Yes, and post link here , than we will know when to react 8)open up a new thread (feeler) in BM, that would keep the business out of this thread
Yes, and post link here , than we will know when to react 8)open up a new thread (feeler) in BM, that would keep the business out of this thread
Rochey said:OPA1612 is the way to go.
audiovisceral said:Personally not a big fan of the Burr Browns. I'd prefer Analog Devices opamps like what Raphael mentioned. AD8620, AD8599, AD823, etc.
But I doubt we'd ever all agree on opamps.
snipsnip said:Congratulations!
How do they sound?
audiovisceral said:I've contemplated the idea of discrete converters a bit in the past and I think there may be better places to integrate DOA's.
DOAs are big and expensive so I think it makes sense to put them where they will have the most audible effect. Technically perhaps this may be the ADC/DAC, since everything goes through them. But if the opamps are hardly applying any gain in the circuit, an IC might be near perfectly transparent, and you might be better off saving that discrete $$$ for use in the pres, comps, and perhaps eq's.
I don't know how much gain the opamps will be pushing through a circuit like this, so I don't know if it's even useful to think of it in these terms. But that's the first thought that runs through my head on the subject.
rkn80 said:Well, they do not sound. They simply play what you send to it.
To be honest, I find it a bit hard to describe how they sound. The only thing I can compare to is my Presonus Firebox. I could say that my DAC sounds better then the Firebox but that would not be fair because I think if you build something yourself you'll always find that your self-made unit sounds better because you made it.
Raphael
living sounds said:I think the difficult part is to get really low jitter. But as long as the converter is the master in a given setup at least sophisticated reclocking circuitry won't be needed. A very clean power supply is important, too.
Is there a way to configure the converter chips to use minimal filtering, especially no linear phase filters? I'd prefer some phase shift from an analog filter (in both DAC and ADC) to compromises in the impulse response caused by (real time) digital filters. I think one of the reasons for the Lavry Gold's big latency is to minimize these problems.
audiovisceral said:But I believe it would require a fair amount of added analog circuitry if you want a different analog filter point for each sample rate, unless everyone will agree on a fixed rate of say 96. Plus I imagine the troubleshooting to get that working seamlessly so the ADC isn't being overfiltered or being fed extraneous frequencies from a not steep or low enough filter might be difficult.
A simple optional set-frequency LPF (eg. 70-90 kHz) in the signal flow might work if the chips can be so configured.
rkn80 said:Hi,
sorry the project was delayed for some days now due to a lot of work in my real life job (new product coming out).
I did not concern about external vs internal clocking so much. I simply removed the clocking from the DAC board because the intention of the project is to keep it modular and flexible. So you can simply decide how to clock it by adding the right board for it. The dac pcb offers the option to select other formats then 24bit I2S or to set the dac as master or slave, it simply offers every configuration option you can find in the datasheet of the dac.
@Greogor: I'm in Bremen, where are you?
Raphael
living sounds said:An analog filter that works for 44,1 khz (= 20-22khz) should suffice IMO. The additional high frequency content at higher sample rates isn't imporant anyway.
A summary of this paper's findings. Column one refers to the figure showing the spectrum in question. Column two identifies the instrument. Column three gives the sound pressure level measured at the microphone. Column four gives the measured frequency extension: For instruments with harmonics, this is the highest frequency where harmonics are still present; for those without harmonics, the highest frequency where the sound is still at least 10 dB above the background. (See text.) The last column tells what percentage of the total energy is contained in the range between 20 kHz and the limit given in the previous column.
Code:Instruments With Harmonics Fig. Instrument SPL Harmonics Percentage (dB) Visible To of Power What Freq.? Above 20 kHz 1. Trumpet (Harmon mute) 96. >50 kHz 0.5 2. Trumpet (Harmon mute) 76. >80 " 2. 3. Trumpet (straight mute) 83. >85 " 0.7 4. French horn (bell up) 113. >90 " 0.03 5. French horn (mute) 99. >65 " 0.05 6. French horn 105. >55 " 0.1 7. Violin (double-stop) 87. >50 " 0.04 8. Violin (sul ponticello) 77. >35 " 0.02 9. Oboe 84. >40 " 0.01 Instruments Without Harmonics Fig. Instrument SPL 10 dB Above Percentage (dB) Bkgnd. to of Power What Freq.? Above 20 kHz 10. Speech Sibilant 72. >40 kHz 1.7 11. Claves 104. >102 " 3.8 12. Rimshot 73. >90 " 6. 13. Crash Cymbal 108. >102 " 40. 14. Triangle 96. >90 " 1. 15. Keys jangling 71. >60 " 68. 16. Piano 111. >70 " 0.02
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