Question: Cheap pure sine wave genertor 1khz? - Page 1
Forum section: electronics / testgear

author: innkeeper
author_rank: Unknown
timestamp: January 09, 2020, 09:29:59 pm
content: I'm looking to generate a low thd sine wave to do some THD testing with a DSA.the internal generator on my DSA is not clean enough for any critical THD testing.id like it to be 1khz, and will be used for line level input to amplifiers. so 2.1v out max or less. probably down to like 1.2v in that range.happy building up a circuit or buy one pre built on the cheap. any ideas?i could swear i've seen such projects in the past but came up empty in my recent searches.

author: Yansi
author_rank: Super Contributor
timestamp: January 09, 2020, 09:33:28 pm
content: Use a decent computer soundcurt plus a suitable software. Easy, done.You can then easily change test frequency, and even do test like a two tone (IMD).

author: RoGeorge
author_rank: Super Contributor
timestamp: January 09, 2020, 09:42:20 pm
content: If you want to build a generator, this type is hard to beat in performance:https://www.eevblog.com/forum/projects/low-distortion-audio-oscillator-stabilized-via-trigonometric-identity/There are some linear application notes, too, from the '70s or the '80s, explaining this topology. Can't find them right now.

author: David Hess
author_rank: Super Contributor
timestamp: January 09, 2020, 09:55:36 pm
content: The simplest single operational amplifier Wien Bridge can achieve 0.0025%. Doing better requires buffering the output and adding common mode suppression which will get you to 0.0003%. SeeLinear Technology application note 43.Doing better than that gets complicated but seeLinear Technology application note 67for an example of what is required to test a modern audio ADC to 18 bits or better.

author: Kleinstein
author_rank: Super Contributor
timestamp: January 09, 2020, 10:02:07 pm
content: A good sound card is an easy solution.For a DIY solution the best bet is likely a Wien bridge oscillator of some kind with really low THD amplifiers. There a quite a few old plans and application notes. With fixed frequency, there is no real need for the fancy amplitude regulation using instant amplitude measurement, as much less action from the amplitude stabilization is needed. There is an LT appl. Note to get super low THD by combing 3 OPs for each inverting amplifier.

author: AG6QR
author_rank: Frequent Contributor
timestamp: January 09, 2020, 10:13:12 pm
content: If working at a fixed frequency, you can take any oscillator and improve it by suppressing its harmonics with a low-pass filter having a corner frequency somewhere between the fundamental and second harmonic.Low-pass filters for a fixed frequency aren't that hard or expensive to make.

author: innkeeper
author_rank: Unknown
timestamp: January 10, 2020, 04:13:28 am
content: i think i found a decent thread on this.https://www.diyaudio.com/forums/equipment-and-tools/205304-low-distortion-audio-range-oscillator.htmlmost of the solutions are along the lines of what david hess and ag6qr recommended.theirs a guy there building decent units but not available at the moment. and a few others that have pcbs for designs etc...the tread is like over 8000 messages long so thats just what i found by skimming it.certainly no simple solutions,seems there talking about designs with an output of -140db or better thd+n .. that is along the lines of what i am looking for.

author: Fungus
author_rank: Super Contributor
timestamp: January 10, 2020, 08:18:45 am
content: Quote from: innkeeper on January 09, 2020, 09:29:59 pmHow low?There's tons of cheap programmable generators out there if you don't mind adding a microcontroller to set the frequency.eg.https://www.ebay.com/sch/i.html?_nkw=AD9850https://www.analog.com/media/en/technical-documentation/data-sheets/AD9850.pdfQuote from: AG6QR on January 09, 2020, 10:13:12 pmProbably easier than building something with no distortion.

author: Fungus
author_rank: Super Contributor
timestamp: January 10, 2020, 08:35:14 am
content: There's a whole bunch of other kits/modules on eBay, too:https://www.ebay.com/sch/i.html?_nkw=DDS+Function+Signal+Generator+Module+DIY

author: Gyro
author_rank: Super Contributor
timestamp: January 10, 2020, 10:49:03 am
content: There are assembled versions available on ebay too, but populating one of these bare boards with genuine components (and maybe lower distortion opamps) ought to get you close to what you need. It uses a J-FET for amplitude control. Jim Williams found that there was some channel resistance modulation when using a J-FET, that could be trimmed out with a little local feedback[Analog Circuit Design - Art, Science and Personalities, Chapter 7 (Max Wein, Mr Hewlett and a Rainy Sunday Afternoon)].You might get better than the quoted -124dB with some careful component value matching anyway...https://www.ebay.com/itm/Low-Distortion-Audio-Range-Oscillator-1KHz-Sine-Wave-Signal-Generator-PCB-Board/262844562138

author: RoGeorge
author_rank: Super Contributor
timestamp: January 10, 2020, 12:16:49 pm
content: Quote from: Gyro on January 10, 2020, 10:49:03 amThat one is a good example, too, available online:http://www.introni.it/pdf/Williams%2007%20-%20Book%20Chapters.pdf

author: Vovk_Z
author_rank: Super Contributor
timestamp: January 10, 2020, 12:24:40 pm
content: The best 1kHz oscillator I know (with best price/thd ratio) is Victor's ultra low distortion oscillator. You can find him at diyaudio.com, he sells sometimes readymade modules via ebay. It is Wien bridge type, and it is cheap.Here it is.But the answer depends on what TS mean at "pure sine wave"? How much pure?DDS signal gereators need something like Arduino to program them.

author: Grandchuck
author_rank: Super Contributor
timestamp: January 10, 2020, 09:43:43 pm
content: Low distortion is easy but very low distortion is not. The best one (circuit) here is showing 0.0016% but is probably better since the old HP analyzer is not great with very low distortion measurements. The circuit is described here:http://www.redcircuits.com/Page82.htmThe problem with this one is that the dual potentiometer should have very low tracking error and such a part is hard to find.Having tried several different kinds of very low distortion circuits, my conclusion is they are all 'fiddley".

