the use of simple analog switches. It is a big aid for people unrelated with microcontrollers which are the vast majority of diyAudio forum members. On the other hand WM8741, although offers as bonus the DSD PLUS functionality, it can be configured only through I2C or SPI interface when is working in software mode and where it offers a big variety of internal filters selection. But I2C or SPI interface presupposes the use of microcontroller, and hence the difficulty for most diyers.
In my current project (integrated amplifier offering 7 SPDIF inputs plus 1 analog) DAC section, i use the pair WM8805 - WM8741. To make its management as simple as possible for the end user, i use only the SPDIF receiver WM8805 in software mode, managed by a micro through SPI interface just to benefit from two of its functions: a) its internal 8:1 multiplexer which offers the possibility of selection among the 7 spdif inputs b) WM8805 recognizes the sampling rate of incoming spdif signal. This information is transfered into micro and accordingly it selects the proper oversampling rate (or set of filters) for the DAC WM8741 which is working in hardware mode. IMHO the automatic oversampling rate selection of DAC, is the most painful story around DACs, all other are mostly conjectures, Hi-End stories, exotic plugs - sockets - cables etc. I have seen it enough times into this forum in other instances and of course i gave any possible help as for the software part of micro.
Almost all configuration settings of WM8741 are accesible through hardware, yet the OSR selection. But it is not convenient each time the sampling rate of incoming spdif signal is changed, to you move in the DAC device to change the position of a switch. To not say that you could not have information of the sampling rate each time. This function to be executed automatically needs the use of microcontroller. That is.
Very interesting mister!
WM8741 I have not heard yet.
I think it is popular for the minimum phase filter?
Well this thread seems to have relaxed!
It was very educational about the word "resolution" and I think I will write an article about this word in audio later.
By itself it's meaningless, we must attach "volume resolution" or "sample resolution" to it and so forth.
I think "resolving power" is a bit of a weird term as well.
DSD128 and DSD256 claim higher resolution, this doesn't seem to exist in PCM, i.e. something like "super-rate PCM" sampling at 100,000 times per second for a 44,100 Hertz bandwidth!
Anyone care to dispell DSD128?
I clearly remember there was a "SACD is finished" phase as if it were a failed project, much like Minidisc, but now it is coming back with full force!
I suspect it's thanks to reasons like very cheap external hard-drives, the human desire to keep improving and neophytes.
It was very educational about the word "resolution" and I think I will write an article about this word in audio later.
By itself it's meaningless, we must attach "volume resolution" or "sample resolution" to it and so forth.
I think "resolving power" is a bit of a weird term as well.
DSD128 and DSD256 claim higher resolution, this doesn't seem to exist in PCM, i.e. something like "super-rate PCM" sampling at 100,000 times per second for a 44,100 Hertz bandwidth!
Anyone care to dispell DSD128?
I clearly remember there was a "SACD is finished" phase as if it were a failed project, much like Minidisc, but now it is coming back with full force!
I suspect it's thanks to reasons like very cheap external hard-drives, the human desire to keep improving and neophytes.
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Maybe you could get it published on 6Moons....I think I will write an article about this...
Or maybe you can back up your remarks with something of substance rather than make vague comments referring to "the believers" which are recycling lunacy, instead of using $10 K-Mart CD players which is flat from 20 to 20 and have less than 0.1% THD.
Godfrey's remark was funny, at least I found it so. What kind of substance were you hoping for?
'Perfection' is not possible in electronics.Kastor L said:3 - I don't believe in reconstruction filters, due to......
- A lack of perfection
- A lack of evidence
- Time-domain error
'Evidence' would require that you read and understand a book about sampling and reconstruction.
'Time-domain' and frequency domain are inextricably linked, but fortunately our ears mostly work in the frequency domain so are not too bothered by alleged 'distortion' caused by simple filters.
The simple question is this: do you want to reproduce (to a high degree of accuracy) the signal which entered the ADC chip in the recording studio? If so, you need a reconstruction filter. That's it. This remains true whatever technology your DAC chip uses.
Read a book. Stop asking meaningless questions - like asking for 'zero noise'!
DF96 how about something constructive then.
The world is full of fact and myth, that's just the way it is.
The usefulness of a 20 Megapixel camera is myth, unless you want photos the size of a football field.
We need reconstruction filters, okay! I'll say Nos is myth from here on, unless someone like XXHighEnd wants to join us.
Are reconstruction filters near-perfect?
The link I sent to the Sabre ESS technical staff and Rob Watts, they have read many books and neither of them seem to think that normal filters are perfect!
So are we talking about 20MP here or 7MP? No book I read will answer that question, the books will only tell me how the filters work on a technical level.
The world is full of fact and myth, that's just the way it is.
The usefulness of a 20 Megapixel camera is myth, unless you want photos the size of a football field.
We need reconstruction filters, okay! I'll say Nos is myth from here on, unless someone like XXHighEnd wants to join us.
