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E1DA Cosmos ADCiso Review

Rate this ADC:

  • 1. Poor (headless panther)

    Votes: 4 2.5%
  • 2. Not terrible (postman panther)

    Votes: 1 0.6%
  • 3. Fine (happy panther)

    Votes: 16 9.9%
  • 4. Great (golfing panther)

    Votes: 141 87.0%

  • Total voters
    162
But the selfprotection of the devices is not state of the art.
Particularly Cosmos ADC is very well protected. I sold nearly 2000pcs and had just 2-3 cases when users burned the input caps during a poweramp testing(XLR>20Vrms). 1 case when the handmade power supply was used and killed the USB bridge. APU's and Scaler's inputs are protected quite tricky, however, 10-100K Zin is more fragile anyway. Cosmos LPF has a low Zin again, and I believe nobody will complain about its protection ;)

PS: The reason why old vs new records/masterings sound different is trivial and not related ADC/DAC quality(even 10x times worse ADC than Cosmos will be 100.00% neutral for an LP ripping). The inherent part of an LP production is a compressor, while the digital record doesn't need to compress the dynamic range.
 
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Good to hear that. I worried they were a little fragile.
So overloading the ADCiso in low R setting, will not kill the input stage?
Lets say 10 volts AC wih 5 volts DC and 600R setting
 
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Maybe there will be a big jump, doubling the cost:
Using two ES9823(M)PRO devices in ESS Proprietary multi-chip mixing configuration, designers can achieve dynamic range performance exceeding 136dB

And the RAW output give sccess to both phases of differential input. So mono is 2 datastreams. Interesting. Hope @IVX will release a adciso mk2 and expose them
 
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well, the paralleling is fine(BTW, 9822 it seems also has some I2S features for that) but cascading, idunno. IMO, it is a 99.99% marketing feature for musicians or others who can't realize that such DR is fake arguable. It's similar to a DSD feature of 9822, it is there but.. made from PCM and what? So, I think we need to test the linearity of 9823 first(especially at 10kHz, after I took a look ESS recommended frontend). I noticed that TD+N specs were done at -1dbfs, and I worry a little. THD+N of a calibrated MONO 9823 is expected to be around 3db better than APx555, how many DIYers want even better and are ready to pay for that?
 
So, I think we need to test the linearity of 9823 first(especially at 10kHz, after I took a look ESS recommended frontend).
Like many times before, ESS datasheet circuits are usually far from optimum. Way to high output impedance at higher frequencies for those 9822 and 9823 final passive output RC filters. Much better is a feedback circuit similar to the one shown for the OPA1633 driving a PCM1804.
 
how many DIYers want even better and are ready to pay for that?
I also suspect it just marketing.
But if it is not and actually realizeable to get a 9 dB improvement with 2 chips I think some are willing to pay double.
But the analogs in the frontend may be to expencive to produce or impossible to realize?
I thought the ADCiso and 9039 dac was to good to be true
 
well, the paralleling is fine(BTW, 9822 it seems also has some I2S features for that) but cascading, idunno. IMO, it is a 99.99% marketing feature for musicians or others who can't realize that such DR is fake arguable. It's similar to a DSD feature of 9822, it is there but.. made from PCM and what? So, I think we need to test the linearity of 9823 first(especially at 10kHz, after I took a look ESS recommended frontend). I noticed that TD+N specs were done at -1dbfs, and I worry a little. THD+N of a calibrated MONO 9823 is expected to be around 3db better than APx555, how many DIYers want even better and are ready to pay for that?
The last 3% are usually more expensive to develop than the first 97%.
Therefore, such a device can cost twice, three times, or even four times as much.
Maybe not only do-it-yourselfers are interested in it and you also have to take the smaller number of pieces into account when setting the price.
 
A good studio microphone achieves a dynamic range of 140dB. (1uVrms(A) noise to 10Vrms max output)
Building an A/D converter to get this range into digital would be GREAT. Then you really only have to press record without any chance of clipping or burying your signal in noise.
 
A good studio microphone achieves a dynamic range of 140dB. (1uVrms(A) noise to 10Vrms max output)
Can you name one studio mic with 140dB DR or more?