author: innkeeper
author_rank: Unknown
timestamp: January 10, 2020, 09:57:22 pm
content: Quote from: Gyro on January 10, 2020, 10:49:03 ami've actually ordered one of these .. figured its a start and maybe i can tweak it. i was thinking of just replacing the tl071 for something quieter. i didnt know aout that article, i will defiantly try and find it ... i figure it would do until i can make or get somthing better. like vidnic's on diyaudio...i;ll look into that article if i can find it.

author: Gyro
author_rank: Super Contributor
timestamp: January 10, 2020, 10:09:44 pm
content: Quote from: Grandchuck on January 10, 2020, 09:43:43 pmThis is the first time I've seen independent verification of (the performance of) that circuit, thanks. [Edit: I was planning to build it a few years back but a Topward TAG-4005 came up really cheap on ebay that manages quite a bit better than its specified 0.005%.]Construction and trimming would be a considerably easier for one (or a few) spot frequencies of course.Fiddley is the word, that is demonstrated very well in that Jim Williams chapter.

author: Gyro
author_rank: Super Contributor
timestamp: January 10, 2020, 10:13:10 pm
content: Quote from: innkeeper on January 10, 2020, 09:57:22 pmA good start for sure.RoGeorge actually found an online copy of that chapter - he linked it above.

author: innkeeper
author_rank: Unknown
timestamp: January 10, 2020, 10:15:59 pm
content: Quote from: Gyro on January 10, 2020, 10:49:03 amthanks RoGeorgehttp://www.introni.it/pdf/Williams%2007%20-%20Book%20Chapters.pdfalso found it here - in the book Gyro pointed tohttp://s1.nonlinear.ir/epublish/book/Analog_Circuit_Design_Art_Science_and_Personalities_0750696400.pdffascinating article, Jim Williams is always insightful. I think this is the first thing i've read of his that was not an application note. i really like how he walks you though the process of what he did and why.

author: innkeeper
author_rank: Unknown
timestamp: January 10, 2020, 10:28:02 pm
content: Quote from: Yansi on January 09, 2020, 09:33:28 pmcan you give me an example setup with its thd+n figures?

author: Gyro
author_rank: Super Contributor
timestamp: January 10, 2020, 10:42:33 pm
content: Quote from: innkeeper on January 10, 2020, 10:15:59 pmThere's another, equally good book, also edited by Jim Williams, in that EDN series called 'The Art and Science of Analog Circuit Design'. I have paperback copies of them both and find them nice to settle down with on the occasional rainy afternoon.

author: MasterT
author_rank: Frequent Contributor
timestamp: January 10, 2020, 10:53:54 pm
content: Wien-SG_Elektor-1994-12.pdf(131.28 kB - downloaded 397 times.)I've build one recently, to test my 18-bits differential AD7984 board. I changed OPA to LM4562, and use 7370-18V-0.04mA lamp. Circuits does need "manual" tuning to get the lowest THD, but accounting simplicity it's not an obstacle. Results thd-2 ~0.0004% (-110dB) and thd-3 ~0.0002 (-114dB) at 4kHz. Not sure about 100kHz, ADC itself has thd-3 ~0.002% at this frequency.My attempts to replace lamp with CdS photocell or thermistor MF52 in the amplitude control loop were really disappointing. Despite what they say in app note, CdS is highly non-linear material, and may works only in the <100 mV range, making control loop very limited in the dynamic range sense.

author: Fungus
author_rank: Super Contributor
timestamp: January 11, 2020, 12:21:46 am
content: Quote from: innkeeper on January 10, 2020, 09:57:22 pmOk, so -124dB is only "a start"? Got it.

author: innkeeper
author_rank: Unknown
timestamp: January 11, 2020, 03:12:15 am
content: Quote from: MasterT on January 10, 2020, 10:53:54 pmthat is amazing for such a simple circuit, i;ll have to throw that together and put it on the analyzer and check it out. thanks

author: innkeeper
author_rank: Unknown
timestamp: January 11, 2020, 03:13:46 am
content: i've hit up victor from the diyaudio forum to see if he has any of his boards available. he is getting better then -150 Db.

author: MasterT
author_rank: Frequent Contributor
timestamp: January 11, 2020, 04:02:09 am
content: Quote from: innkeeper on January 11, 2020, 03:13:46 amWhat circuits? I've seen link to e-bhttps://www.ebay.com/itm/Low-Distortion-Audio-Range-Oscillator-1KHz-Sine-Wave-Signal-Generator-PCB-Board/262844562138,let's call it JFET-Wein, description says:Quotedoesn't look right, in my math -120dB is 0.0001% ( or 1ppm), 0.0064% is about -84-85 dB, worse than cheapest sound card has. And NE5532 ? Common, be serious.Or is it CdS-Wein?http://www.redcircuits.com/Page82.htmProbably, just simplified version ofhttps://www.analog.com/en/design-center/evaluation-hardware-and-software/evaluation-boards-kits/dc1858a.html#eb-overviewI trust AD, performance is outstanding, but same with a price and complexity.Though -120 only:Table 1. Amplitude and Offset Options for DC1858A (SNR and THD Taken at f OSC = 2kHz)Range (V) R22 (kΩ) R18 (kΩ) C11 (μF) Sine Out (V P-P ) Offset (V) SNR (dB)* THD (dB)*0 to 2.5 14 36.5 0.47 2.25 1.25 100 –1190.5 to 4.5 22.6 12.1 0.47 3.6 2.5 103 –1210 to 4.096 23.2 17.4 0.47 3.7 2.048 103 –1210 to 5 28 12.1 0.47 4.5 2.5 104 –118±2.5 28 ∞ 0.47 4.5 0 104 –119±5 56.2 ∞ 0.47 9 0 107 –117±10 118 ∞ 0.1 18 0 100 –107page 3https://www.analog.com/media/en/dsp-documentation/evaluation-kit-manuals/dc1858af.pdfAnd most resistors precise 0.1% tolerance, they can't get -120dB with CdS even for single Tone generator using common 1% parts, and adjustable frequency doesn't go along with precision parts at all.

