thanks to Fedde and others who pointed out
that the BCK, WS, DATA signals that came out the Marantz
CD5000 controller are not I2S, and already oversampled, I
took the SPDIF output of the Marantz and feed it to a
California audio labs Gamma DAC. I then proceeded to hack up the CAL DAC so its CS8412 chip would output in I2S format.
Next I connect the I2S lines from the cs8412 to the non OS DAC based on one
of Peter Daniel's schematic. The sound is relaxed, but does not
seem to be any better than the what my tweaked CD5000
can put out. I put a scope on the waveform and it looks like
the picture below. Notice the dotted waveform instead of
being continous, it is dotted. It is as if every other digital sample
is not being converted to analog. May be this is why the sound
does not seem quite as good as it could be. Would a reclocking of the I2S lines from the CS8412 before feeding them to the TDA1543 fix the dotted waveform problem?
that the BCK, WS, DATA signals that came out the Marantz
CD5000 controller are not I2S, and already oversampled, I
took the SPDIF output of the Marantz and feed it to a
California audio labs Gamma DAC. I then proceeded to hack up the CAL DAC so its CS8412 chip would output in I2S format.
Next I connect the I2S lines from the cs8412 to the non OS DAC based on one
of Peter Daniel's schematic. The sound is relaxed, but does not
seem to be any better than the what my tweaked CD5000
can put out. I put a scope on the waveform and it looks like
the picture below. Notice the dotted waveform instead of
being continous, it is dotted. It is as if every other digital sample
is not being converted to analog. May be this is why the sound
does not seem quite as good as it could be. Would a reclocking of the I2S lines from the CS8412 before feeding them to the TDA1543 fix the dotted waveform problem?
Attachments
Hi,
I can't see it. Could you post a better picture. Try fiddling with your scope a bit.
Regards,
Thijs
I can't see it. Could you post a better picture. Try fiddling with your scope a bit.
Regards,
Thijs
bqc said:
Hi bqc,
I don't see it either. Better use a test CD with a 3150 Hz sine tone on it. This will show up any discontinuities and or "steps" with suitable scope settings. Steps should be visible with a NON-OS DAC with no analog anti-aliasing filtering applied.😎
Reclocking the I2S lines would not alter this situation.😎
That looks normal !
Hi BQC,
The output of a DAC without a filter would be like a staircase. They are discrete samples at whatever the sampling frequency is. It IS a staircase. It gets "smoothened" when you filter it . That's what happens in all dacs.
IIUC a non over sampling DAC depends on other parts of the listening chain to achieve this . Like the Hf roll off of tubes , our ears etc.
For a signal sampled at much higher frequency the steps are smaller on the staircase.
I wonder if anyone has compared the audible difference between a 44.1Khz non-filtered DAC with a 8x44.1Khz upsampled non-filtered DAC . Anyone?
Hi BQC,
The output of a DAC without a filter would be like a staircase. They are discrete samples at whatever the sampling frequency is. It IS a staircase. It gets "smoothened" when you filter it . That's what happens in all dacs.
IIUC a non over sampling DAC depends on other parts of the listening chain to achieve this . Like the Hf roll off of tubes , our ears etc.
For a signal sampled at much higher frequency the steps are smaller on the staircase.
I wonder if anyone has compared the audible difference between a 44.1Khz non-filtered DAC with a 8x44.1Khz upsampled non-filtered DAC . Anyone?
steps
Hi Ashok,
Yes I compared NON-OS DAC with 8x oversampled DAC both unfiltered. In fact I tried all possible combinations. Though analog low-pass filtering improves the sound, my conclusion was: the digital filter (CXD1244) was the culprit for bad sound.
Hi Ashok,
Yes I compared NON-OS DAC with 8x oversampled DAC both unfiltered. In fact I tried all possible combinations. Though analog low-pass filtering improves the sound, my conclusion was: the digital filter (CXD1244) was the culprit for bad sound.
Is that what you mean Ashok? Is upsampling the samen as oversampling?
I thought they weren't the same.
Regards,
Thijs
I thought they weren't the same.
Regards,
Thijs
Up or Over sampling ?
