Support Head-Fi.org by
starting all of your
Amazon.com shopping by
clicking here.
____________________________________________________________________
Today's Featured Head-Fi Blog: Jude's Blog
____________________________________________________________________
Please help
support Head-Fi by becoming a Contributing Member
CLICK
HERE -- Contributing Members, thank you
for your generous support! --
Thanks for clearing that up. That was my understanding from our email exchange, but after Elias' comment I started wondering whether there was some issue with the way Windows was handling things. My guess is he was referring to the kmixer resampling.
What I don't understand is why you went NOS. Async removes the jitter issue, but what does that have to do with oversampling?
Since I don't use upsamplers ASYNC is the mode for me.
This statement seems to connect two unrelated things, NOS and asynch mode.
There's no reason to make a DAC NOS whether one has low or high jitter or uses asynch or other modes. There are two cases with NOS: either there's no analog filter, which can be OK within the DAC with a non-feedback high bandwidth analog stage, but then is almost guaranteed to cause TIM and intermodulation related problems in the power amplifier or headphone amplifier, or have an analog filter which since it can't be infinitely steep to filter an image beginning immediately above the band, will alias image energy into the audio band. And even if the whole system has extremely high bandwidth into the MHz (including the speakers) so that images are not a problem, you end up relying on the very imperfect lowpass filter of the human ear, which is a problem given the references posted here and elsewhere that ultrasonic energy can influence perception.
__________________
Chat with us live at #diyaudio on irc.rizon.net !
"Good people do not need laws to tell them to act responsibly, while bad people will find a way around the laws." -Plato
On the question of: Why does the DAC1 re-sample to 110 kHz?
Here is why: it is the highest frequency to maintain the full oversampling of the D-A chip. EVERY D-to-A chip on the market cuts the oversampling rate in half to accommodate 192 kHz. This will also implement a different type of digital low-pass filtering which is inferior to the filter used at and below 110kHz.
This is also why most recording engineers don't use 192 kHz. The higher bandwidth seems appealing, but the stat-of-the-technology is such that 192 kHz conversion is actually inferior to 96 kHz.
Also, the DAC1's oversampling ASRC and resulting 110 kHz sample rate reproduces 96 kHz signals much more faithfully then a D-A converting the original 96 kHz signal. This is because the Nyquist frequency is on the slope of the filter (attenuated, but not completely). This is undesirable for two reasons. The first reason is the Nyquist frequency is not faithfully converted to analog (ie, the analog bandwidth of 96 kHz conversion is actually less then 48 kHz). With the DAC1, the full bandwidth of a 96 kHz signal can be faithfully reproduced. The second problem with 96 kHz conversion is the frequencies at and above Nyquist (48 kHz and up) are not completely attenuated, so some aliasing and imaging will occur. With the 110 kHz upsampling and conversion in the DAC1, the frequencies below 55 kHz are not in danger of being aliased.
If I remember correctly, the DAC1 uses AD1853. That has internal oversampling (called interpolation filter in the datasheet). Are you bypassing that when feeding the 110 kHz in, or does it get 4x or whatever by the DAC chip's internal filter?
__________________
Chat with us live at #diyaudio on irc.rizon.net !
"Good people do not need laws to tell them to act responsibly, while bad people will find a way around the laws." -Plato
When you listed possible DAC's you might buy, you mentioned that the Bel Canto has the same UltraLock system as the DAC1. This is not true. They call their system UltraClock (which sounds strikingly similar to UltraLock...perhaps just coincidence ). More importantly, however, it is not the same circuitry. I am not familiar with the effectiveness of their jitter reduction technology, but I can assure you that the DAC1's jitter reduction technology is much more then a plug-and-play solution. Their are several design considerations that require meticulous engineering to achieve.
You also mentioned that it has a better input stage then the DAC1. Can you elaborate on this a little bit?
Also, please see my previous post concerning 192 kHz.
Thanks,
Elias
ps. I realize that the thread has moved on to new topics, but I don't want to leave any questions unanswered.
As for the jitter in a stock DAC-1, this is addressed in their design by using asynchronous upsampling like many other DAC's. It certainly does reduce the jitter of the incoming stream, probably better than many DAC's.