author: innkeeper
author_rank: Unknown
timestamp: January 11, 2020, 04:48:24 am
content: @masterTvictor sells is an updated version of this:https://www.diyaudio.com/forums/equipment-and-tools/205304-low-distortion-audio-range-oscillator-21.html#post3116055update: i think this is the latest schematichttps://www.diyaudio.com/forums/equipment-and-tools/205304-low-distortion-audio-range-oscillator-743.html#post5873578i am sure there is a more up to date schematic in that thread, but that thread is over 8000 messages longhere is the web page he put up just wiht the specshttps://viccc42.wixsite.com/uld-audio

author: hamster_nz
author_rank: Super Contributor
timestamp: January 11, 2020, 05:06:21 am
content: I've had others recommend using the PC audio too, so I though I'ld give it try to see. I just wrote a quick hack to split 1kHz out of my Intel NUCs onboard audio, with the volume set to just below where bad harmonic distortion sets in (about 0.5V pk-pk) Spectrum is below.If anybody has any good cheap sound dongle recommendations, I'm all ears...Contents of Makefile:Code:all : sin./sin | aplay -r 48000 -c 2 -f S16_BEsin : sin.cgcc -o sin sin.c -Wall -pedantic -O4 -lmContents of "sin.c":Code:#include <stdio.h>#include <stdint.h>#include <math.h>int main(int argc, char *argv[]) {int i = 0;while(1) {double angle = i * 2 *M_PI / 48.0;double s = sin(angle);uint32_t u_sample = (s+1.0) * 16384;int32_t i_sample = u_sample - 16384;putchar((i_sample>>8) & 0xFF);putchar((i_sample>>0) & 0xFF);putchar((i_sample>>8) & 0xFF);putchar((i_sample>>0) & 0xFF);i = (i == 47 ? 0 : i+1) ;}}

author: andy3055
author_rank: Super Contributor
timestamp: January 11, 2020, 05:06:36 am
content: I use this:https://www.szynalski.com/tone-generator/

author: innkeeper
author_rank: Unknown
timestamp: January 11, 2020, 06:22:02 am
content: do you know the thd+n of the output signal?

author: Marco
author_rank: Super Contributor
timestamp: January 11, 2020, 07:44:25 am
content: Vojtěch Janásek hasa circuitdescribed on his website, but maybe more importantly he also has a relatively simple circuit to knock down the fundamental frequency so you can measure the distortion with lower quality equipment.

author: RoGeorge
author_rank: Super Contributor
timestamp: January 11, 2020, 03:24:59 pm
content: Quote from: hamster_nz on January 11, 2020, 05:06:21 amNeat!I didn't know there is a tool (aplay) that can take numbers from a pipe and throw them at the sound card.Questions:- is "i_sample>>0" there for readability only, and optimized out during compilation, or is it for something else?- where from is the FFT screen capture, and why does it show a -14 dBV DC component?- did you used some loopback between play and rec inside the audiocard, or physical cables?

author: cncjerry
author_rank: Unknown
timestamp: January 12, 2020, 08:43:34 pm
content: I've found that you have to spend a fair amount on a sound card to get anything near the performance of Williams, Wien bridge, etc. First off, the sound card has to be external and then you have USB noise to contend with. You can fix some of that with good cables and opto isolated USB adapters and lifted grounds. But even then, and I've been messing with this stuff for 45yrs, it's tough to find something that will outperform those old lightbulb circuits.One thing I didn't see was the DSA you are testing? I just bought an HP 3562a DSA and that is good for a boat anchor but the best it will do in range is 120db. It is listed at 80db but there is a way I read someplace to get it to 120db. It must be averaging or something or maybe that was the minimum signal. I've only had it a few days.One other thing, stop posting all the HP distortion analyzers or all I have to buy one soon. I've wanted one and every time I get it out of my system, someone posts another picture.Lastly, I was wondering is that I have HP Selective Level meters that are really accurate. I don't know how low they go but I was wondering if I can use one of them for THD testing. I'll have to go back and see if they are flat down to 1khz.Thanks,Jerry

author: cncjerry
author_rank: Unknown
timestamp: January 12, 2020, 09:11:42 pm
content: when you are generating sin waves you have to be aware of the large number problem. depending on how long you let that code run, sooner or later you are going to glitch. You need to generate the wave in an array so that the start at zero and the end will perfectly meet and then write the array blocks out. I say blocks because I don't think, and I've not thought about it in a while, you can get a 'perfect' sine wave out of one block at 48k. Here's a simple test. Calculate two 48k blocks generating the code as you did sequentially and then compare block 0 with block 48,000 and on to see if you have the same numbers. Then write the file to disk and plot the spectrum before looping it out the card and back in.You should also plot your noise floor to see how much junk you have coming from the sound card.

author: David Hess
author_rank: Super Contributor
timestamp: January 13, 2020, 02:15:09 am
content: Quote from: cncjerry on January 12, 2020, 08:43:34 pmNo sound card will produce a signal as clean as the second example I gave which was designed for *testing* audio ADCs so necessarily has lower distortion.