Hi Thijs,
With respect to the topic we were discussing I think we all understood what we meant. But to address the subject by itself I am quoting a well written article from Hi-Fi news and Record Review.
UPSAMPLING AND OVERSAMPLING
While there is certainly overlap (but never ‘uplap’!) in the use of the terms ‘oversampling’ and ‘upsampling’, some guidelines can be given to differentiate the processes.
Oversampling is typically used to describe a technique used when transferring between the analogue and digital domain, where a signal is sampled many times over and above that actually required by the sampling frequency.
Oversampling in the context of the D-A process involves multiplying the sampling frequency by a whole number, typically between 4 and 32, or even higher. For example, in ‘8x oversampling’, CD’s base rate of 4.4.1kHz is raised to 352.8kHz by introducing seven new ‘empty’ samples between the original data samples. These new samples, though, are often not just empty strings of noughts, but based on mathematical models to assist the DAC to work more linearly with the extracted data.
Oversampling, as well as easing the workload of the anti-aliasing filter, which can now operate more gently at a higher frequency, can also reduce distortion created when those analogue signals are first turned from continuous, analogue waveforms into stepped, digital, stair-like curves. This quantization noise is now spread over a larger band after oversampling, and can even be somewhat shifted out of the audible envelope by the technique of noise-shaping. Sony/Philips’ Direct Stream Digital, as used
in SACD, takes this idea to its limit, in order to dump high levels of digital noise up to higher frequencies than are not directly audible.
Upsampling is a solely digital domain process where the data stream is also stretched out by interpolation — guessing the points in between, again mathematically — and is typically used to refer to small, non-integer changes, such as from 44.1 kHz to 48kHz. When the change is larger than this, such as 44.1 kHz to 192kHz, ‘upsampling’ is a more popular term.
'There is apparently no extra information in the upsampled signal that was not present in the initial signal,’ says Mike Story of dCS. ‘With a 44.1 kS/s input, both the input data stream and the upsampled data stream will only contain a spectrum that must be between 0 and 22.05 kHz and is probably only between 0 and 20kHz.'
'This conventional analysis starts from the viewpoint that the behaviour of the ear can be described in mathematical terms using Fourier analysis. This assumption is probably pretty good — it means we are interested in frequency responses, for example, and these do provide good guides to the performance of equipment and to descriptions of what we hear. The analysis was right at the heart of the definition of the audio coding used on CDs.'
‘For those working with audio, it is also apparent that thearies based on these descriptions are not completely adequate, and that there can be significant differences in the performances of pieces of equipment with similar "conventional" specifications. It seems that two things are going on here: the ear may have more than one mechanism at work; and sine waves may not be the best function to use as the basis for analysis. On the mechanism front, it seems highly likely that the ear has a sound localisation mechanism ("where is it?") that is fast, and independent of the mechanism that says "it’s a violin", and that is related to transient response. There may also be a third mechanism at work. On the analysis front, it may be that some form of wavelet is the best basis for mathematical modelling. The problem here is that sine-wave theory is relatively simple, and has been fully worked out by generations of mathematicians, following on from Fourier. Wavelet maths is just plain hard work, and does not yet have anything like such a solid core of mathematical results to call upon. Our ears, however, are not waiting.’
( From Andrew Harrison - Hi-Fi News and Record Review
August 2000 )
On our DIY forum I think we could use the terms interchangeably because we are practically only talking about DAC's.
Cheers.
Ashok.
Hi Thijs,
With respect to the topic we were discussing I think we all understood what we meant. But to address the subject by itself I am quoting a well written article from Hi-Fi news and Record Review.
UPSAMPLING AND OVERSAMPLING
While there is certainly overlap (but never ‘uplap’!) in the use of the terms ‘oversampling’ and ‘upsampling’, some guidelines can be given to differentiate the processes.
Oversampling is typically used to describe a technique used when transferring between the analogue and digital domain, where a signal is sampled many times over and above that actually required by the sampling frequency.
Oversampling in the context of the D-A process involves multiplying the sampling frequency by a whole number, typically between 4 and 32, or even higher. For example, in ‘8x oversampling’, CD’s base rate of 4.4.1kHz is raised to 352.8kHz by introducing seven new ‘empty’ samples between the original data samples. These new samples, though, are often not just empty strings of noughts, but based on mathematical models to assist the DAC to work more linearly with the extracted data.