Are reconstruction filters near-perfect?
The link I sent to the Sabre ESS technical staff and Rob Watts, they have read many books and neither of them seem to think that normal filters are perfect!
So are we talking about 20MP here or 7MP? No book I read will answer that question, the books will only tell me how the filters work on a technical level.
DF96 how about something constructive then.
He gave you something constructive. So have I. So has Don. The advice remains: educate yourself on the basics so that you actually understand Fourier theory, sampling theory, Shannon-Nyquist, resolution, noise, accuracy, and precision. That way, your posts will not be meaningless word salads of vague and meaningless terms with internal contradictions, and you won't rely on poor analogies, ad material, and websites written by people with even less understanding.
Excuse me, but your mien is some aggressive... for a moderator. I have noticed this and in other threads. Please be more resigned with members of this forum.He gave you something constructive. So have I. So has Don. The advice remains: educate yourself on the basics so that you actually understand Fourier theory, sampling theory, Shannon-Nyquist, resolution, noise, accuracy, and precision. That way, your posts will not be meaningless word salads of vague and meaningless terms with internal contradictions, and you won't rely on poor analogies, ad material, and websites written by people with even less understanding.
Thank you
Excuse me, but your mien is some aggressive...
Ahhh - but is what he says correct? I think he is spot-on.
Sometimes you only have two choices: say it how it is, or say nothing at all.
Not always an easy choice, because there are always people who attack you about how you say it and are not interested in what you say.
I believe in your country it was once called 'shooting the messenger'. 🙂
Jan
If we are going to conclude this thread shortly I recommend you experts give your final thoughts on bit depth and dynamic range before I tell my friend to edit the Wikipedia.
Digital dynamic range and perceived.
Digital dynamic range and perceived.
I think most of those who know what they are talking about have already left the building.
It's all yours.
Jan
It's all yours.
Jan
Apart from the slight exaggeration, yes. For normal everyday photography 5Mpixel is quite adequate, especially with a cheap lens and marginal competency in the photographer. Camera resolution figures are like damping factor figures: once the number is sufficiently high (which is lower than most people think) then bigger numbers are mere specmanship - a useful way to take money off people who have learnt a new word but not yet thought about what it means.Kastor L said:The usefulness of a 20 Megapixel camera is myth, unless you want photos the size of a football field.
If NOS works, then it works by substituting wideband bursts of near-ultrasonic images (i.e. something not in the original music signal) for wideband bursts of slightly lower frequencies (percussive sounds which were in the signal, but partly removed by the anti-aliasing filter). Fortunately, our ears can't distinguish pitch at such high frequencies so a burst of the wrong frequency sounds much like a burst of the right frequency. This extra signal energy partly compensates for the sinc frequency response (drooping HF) which is normally a feature of uncompensated NOS. NOS is definitely not more accurate, but it may be more acoustically pleasing to some people.We need reconstruction filters, okay! I'll say Nos is myth from here on, unless someone like XXHighEnd wants to join us.
I am not aware of any competent engineer who thinks that any filter is "perfect". Some well-designed filters may approach 'ideal' behaviour, but that is different.The link I sent to the Sabre ESS technical staff and Rob Watts, they have read many books and neither of them seem to think that normal filters are perfect!
I think that I read almost the full tread...
But I do not find the explanation of differences between this two screenshots from post 110 even if was posts by "experts" in the field in my opinion.
The explanation it is very simple:
None exist in the real world but the first is much closer to reality.
But I do not find the explanation of differences between this two screenshots from post 110 even if was posts by "experts" in the field in my opinion.
The explanation it is very simple:
the first wave form show a square wave form with a limited bandwidth and it is not necessarily connected to an over-sampled DAC but to a reconstruction filter. To obtain a perfect square wave form you need a infinite bandwidth. This are the theory, it is not my statement. The "ringing" are not distortions, only show lack of superior harmonics necessary reconstruction of a rectangular waveform.
the second one show a voltage step, not reconstructed waveform, and the bandwidth it is much larger because it is a commutation and it is limited only by parasitic capacity and by slew-rate of the output of commutators from inside of cip. From my point of view it is more close to digital waveform not to analog waveform.
I would like to know what "acoustic" signal source can generate such a waveform (it is need a source with an infinite power because it is need a infinite bandwidth).
I would like to know what "acoustic" signal source can generate such a waveform (it is need a source with an infinite power because it is need a infinite bandwidth).
None exist in the real world but the first is much closer to reality.
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Surely is difficult, but you could embellish your phrasing, this is not so difficult.Truth is often difficult
We could accept any truth, provided it is well documented.and often difficult to accept.
Thank you
Surely is difficult, but you could embellish your phrasing, this is not so difficult.
I did, the first four or five times. 😀
Dear JanI believe in your country it was once called 'shooting the messenger'. 🙂
Jan
Indeed in history are mentioned many tyrants who executed the messenger when brought them bad news.
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