author: MasterT
author_rank: Frequent Contributor
timestamp: January 13, 2020, 04:25:42 am
content: I checked out AN67, sorry for my anglaise, but it's the most ridiculous /stupid circuits I've seen in a years. 180 dB, with funny two diodes?

author: Marco
author_rank: Super Contributor
timestamp: January 13, 2020, 06:32:50 am
content: It does look like an April fools joke doesn't it ... you can't just have a controlled current source drive current into the overcompensation input of an opamp and not explain what the hell that's supposed to be doing.

author: innkeeper
author_rank: Unknown
timestamp: January 13, 2020, 07:07:40 am
content: Quote from: cncjerry on January 12, 2020, 08:43:34 pmI am looking for my manuals with the application n notes.i know it has 150 DB of measurement range.but the operators manual does say 80db dynamic range.not sure how that plays out doing thd....if its limited to 80dbdamn if that is the case that sucks.. but well might have to find another method when i get down to low thd readings,i know there are some tricks, but..thats all in the application notes not in the operators manual.I hope to play with it a bit again this week

author: cncjerry
author_rank: Unknown
timestamp: January 13, 2020, 05:28:58 pm
content: I went back and looked at my DSA and pulled out the section, attached 2nd, that speaks to the dynamic range in an automatic function that I wasn't familiar with. I also attached the noise floor but have to figure out how that compares on the DSA. I don't know what DSA you are using, but I know some DSAs have lower noise floor and if you want to look at the 3rd harmonic, for instance, you just focus the analyzer there. You can't see the entire spectrum but you can calculated the THD by summing the harmonics.This guy has a well formatted formula for THD calcs:http://www.onmyphd.com/?p=thd.total.harmonic.distortionHe also has the thd+noise calc there.I'm surprised the HP analyzers go that low automatically calculating THD. I really want one, just haven't been able to find one locally.Jerry

author: peter-h
author_rank: Super Contributor
timestamp: January 13, 2020, 06:12:33 pm
content: I would use two EPROMs to get 16 bit words, drive them from a synchronous counter, and feed the 16 bits of data into a 16 bit audio DACI designed such a thing for a friend of mine's (!) final year univ. project in 1978. We used fusible link PROMs back then, only 8 bits of data so not smooth, but frequency changes are instant and there is no amplitude bounce.If you use a good DAC, no missing codes, you can get THD below 0.005%. Probably less by filtering the output.Won't be dirt cheap admittedly.I believe there is a way of connecting up shift registers in a feedback arrangement which gives you a sin(x) output, but you still need a DAC.

author: Fungus
author_rank: Super Contributor
timestamp: January 13, 2020, 07:23:49 pm
content: Quote from: peter-h on January 13, 2020, 06:12:33 pmDid you build the counter yourself using D-type flip-flops?These days we have microcontrollers that are a lot cheaper than EPROMs and can do the counter, the feeding of the DAC, andeverything.

author: trobbins
author_rank: Frequent Contributor
timestamp: January 13, 2020, 09:04:30 pm
content: There are good external USB soundcard/interfaces around - I have a 15 year old EMU 0404 USB - and although I haven't used one, it seems stock product like FocusRite 2i2 are now pretty cheap new.REW software allows insertion of selected harmonics (2 to 9) in to the test signal from its tone generator, with settable magnitude and phase. REW also provides bandwidth limiting and distortion component measurements of the loopback spectrum.Using my EMU0404 USB as DUT with loopback, I did some testing last year and had 0.0006% 2nd and THD at 389.6Hz test tone when limiting bandwidth to just include 2nd harmonic. I can then null that out (0.0001%) with -104dB of added 2nd H with 90 deg phase shift. I effectively did the same for higher order harmonics.This method will work for any typical soundcard, so the generation of a very low harmonic distortion waveform is not limited to the raw performance of the soundcard itself. As such, a cheap or even zero-cost, and quick setup can be prepared by most.

author: hamster_nz
author_rank: Super Contributor
timestamp: January 13, 2020, 09:27:28 pm
content: I think this will be my next project - a < $10 low distortion 1kHz generator.Thinking of using a Wein Bridge, but with an LED coupled to an LDR to replace the classic light build to control gain. Bit worried about that being non-linear though, and spoiling the result.

author: Gyro
author_rank: Super Contributor
timestamp: January 13, 2020, 09:51:44 pm
content: The best ones do seem to use LED+LDR. There's effectively zero voltage modulation of the LDR resistance.

author: MasterT
author_rank: Frequent Contributor
timestamp: January 13, 2020, 10:00:47 pm
content: Quote from: trobbins on January 13, 2020, 09:04:30 pmThis trick IS limited by performance of the adc. I don't know what kind of chipset inside EMU0404, but I doubt it has THD+Noise level below 0.0001% (-120dBc). I'm working on similar project, doing software correction THD by magnitude and phase tracking in antiphase on stm32F767 dual dac DDS generator. The problem I have is non-linearity of AD7984, about -110 dBc, so can't break this barrier and my DDS always has THD above this flour.

author: MasterT
author_rank: Frequent Contributor
timestamp: January 13, 2020, 10:11:48 pm
content: Quote from: Gyro on January 13, 2020, 09:51:44 pmPage 35, distortion.

author: trobbins
author_rank: Frequent Contributor
timestamp: January 13, 2020, 10:41:02 pm
content: Quote from: MasterT on January 13, 2020, 10:00:47 pmThe technique also has to contend with the measurement null being the result of both DAC and ADC, and not just DAC, so some deduction needed to confirm that DAC result by itself is nulled.Noise level will depend on a few factors, so not straightforward to report on. The mid-band noise level for that EMU soundcard is down around -130dB from full-scale signal, and uses 24-bit 192 kHz AKM AK5385A ADC and AK4396 DAC.Still, the trick can also be used to null out any additional harmonic distortion from added devices up to the point of DUT connection, so could be applied to an add-on signal amplifier if needed for the DUT and test environment. The power of the technique is simplicity and ability to make a mid-range silk purse out of a sows ear.