From what I know, such high DR in mics is squarely in the realm of laboratory equipment (though happy to be corrected :D)
 
Can you name one studio mic with 140dB DR or more?
For example AKG C4000. (Datasheet is on the save side with 137dB but I measured them often enough at my time at AKG, when your preamp has a high enough input impedance you get 140dB dynamic range)
C414 should also come close. Some of the Lewitts should be able to do it cause of very low self noise. Rode NT1 - that's a 230,- microphone.
And I have an electronic design for microphones on my table which is able to do the 140dB dynamic range. A good JFet impedance converter has <1uVrms(A) self noise and max output depends on the current you use for your mic (cause phantom power goes down when you need too much current). With a proper output buffer (e.g. diamond transistor design) you can maintain these properties. Of course it can't provide the steady output current for a low impedance mic input at these levels but real signals have a crest factor, 15V peak is doable.
Studio mics are also specified with 0,5% THD as max SPL for a steady sine signal, you often get at least 3dB more for a peak signal.

Measurement mics have pretty high noise but crazy high max SPL - so while they can achieve this DR it's not very realistic for musical signals.
 
@IamJF, Do you believe that someone in a production studio gonna use the same mic and preamp with the same gain for a jazz vocal and hardrock drums?
Yes, this is the whole idea. No more gain adjustment to consider, and still always recordings that are neither clipped (or needlessly compressed if you are lucky) nor noisy.
Stacked ADCs are becoming more and more popular, for example in field recorders, and the movie sound crews love it! As do wildlife researchers etc.
 
IamJF, Do you believe that someone in a production studio gonna use the same mic and preamp with the same gain for a jazz vocal and hardrock drums?
What has this to do with the solution to have a wide DR A/D converter? You can use every mic you want with such an input.

I know that sound people love their preamps and it's a slow minded community, "new" ideas take time. Just connect your mics to the inputs and press record. That's THE concept for live music, no clipping. But also in the studio it would be nice to have and make things quicker and easier.

p.s.: Since I use Console 1 in the studio I am happy with linear and neutral preamps. You can get plenty of colour with these channel strip plugins and enough "glue" for the mix. No, it's not the SAME as preamp XY. But it's what you need for the mix/master.
 
Yes, this is the whole idea. No more gain adjustment to consider, and still always recordings that are neither clipped (or needlessly compressed if you are lucky) nor noisy.
Stacked ADCs are becoming more and more popular, for example in field recorders, and the movie sound crews love it! As do wildlife researchers etc.
Maybe some exotic cases exist, however, we forgot about the splitpoint artefacts. The level where a low-gain ADC is replaced with a high-gain one and vice versa. Did you see how people complained about a short clipping clicks in CS43***, which implemented a similar gain-switching tech? Do you think cascaded ADCs would be flawless? ;)
From the marketing point of view, it makes sense, I agree, but it is not about audio quality.
 
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A "full range" converter with no clipping possible is a game changer for pro audio, not a marketing fluff. These cheap 32bit float devices are a marketing thing but a good start in the right direction.

Getting a stacked A/D right is a big challenge! But a task which can and will be solved over time and enough R&D in this direction.

For home audio ... why even bother with A/D ;-). Current ones are easy good enough for analog sources when you need it. Just take care about the right gain staging, that's a way bigger problem with HiFi. Often >10dB S/N are given away with crazy preamp gains and analog inputs not matching the source.

system before gain structure.gif
system after gain structure.gif
 
Maybe some exotic cases exist, however, we forgot about the splitpoint artefacts. The level where a low-gain ADC is replaced with a high-gain one and vice versa. Did you see how people complained about a short clipping clicks in CS43***, which implemented a similar gain-switching tech? Do you think cascaded ADCs would be flawless? ;)
From the marketing point of view, it makes sense, I agree, but it is not about audio quality.
It of course all depends on the implementation.

In a proper stacked multi-path ADC with, say, 4 range sections and fine-tuned level- and time-based crossfades, the noise-floor change is minimal and irrelevant. Same will certainly be the case for a similar multipath DAC.

When using a single DAC and changing the digital and analog gains reciprocally like it is done in the Cirrus DACs, well, that's a cheap single-path approach with sub-optimal processing and therefore it has audible and well-measureable artifact.
 
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