IVX
Major Contributor
I'm an ordinary chinese worker. Aliens nowadays look like that https://en.wikipedia.org/wiki/Grigori_Perelman
Then how did you come into possession of alien technology that beats an AP by 5-6dB?I'm an ordinary chinese worker. Aliens nowadays look like that https://en.wikipedia.org/wiki/Grigori_Perelman
pkane -Here's Multitone Analyzer noise computation for APU+6dB notch into Cosmos ADC 192k (SMSL DO300ES 96k). Without bin subtraction, but with APU compensation curve applied.
Noise is -135 dBA as measured seems in line with your estimates:
View attachment 359356
On the chart, the result is indexed to 0dB, so what you're eyeballing (140dB) is relative to the amplitude of the signal. SFDR is reported in dBFS.pkane -
Eyeballing, the SFDR seems to be ~140 dB (H3), rather than the reported 149.4 dB. Please explain.
I believe that spectral free dynamic range should be calculated in dBc, since it portrays the dynamic range between the carrier and largest spectral artifact. That's how I've always calculated it. But, perhaps I'm mistaken.On the chart, the result is indexed to 0dB, so what you're eyeballing (140dB) is relative to the amplitude of the signal. SFDR is reported in dBFS.
I believe that spectral free dynamic range should be calculated in dBr, since it portrays the dynamic range between the carrier and largest spectral artifact. That's how I've always calculated it. But, perhaps I'm mistaken.
So the default +2db is actually better and should be used?Not by much. I'm measuring this with E1DA's APU notch + Cosmos ADC, so the result is different than Amir's as this setup has lower noise than APx555.
At -0.0dB THD+N is -127.3dB, and at +2.0dB it's -127.8, so 0.5dB better at +2dB setting.
View attachment 356565
View attachment 356566
SMSL DO300 DAC Review
Minimun Phase is the default PCM filter setting after a reset of my DO300. I did some tests these days and in my opinion Linear Fast and Apodizing are very similar in my audio chain. The Linear Fasts have only a very small bass boost. which is your chain?audiosciencereview.com
#228
My Wattson Emerson doesn't operate over 192kHz : should I understand I'll not be able to use "1 filter off" setting
What happens if i try?
Thanks by advance for your support
It's also my understandingSo the default +2db is actually better and should be used?
Thanks for the test!
How are you reaching the correct filter off configuration? You should not have the +4dbHi Finally no problem using the Filter Off configuration with a 44.1 or 48kHz input (I guess you should filter above 20kHz)
Seems a little more vivid (whenever correcting -5 or -6dB for 4.2dB announced)
perhaps because filtering is rejected above 20kHz..
Another explanation ? (apart subjectivity we cannot exclude)
You may have a look at this postIt's a pretty complex topic, and I don't remember all the details myself, but:
- the volume level it's not causing the issue
- digital filters are in this case needed to avoid imaging(not aliasing), and so the distortion caused by the intermodulation of these ultrasonic images.
so, I don't really know why we have that +4db, but for sure the no-filter is not intended to be used with low frequency material for the aforementioned problem, plus the not good reconstruction of the sine. here you can see how a digital vs nos filter works, and how a 2x oversapled signal is making a better sinewave out from a nos dac http://archimago.blogspot.com/2018/11/nos-vs-digital-filtering-dacs-exploring.html
Then, what you hear from this dac with NO Filter, it's what actually the source would be heard everywhere if a reconstruction filter would not be implemented(both digital or analog).
This mode would work only with an oversampled source to avoid issues, possibly at 8x for a 44.1 file.
you can try it yourself, use a Jriver, foobar or whatever player you use on pc, load a 44khz music file in it, use a builtin oversampler and rise it to 352.8kHz so that the player does the oversampling, turn off the filter from the DAC.....and booom the distortion is gone too
Hi, just switching on the pcm filter on the first position "Filter Off" (instead of previously used (Linear Fast")How are you reaching the correct filter off configuration? You should not have the +4db
Actually I'm no expertHi, just switching on the pcm filter on the first position "Filter Off" (instead of previously used (Linear Fast")
Nothing else on the DO300
And filtering with LR48 at 20khz on the DCX 2496
It's highly probable I do something wrong as it seems, my experience is not thé common one.
Hope I can benefit from your expertise
Thx by advance
Thanks for this litterature. In fact, measuring with REW I do not sée thé HF drop and frequence response AT 48, 96, 192 Filter Off are all the same and do not differ from thé Linear Fast curveYou may have a look at this post
This mean you are actually filtering the file before reaching the dac? This actually may be the same, but honestly I don't understand why doing so instead of using dac filtersThanks for this litterature. In fact, measuring with REW I do not sée thé HF drop and frequence response AT 48, 96, 192 Filter Off are all the same and do not differ from thé Linear Fast curve
I'm using 3 voices amplification , so limiting signal sent to the treble to 3khz 20khz thru active filtering.
No ways to measure the pollution the author is mentionning...
In fact using Butterworth 48dB filtering at 20khz because attenuation slightly later than Linkwitz Riley