Hi,
from where does a 20bit DAC chip take those 4 extra bits ?
Interpolation from oversampling in the digital filter ?
Isn't resolution lost at high frequencies ?
Bernhard
from where does a 20bit DAC chip take those 4 extra bits ?
Interpolation from oversampling in the digital filter ?
Isn't resolution lost at high frequencies ?
Bernhard
You are right about the oversampler. The input signal to an oversampler may be 16bits/44.1 while its output may be 20bits/352.8.
In one implementation, the digital oversampler keeps the original samples and fills in 7 interpolation values between two originals. With 16bit input, the originals are simply shifted 4 bits up while the generated interpolation samples are truncated or rounded down to 20 bits.
If the DAC doesn't feature a built-in analog filter, it shouldn't care about the data source, and thus be capable of reproducing any 20-bit signal you throw at it. (As long as you don't exceede its maximum update frequency etc.)
Greetings,
Børge
In one implementation, the digital oversampler keeps the original samples and fills in 7 interpolation values between two originals. With 16bit input, the originals are simply shifted 4 bits up while the generated interpolation samples are truncated or rounded down to 20 bits.
If the DAC doesn't feature a built-in analog filter, it shouldn't care about the data source, and thus be capable of reproducing any 20-bit signal you throw at it. (As long as you don't exceede its maximum update frequency etc.)
Greetings,
Børge
If you have 16 bit in you can't get more out in total quality but you can get better filtering, more ideal with some interpolation tricks. CD-qualilty stays CD-quality no matter what you do but you can come closer to the theoretical limit.
So a 20 bit DAC like PCM63 converts lots of samples that have values which are somewhere between original 16bit values.
Beside positive effects regarding filtering etc. , doesn't that reduce resolution / loose detail ?
If a picture is interpolated, quality does not get better.
It can be blown up without blowing up pixels, but some detail is lost when 2 pixels were black and white and the space between is filled with shades of grey after interpolation.
Also after resizing to the original , detail is lost.
Beside positive effects regarding filtering etc. , doesn't that reduce resolution / loose detail ?
If a picture is interpolated, quality does not get better.
It can be blown up without blowing up pixels, but some detail is lost when 2 pixels were black and white and the space between is filled with shades of grey after interpolation.
Also after resizing to the original , detail is lost.
In a digital filter (i.e. the oversampler) the actual 16-bit signal is convolved with the filter coefficients. These coefficients themselves are described with an accuracy of , say, 16 up to 32 bit. The resultant stream is meaningful in 32-48 bits, and these are prior to output reduced to 20 bits, because, of course, no DAC in our universe can reproduce these.
Doing digital filtering on a 16bit signal while restricting the output to 16 bit would indeed harm the information in the stream, i.e. the resolution.
Outputting an original 16 bit stream at 20 up to 24 bits is almost lossless.
Doing digital filtering on a 16bit signal while restricting the output to 16 bit would indeed harm the information in the stream, i.e. the resolution.
Outputting an original 16 bit stream at 20 up to 24 bits is almost lossless.
does this mean that most of the cd players, that use 16 bits dacs with oversampling, are loosing resolution?
Hi Bernhard,
Have you read this thread ?
http://www.diyaudio.com/forums/showthread.php?s=&threadid=28814
It should answer your questions 😉
Have you read this thread ?
http://www.diyaudio.com/forums/showthread.php?s=&threadid=28814
It should answer your questions 😉
From the above thread:
So, there is error and in case of 8 x os the error will be 1/8 LSB if a nearest neighbour sample is taken instead of the right one ?
I know how disturbing this sounds. But we must remember a couple things : One, a bandlimited signal is NOT free to do whatever it pleases between Nyquist samples ... in fact, it's pretty smoothly varying between those samples. And two, the more samples we can calculate beyond the Nyquist rate, through a very accurate interpolation process, the more adjacent samples will look the same. These are simple, indisputable facts of life.
But no matter what, grabbing the most recently calculated interpolated sample, instead of the exact sample required at Fs_out, will introduce error. The question is ... How far do I have to interpolate, by what integer, so that grabbing the most recent sample will give me acceptably LOW error ??? We can never achieve perfection here, but we can get ARBITRARILY CLOSE ... simply because the higher I interpolate, the lower the error
So, there is error and in case of 8 x os the error will be 1/8 LSB if a nearest neighbour sample is taken instead of the right one ?
Beware of comparing apples with oranges. That thread was about Asynchronous sample rate conversion. What takes place in an oversampling filter like the DF1704 is different.
rfbrw said:Beware of comparing apples with oranges. That thread was about Asynchronous sample rate conversion. What takes place in an oversampling filter like the DF1704 is different.
You are right.
Still the question:
Is there some detail lost using 16bit in 20bit out digital filter + 20bit converter 😕
I think there must be a relation between the n times oversampling your filter applies, and the x more bits it has at its output compared to its input
rfbrw said:Beware of comparing apples with oranges. That thread was about Asynchronous sample rate conversion. What takes place in an oversampling filter like the DF1704 is different.
Well, from what I recall, oversampling could be seen as a particular case of ASRC, but feel free to correct me. For starting point, jump to post #8 of the mentionned thread.
I read the whole thread and it is still not clear to me.
Is there some detail lost using 16bit in 20bit out digital filter + 20bit converter ?
Is there some detail lost using 16bit in 20bit out digital filter + 20bit converter ?
Bernhard said:I read the whole thread and it is still not clear to me.
Is there some detail lost using 16bit in 20bit out digital filter + 20bit converter ?
with how much times oversampling?
Yes some detail is still lost, however it is not very much.
For example: You need to solve 10/3.
At "16bit" you can only use whole numbers, so the result will be 3.
At "20bit" you can use 2 decimals, so the result will be 3.33. A 100 times more precise than with "16bit", but still not 100%perfect.
Next problem, in order to hear difference the least significant bit in a 20bit system you need a dynamic range for you entire system of 120db or better.
So you will always lose some detail, but at some point you will not gain anything by getting more detail.
For example: You need to solve 10/3.
At "16bit" you can only use whole numbers, so the result will be 3.
At "20bit" you can use 2 decimals, so the result will be 3.33. A 100 times more precise than with "16bit", but still not 100%perfect.
Next problem, in order to hear difference the least significant bit in a 20bit system you need a dynamic range for you entire system of 120db or better.
So you will always lose some detail, but at some point you will not gain anything by getting more detail.
Bricolo said:
with how much times oversampling?
This seems not to be important.
For me it is important n bit in >>> n+x bit out
Bernhard said:
This seems not to be important.
For me it is important n bit in >>> n+x bit out
I think it is simportant
IMHO, the more oversampling you apply, the higher x has to be
Bernhard said:
Still the question:
Is there some detail lost using 16bit in 20bit out digital filter + 20bit converter 😕
What do you mean by detail?
CheffDeGaar said:
Well, from what I recall, oversampling could be seen as a particular case of ASRC, but feel free to correct me. For starting point, jump to post #8 of the mentionned thread.
It goes without saying that at some point 'its all filtering' or 'its all oversampling' but that does not preclude the use of completely different processes. Then again perhaps you have a method of making the DF1704 and its ilk support arbitrary oversampling ratios.
rfbrw said:What do you mean by detail?
Some signal information lost because of interpolation ?
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