author: Vovk_Z
author_rank: Super Contributor
timestamp: January 13, 2020, 10:47:34 pm
content: Quote from: MasterT on January 11, 2020, 04:02:09 am- this one isn't Victor's, it's unnamed RC-oscillator. It really have 0.006%, but have a potential to go to 0.001% with smal change (resistor "120-150k" change to value 27-30k if I'm not wrong). Best side of it that it is variable frequency with stable amplitude.Victor's oscillator is constant frequency and ultra low distortion.

author: f5r5e5d
author_rank: Frequent Contributor
timestamp: January 13, 2020, 11:29:42 pm
content: much work on ADC/DAC THD cancellation:https://www.diyaudio.com/forums/equipment-and-tools/328871-digital-distortion-compensation-measurement-setup.html

author: MasterT
author_rank: Frequent Contributor
timestamp: January 14, 2020, 12:12:17 am
content: Quote from: trobbins on January 13, 2020, 10:41:02 pmDreams, dreams... Based on a DS from thishttp://www.datasheetcatalog.com/datasheets_pdf/A/K/5/3/AK5385A.shtmlTHD of the appointed ADC is -103 dBc. Period. Nothing could be measured below this level. More than that, depends on a driver analog front end , likely results never cross -100dB . I see herehttps://www.soundonsound.com/reviews/emu-0404they honestly print 0.001% THD+N. No where close to -130dB.

author: rhb
author_rank: Super Contributor
timestamp: January 14, 2020, 03:10:20 am
content: Unless I were prepared to launch into a serious development project, I'd use a sound card.Write .WAV files covering the test amplitudes and frequencies.Play them back through the sound card.Record L->L, L->R, R->L & R->RSolve for the input and output errors of each channel and correct. At each frequency and amplitude there are 4 equations and 4 unknowns.You can get almost arbitrary accuracy via signal processing with large datasets.Otherwise, I'd recommend building a single frequency version of Jim Williams last version from "Max Wien, Mr. Hewlett and a Rainy Sunday Afternoon" using well matched fixed parts. But then you have to design and test the distortion analyzer.Having lived through the THD wars in audio gear in the late 70's I feel obliged to point out that the distortion of the best sound transducers (microphone and speaker) are an order of magnitude larger than a cheap audio IC.Have Fun!Reg

author: MasterT
author_rank: Frequent Contributor
timestamp: January 14, 2020, 03:32:06 am
content: Quote from: rhb on January 14, 2020, 03:10:20 amNo, you can't. The issue is that harmonic's magnitude & phase (thd-n) are not constants. Doesn't matter how big data sets are, how accurate equation written or what precision of the floating math. The data becomes a junk after a few milliseconds, because ADC is drifted, always changing - there is no single static constant magnitude & phase below -100 dBc with audio ADC, or -110 with precise SAR. Same with a DAC. During debugging my project, I came to conclusion that uCPU has to continuously monitor each thd-n and make adjustment "on the fly" .Calibration & look-up table doesn't work, actually it does for stm32f767 internal 12-bits DAC with initial start-up thd-3 about -70 dBc, LUT could fix this value to -95/-100, but than magic show begins.

author: trobbins
author_rank: Frequent Contributor
timestamp: January 14, 2020, 03:49:53 am
content: Quote from: MasterT on January 14, 2020, 12:12:17 amCirca 130dB plot taken whilst having fun with the harmonic null technique:https://dalmura.com.au/static/Loopback%20distortion%20trim.JPG

author: rhb
author_rank: Super Contributor
timestamp: January 14, 2020, 03:30:55 pm
content: Quote from: MasterT on January 14, 2020, 03:32:06 amI'm shocked! Truly shocked! You mean there is noise in the data? We'll have to give up!In seismic exploration it is routine to get 40-60 dB or more improvement in the signal to noise ratio. I have seen many examples where the initial SNR was so bad that it did not appear that there was any data at all, just noise. But after some DSP the signal is 30-40 dB above the noise.The data of interest is the reflected energy from a rock layer over 40,000 ft below the surface arriving 10-12 seconds after the source impulse was initiated. That is *only* possible because we exactly match the phase from sample to sample while summing hundreds of thousands of samples into each output sample. Every sample has a different propagation delay which has to be determined in order to sum the correct samples.I've suggested this every time someone has come along wanting to do audio THD measurements. With a good 24 bit, 192 kHz card one should get quite spectacular results. I've got a page of mathematics floating around on which I analyzed the problem to verify that my assertions were correct.If someone wants to do it, I'll help them over the bumps, but I'm interested in RF, not audio and am not interested in doing a grad school semester project to simply prove I can. I stopped doing that sort of thing 30 years ago.Once you can send the integer 6e6 followed by 191,999 zeros to the sound card and record it, standard seismic software (Seismic Unix et al) can take it from there. Record a loop of [6e6,zeros(1,191999)] sent as integers to the card for 24 hours and you can suppress random noise by 49 dB. Add 6 dB for each extra day of recording. Systematic and correlated noise will take more processing, but summing 86,400 192,000 sample FFTs will get you a *very* long way towards the goal. Systematic error is handled by the L-L, L-R, R-R & R-L error solution for the input and output ADC & DAC errors. Once random and systematic error terms have been suppressed one can address the correlated noise. How hard that is is dependent upon to nature of the noise. Until someone shows me the result of the first two steps I can't say how to do that. It could easily turn into many weeks or months of work. This is *NOT* something you can do on an MCU. This requires a fast workstation with lots of memory and is only practical in recorded time. Even a many core desktop is unlikely to do it in real time.The preceding will require three 16.5 GB data files recorded over 3 days. Straight thru, cross over and DUT to a splitter to both channels to test a single output amplitude. Each amplitude will be different. However, testing every amplitude is not practical over the range of 24 bits.For an intro to signal processing intended for scientists and engineers:An Introduction to Digital Signal ProcessingJohn H. KarlAcademic Press 1989The canonical mathematical reference on Wiener-Shannon-Nyquist analysis (the predominant techniques used):Random DataBendat and PiersolWiley 4th ed 2010Those cover all the work up to 2010. Work by Donoho and Candes in 2004-2008 has started another advance of equal importance. However, the computational burden is much higher, so it will not replace Wiener-Shannon-Nyquist. And at current oil prices the seismic industry has completely collapsed. So no money is being spent developing it further in the oil industry which pioneered DSP in the first place.Have Fun!Reg