Oversampling, as well as easing the workload of the anti-aliasing filter, which can now operate more gently at a higher frequency, can also reduce distortion created when those analogue signals are first turned from continuous, analogue waveforms into stepped, digital, stair-like curves. This quantization noise is now spread over a larger band after oversampling, and can even be somewhat shifted out of the audible envelope by the technique of noise-shaping. Sony/Philips’ Direct Stream Digital, as used
in SACD, takes this idea to its limit, in order to dump high levels of digital noise up to higher frequencies than are not directly audible.
Upsampling is a solely digital domain process where the data stream is also stretched out by interpolation — guessing the points in between, again mathematically — and is typically used to refer to small, non-integer changes, such as from 44.1 kHz to 48kHz. When the change is larger than this, such as 44.1 kHz to 192kHz, ‘upsampling’ is a more popular term.
'There is apparently no extra information in the upsampled signal that was not present in the initial signal,’ says Mike Story of dCS. ‘With a 44.1 kS/s input, both the input data stream and the upsampled data stream will only contain a spectrum that must be between 0 and 22.05 kHz and is probably only between 0 and 20kHz.'
'This conventional analysis starts from the viewpoint that the behaviour of the ear can be described in mathematical terms using Fourier analysis. This assumption is probably pretty good — it means we are interested in frequency responses, for example, and these do provide good guides to the performance of equipment and to descriptions of what we hear. The analysis was right at the heart of the definition of the audio coding used on CDs.'
‘For those working with audio, it is also apparent that thearies based on these descriptions are not completely adequate, and that there can be significant differences in the performances of pieces of equipment with similar "conventional" specifications. It seems that two things are going on here: the ear may have more than one mechanism at work; and sine waves may not be the best function to use as the basis for analysis. On the mechanism front, it seems highly likely that the ear has a sound localisation mechanism ("where is it?") that is fast, and independent of the mechanism that says "it’s a violin", and that is related to transient response. There may also be a third mechanism at work. On the analysis front, it may be that some form of wavelet is the best basis for mathematical modelling. The problem here is that sine-wave theory is relatively simple, and has been fully worked out by generations of mathematicians, following on from Fourier. Wavelet maths is just plain hard work, and does not yet have anything like such a solid core of mathematical results to call upon. Our ears, however, are not waiting.’
( From Andrew Harrison - Hi-Fi News and Record Review
August 2000 )
On our DIY forum I think we could use the terms interchangeably because we are practically only talking about DAC's.
Cheers.
Ashok.
For Elso.
Hi Elso,
It's interesting to see that you did check the differences between the 8x version of the DAC. Now that you concluded that it could be the digital filter that might be the culprit which detracted from the sound - we could probably look for a DAC/digital filter and its implementation that will sound better than the basic NOS unfiltered DAC.
Sounds interesting ?
Cheers.
Ashok.
Hi Elso,
It's interesting to see that you did check the differences between the 8x version of the DAC. Now that you concluded that it could be the digital filter that might be the culprit which detracted from the sound - we could probably look for a DAC/digital filter and its implementation that will sound better than the basic NOS unfiltered DAC.
Sounds interesting ?
Cheers.
Ashok.
Re: For Elso.
Hi Ashok,
I tried also the SAA7220 digital filter (by accident).
I wish you good luck finding a better digital filter/DAC combination. I have given up that search.......
ashok said:
Hi Ashok,
I tried also the SAA7220 digital filter (by accident).
I wish you good luck finding a better digital filter/DAC combination. I have given up that search.......
Re: Up or Over sampling ?
It is not *that* very well written, and yes, oversampling and upsampling are one and the same from a mathematical point of view: increasing sample rate combined with low-pass anti-imaging filtering at the original fs/2.
But we already had a couple of threads on that one.
ashok said:
It is not *that* very well written, and yes, oversampling and upsampling are one and the same from a mathematical point of view: increasing sample rate combined with low-pass anti-imaging filtering at the original fs/2.
But we already had a couple of threads on that one.
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