However, this technique does not totally eliminate jitter IME. When I remove the upsampling chip and replace it with an I2S interface driven by my FIFO reclocker, the audible jitter is noticable lower.
Steve N.
Steve,
What method did you use to determine the jitter attenuation?
What method did you use to determine the jitter attenuation?
Thanks,
Elias
In this case, I relied on my ears. Like I said "audible". It is not interesting to me to have a lot of measurements if they dont correlate to what I can hear. I let my ears be the final judge.
You can throw stones at me all day on this, but after all the smoke clears, it is what is audible that matters IMO.
In this case, I relied on my ears. Like I said "audible". It is not interesting to me to have a lot of measurements if they dont correlate to what I can hear. I let my ears be the final judge.
You can throw stones at me all day on this, but after all the smoke clears, it is what is audible that matters IMO.
Steve N.
Steve,
Don't worry, I don't enjoy stone-throwing matches either. As much as I am capable, I try to limit my discussion to constructive discussion, not destructive.
I agree that no audio equipment can be judged on measurements alone. However, I do have doubts as to how reliably one can distinguish jitter performance well below -100 dBu just by listening. What source material were you listening to?
does the dac1 sound any different if you feed it a 24/192khz signal than if you feed it a 16/44.1. signal. i can't tell if i can really hear a difference. maybe because it just converts everything into 110khz anyways?
music_man
It depends on the source material. If something was recorded at 44.1 kHz/16-bits, then sample-rate converted to 192 kHz/24-bits, then there will be no difference at all. There are no bandwidth advantages because when the audio was recorded, it was low-passed at 22 kHz. Upsampling to 192 kHz will not add any more audio information above 22 kHz (except distortion). Also, the increase in word-length will not add any resolution because there will be no new information in the newly added 8 LSB's.
On the other hand, if something was recorded at 192 kHz/24-bits then down-sampled to 44.1 kHz/16-bit, then it will sound different. This is because the bandwidth will be reduced to 22 kHz and the noise floor will increase to 16-bit levels and resolution will decrease to 16-bit levels.
I hope I answered your question. If not, please follow up.
If I remember correctly, the DAC1 uses AD1853. That has internal oversampling (called interpolation filter in the datasheet). Are you bypassing that when feeding the 110 kHz in, or does it get 4x or whatever by the DAC chip's internal filter?
The filter in the D-A chip is frequency shifted to 55 kHz to not interfere with the Nyquist frequencies (see above post concerning why the DAC1 re-samples to 110 kHz).
This, effectively, replaces the D-A filter with the SRC filter which is a much higher quality filter.
I would like to expound on why 192 kHz conversion is not a good idea.
The filters in a D-A chip ideally pass all audio below the Nyquist frequency and block (filter) all audio above the Nyquist frequency. In reality, however, there will be some audio below the Nyquist that is being filtered some, and there will be some audio above the Nyquist which is not filtered enough. The latter is very dangerous because those frequencies will be aliased and cause distortion.
Here's the problem with 192 kHz: the filter used for 192 kHz is of far less quality. This is true of ALL D/A chips on the market. What happens is this: the filter cut-off becomes less defined, causing audio below Nyquist to be attenuated. And, more importantly, AUDIO ABOVE NYQUIST IS NOT FULLY ATTENUATED!! The filter does not do its job as well at 192 kHz.
Another 'real-life' limitation to these filters is amplitude ripple. This means that the audio below Nyquists will have ripples in amplitude across the frequency spectrum. This is equivalent to inaccurate frequency response. This is also something that happens in all D-to-A chips, some more so then others.
The problem with 192 kHz: the ripples in amplitude become much more exaggerated when filtering 192 kHz signals. Consequently, the frequency response is much less accurate and distortion goes up.
This is why most all converter designers and recording engineers don't recommend 192 kHz. The chip technology has not provided the means to effectively convert 192 kHz without these problems.
So, the trade-off for the extra analog bandwidth is an increase in aliasing and frequency-response distortion. Although some people may 'enjoy' listening to 192 kHz more then lower rates, it is not as accurate, objectively speaking.
). More importantly, however, it is not the same circuitry. I am not familiar with the effectiveness of their jitter reduction technology, but I can assure you that the DAC1's jitter reduction technology is much more then a plug-and-play solution. Their are several design considerations that require meticulous engineering to achieve.
Linear Mode