author: MasterT
author_rank: Frequent Contributor
timestamp: January 14, 2020, 04:41:58 pm
content: What you described is a pure oversampling, based on sqrt(S/N), when averaging 4 samples improves SNR by 2. It's only applicable with random Gaussian noise. THD is different species, it does not go down with averaging, or increasing fft size. And adc noise is not random.

author: trobbins
author_rank: Frequent Contributor
timestamp: January 14, 2020, 11:44:19 pm
content: MasterT, the RightMark testing in link below shows a similar noise floor level below signal FS in the THD+Noise test result.https://www.dearhoney.idv.tw/SoundCard/EMU0404USB/E-MU0404USB_96K_Out.htm

author: MasterT
author_rank: Frequent Contributor
timestamp: January 15, 2020, 12:54:35 am
content: What you think, is if there is a line at -130 than your sound card has THD+N = -130? It's wrong. Look at this picture:Taking test with bigger fft-size, you can drop a line to -160 or -200. There is fft processing gain, 10xlog(M/2). So, getting back to your diagram, you should look at the highest peak except fundamental, and it's slightly below -100. Period. Software thd correction would not be possible with thispeace of junkbetter than 0.001%. Not 0.0001% as you says earlier

author: trobbins
author_rank: Frequent Contributor
timestamp: January 15, 2020, 02:03:25 am
content: The RightMark plot shows raw harmonic performance, as I was also able to reproduce.However, if then the signal source is adjusted to add nulling harmonic levels (as per link in earlier email, and repeated below) and the spectrum shows a particular harmonic descending to the noise floor then is the harmonic now suppressed to circa -130dB below fundamental, or is there a disconnect somewhere ?https://dalmura.com.au/static/Loopback%20distortion%20trim.JPG

author: rhb
author_rank: Super Contributor
timestamp: January 15, 2020, 03:15:07 am
content: Quote from: MasterT on January 14, 2020, 04:41:58 pmYes, grasshopper. The Gaussian random noise goes down. The THD and other noise processes do not. I am glad you have grasped my point.This is very useful as it allows us to to address the non-Gaussian random and correlated noise processes. Of which there are many. Each of which requires appropriate measures. And in many cases, a significant amount of work simply to identify and very often years of effort to eliminate. But eventually one gets to the point that precise and accurate measurements can be made despite the noise. At least in theory. Having spent many years fixing bugs and dealing with weird data, I suspect that much of the time the idea is correct, but the implementation is wrong.It's rather like an onion. There are many layers. But before you can reach one, you have to remove all the outer ones.Most of the EE community has a view of DSP which is 10-20 years out of date. The oil industry was investing millions on DSP for seismic processing before digital data could even be collected. They knew digital data was coming. They were paying for it. TI was created to build recording equipment for GSI, their parent company.My PhD supervisor at Austin had been head of research at GSI before he left for academia. He was one of the 7 members of the Geophysical Analysis Group consortium working under Norbert Wiener with funding from the oil industry in the early 1950's. Enders Robinson and Sven Treitel, whose papers were collected as the "Robinson & Treitel Reader"and eventually merged into a book, were the major apostles as my supervisor had very poor writing skills.Sadly, even within Amoco, which employed Sven, things were often done incorrectly. I ran into this a few months into my career in seismic. I called Sven at the labs in Tulsa to ask him about it. He told me I was correct. And added, "But the prophet is often not believed in his own country", or words to that effect.In summary, yes, it can be done. But you have to know how.Havfe Fun!Reg

author: RoGeorge
author_rank: Super Contributor
timestamp: January 15, 2020, 03:43:37 am
content: It doesn't matter if the noise is Gaussian or not, but it must be uncorrelated with the measuring signal, and must have a zero average for the integrating time.

author: MasterT
author_rank: Frequent Contributor
timestamp: January 15, 2020, 04:48:46 am
content: Quote from: trobbins on January 15, 2020, 02:03:25 amWhat the point in generation distorted signal, that eliminates distortion introduced by adc later on? Makes you happy? This is not what this topic about - to generate clean sine wave.Objective is synthesis pre-distorted digital data that would wipe out non-linearity of the DAC Obviously, you need right tool, THD meter that capable to see results. And something like TI's ads127l01 , sadly I hate a package.

author: trobbins
author_rank: Frequent Contributor
timestamp: January 15, 2020, 05:27:30 am
content: MasterT, the point is that the DAC of a soundcard can be set up to generate a much lower distortion signal than can be generated by just the raw DAC output signal of the soundcard when asked to generate a sinewave.How do you then measure the DAC's performance? Well that is the point of the loopback test, but as indicated earlier that then includes a distortion contribution from the soundcard ADC. So the loopback spectrum plot shows that the DAC plus ADC performance can exhibit a measured distortion profile where the HD components are down at the noise floor level. For a 15 year old soundcard, that noise floor level is circa -130dB below the fundamental, but it appears you don't have confidence in that measurement process (and have yet to explain that succinctly imho).If a much better, and more expensive ADC was available for making measurements, then that should be able to gauge what the actual soundcard DAC harmonic distortion profile really is. And yes maybe some further tweaking of the added harmonic distortion magnitudes/phases (in REW) could achieve a better null. Unfortunately I don't have access to such equipment, so can't confirm the actual performance. But it does appear to provide the cheapest bang for your buck solution imho.

author: Marco
author_rank: Super Contributor
timestamp: January 15, 2020, 06:33:32 am
content: Quote from: trobbins on January 14, 2020, 03:49:53 amYou are correcting for the combined non-linearity of the DAC/ADC this way, which means you are adding real harmonics which the ADC can't "see".Still, it proves the DAC/ADC are stable enough to get 130 dB of consistency in between calibration ... and thereare waysto separate out the ADC non-linearity from the DAC non-linearity. It works over a small frequency band, but as I said last time analogue switches are cheap (and have vanishingly low distortion when feeding an non inverting opamp buffer, a couple 100 milli-Ohm of channel modulation in the switch does nothing with a >10^9 Ohm load).

author: MasterT
author_rank: Frequent Contributor
timestamp: January 15, 2020, 12:10:20 pm
content: Quote from: trobbins on January 15, 2020, 05:27:30 amI already says, thd -n magnitude & phase are variables. They vary with temp, voltage , input level, freq., and coupe hundreds other parameters. It's not clear, how long do you track your equipment to get this down to -130, but I think long time accumulation approach is not right way. Instantaneous, sporadic deviation would not be visible on long term statistical data. Fast, 1--2 msps >16bits adc offers much faster loopback control.I tried with CS4344 & CS5361 last summer, thought sigma-delta would get me better linearity than SAR AD798X, but for some reasons - bread-boarding, adc driver or volt/temp instability, I hardly could get -90 despite whatever they say in DS. Also possible, I got fake chips.

author: rhb
author_rank: Super Contributor
timestamp: January 15, 2020, 03:43:05 pm
content: A few points:1 bit = 6 dB But you need to use ENOB, not the output word size. I don't recall seeing 24 bit sound cards 15 years ago. 130 dB requires an ENOB of 21.67 bits which in turn requires a 24 bit ADC.In seismic we measure and correct for ADC skew, location accuracy, channel to channel gain and phase error. All these things are being controlled as tightly as possible. But if the work requires more accuracy it can be done in post processing. That's a very powerful concept.A common correction these days is water depth changes of fractions of an inch because of tides, currents and subsea topography. Those maps you see showing the topography of the deep ocean seafloor? Those are made by measuring the bumps on the surface of the sea by bumps in the seafloor from satellites. If your equipment is floating in the water you can't dynamically correct for changes in the water depth. The correction is done in processing by shifting everything to a common datum.Once the ADC and DAC errors have been measured using the loopback I outlined earlier, the input sine wave WAV file has a Wiener inverse filter applied to predistort the waveform so that analog errors are corrected before they happen. This is fundamental to modern cell tower transmitters.The ADC errors have their Wiener inverse applied after acquisition to ensure that the DUT actually sees a pure sine wave.You don't need a better sound card to measure a sound card. just do the loopback test I mentioned and the algebra will remove the sound card errors once you characterize them.If anyone wants help implementing the sound card correction send me a PM. I have no interest in doing it myself right now as I need to build a 16' x 12' screened room for a new lab area first. I do hope to work my way through Jim Williams' paper some day. To do that in a meaningful way I have to measure the THD at each iteration of the design. I do not want to buy an HP distortion analyzer to do that. I don't care all that much about audio equipment beyond using it. I have been playing guitar for 50 years.Just for fun, I checked and I have directed people to "Max Wien, Mr. Hewlett and a Rainy Sunday Afternoon" in 16 posts on EEVblog. Like many of my contemporaries, I revere Jim Williams.Reg

author: trobbins
author_rank: Frequent Contributor
timestamp: January 16, 2020, 12:04:40 am
content: March 2006 product entry, so circa 14 years (although the software took some time to mature):http://ixbtlabs.com/articles2/proaudio/emu-0404-usb.html

author: rhb
author_rank: Super Contributor
timestamp: January 16, 2020, 01:18:44 am
content: Boy, I bet those were $$$.I bought a Korg 1600 Mk II DAW circa 2004-2005 after despairing of finding a USB device which would work with Linux. So I had been looking at audio interfaces a lot just before those appeared. I bought a Tascam USB unit, but the software was so flaky I couldn't make it work. So I bought a packaged system that did work even if the UI was agony for ~$1300. 16 bit 44.1 kSa/s only.

author: schmitt trigger
author_rank: Super Contributor
timestamp: January 16, 2020, 01:36:30 am
content: Subscribing to thread

author: Grandchuck
author_rank: Super Contributor
timestamp: June 26, 2021, 08:24:07 pm
content: Thread revival time. Got some new ICs: OPA2810. Time to try another low distortion oscillator circuit and used an old and simple one based on a circuit from a 1994 Elecktor magazine. The modified schematic is shown here along with the THD reading. I changed R3 from 2.05 k to 2.2 k and eliminated C3 and C4 (I might add them later and see if it makes any difference). I also added a 5 k potentiometer on the output.The construction is also shown with a close view of the IC and its adaptor. There is an 8-pin dual-inline socket on the board so it is easy to pop in other ICs such as a TL0821 (tried one and it works, but with a bit more distortion).The actual THD is below 0.001% because the 8903B residual noise and distortion are too high for this measurement. I see a new rabbit hole coming

author: pcprogrammer
author_rank: Super Contributor
timestamp: June 27, 2021, 05:56:05 am
content: Very interesting stuff about the 1KHz sine generation hereI wrote code running on an STM32F303 for making a very good sine wave. I myself do not have the equipment to check the quality, but a friend of mine, whom I wrote it for, told me the distortion is very low.One can check it out here:https://github.com/pecostm32/STM32F303_Sine_Square_Generator

author: Grandchuck
author_rank: Super Contributor
timestamp: June 27, 2021, 12:45:30 pm
content: Thanks. Lots of bits makes for low THD:THD(%) = 1/(2^N) x 100 (where N is the number of bits)That is the ideal case. In practice, the THD is higher. i.e. my Siglent 14 bit generator gives about 0.008% (ideally, 0.006%).

author: David Hess
author_rank: Super Contributor
timestamp: June 28, 2021, 02:15:15 pm
content: Quote from: Grandchuck on June 27, 2021, 12:45:30 pmThat assumes that the linearity is commensurate with the resolution, which is rarely the case, and even achieving better than 0.01% requires attention to the audio path after the DAC. Typically at least 2 bits are lost to non-linearity of the DAC but it is 3 or 4 bits with some devices which is revealed by integral non-linearity specifications or 8 or 16 counts.Testing of 16 bit and greater ADCs is a real challenge and where the various "parts per billion" oscillator designs come into play.

author: Kleinstein
author_rank: Super Contributor
timestamp: July 24, 2021, 07:07:30 pm
content: Quote from: Grandchuck on June 27, 2021, 12:45:30 pmThe formula includes quite some simplifiaction. Looks like it assumes very high BW to also include the very high harmonics. With a limite BW (even the usual 22 kHz with Audio) quite a bit of the digital step noise gets removed, especially with an oversampling DAC, which is the norm.In addition there are INL errors from the DAC and following stages. So real world noise can be higher of lower.The Elector Wien bridge circuit is the very basic one, not specially made for low noise. So it hase relatively high demand on the OPs. It is more like good for testing the OPs than actually generating a very low noise sine. For lowest destortion it often helps to have 2 OPs in the oscillator itself with the OPs working as inverting vonfiguration and this with zero (or near zero) common mode voltage. This avoids the distorion from the limited CMRR.

author: rhb
author_rank: Super Contributor
timestamp: July 24, 2021, 08:21:09 pm
content: The canonical schematics are in:"Max Wein, Mr. Hewlett and a Rainy Sunday Afternoon" by Jim Williams. Because he understood transistors at the most basic level, he was able to achieve better performance than the best instrument available.Have Fun!Reg

author: Grandchuck
author_rank: Super Contributor
timestamp: July 27, 2021, 06:59:00 pm
content: Found on ebay:https://www.ebay.com/itm/224549490206?hash=item3448314e1e:g:ziEAAOSwhvJhAApbVery impressive distortion specs.

author: rhb
author_rank: Super Contributor
timestamp: July 27, 2021, 07:16:39 pm
content: Wow! They claim no measurable distortion using a 24 bit sound card.If it meets the specs it's an excellent soundcard calibrator. Though switched frequencies in octave steps from A 22.5 Hz would be far better.Have Fun!Reg

author: Grandchuck
author_rank: Super Contributor
timestamp: July 28, 2021, 05:36:04 pm
content: Has anyone built one of these?https://www.ebay.com/itm/184691239077?hash=item2b0074a8a5:g:q6wAAOSw4tNgTC70Schematic shown here.

author: Verticon
author_rank: Contributor
timestamp: July 28, 2021, 09:29:49 pm
content: Yes, I've built several of these state variable oscillators. But mostly with wide frequency range from 10Hz to 100kHz. At 1 kHz they have a typical distortion level below 0.001% even with OP-AMPs like 5534. The 1980 Tektronix SG505 (wide frequency range) with a similar oscillator technique has a typical 1kHz distortion of 0.0003% which corresponds to -110dB.So if they mean 0.0064% (which is only -84dB) it is an easy task. But even -124dB should be not out of reach with this oscillating principle tweaked for a fixed frequency.

author: trobbins
author_rank: Frequent Contributor
timestamp: December 28, 2021, 12:17:30 am
content: A recent enhancement on the approach that uses a soundcard to generate %HD levels down at the 0.00001% level is that REW introduced a 'coherent averaging' capability mid last year. That option allows the noise floor for assessing HD levels to be dropped by quite a few dB, and hence assists the nulling process of HD contributions. It should also assist any process that tries to insert a notch filter in the loopback path as a way to confirm that the DAC is actually generating such low levels of HD%, compared to the DAC + ADC generating such low levels.

author: Grandchuck
author_rank: Super Contributor
timestamp: December 28, 2021, 03:18:45 pm
content: trobbins, do you happen to know if REW software plus an interface like the Focusrite Scarlett Solo would be capable of measuring THD levels around -120dB or better?Thanks

author: Marco
author_rank: Super Contributor
timestamp: December 28, 2021, 04:47:39 pm
content: Quote from: trobbins on December 28, 2021, 12:17:30 amIn principle a low pass filter is enough to distinguish adc distortion from dac distortion, though a notch filter simplifies the calculation.

author: pandy
author_rank: Contributor
timestamp: February 08, 2023, 09:05:04 pm
content: Apologies for reviving this thread but as no one before proposed something that will be probably simplest, cheapest yet very clean source of for example sine (but not only) signal - circuit using some memory (uC?), sequencer (uC or dicrete logic) and additional elements (like for example CD4000 series powered with for example 15V to increase overall SNR). In memorysingle bit delta-sigmai.e. PDM single bit sine (or other) waveform is stored and sequentially outputted to single pin - i.e. serial out - with sufficient oversampling and wise embedded resources use probably simple uC with sufficient memory will be more than OK - with help of common SPI output it may deliver 1kHz at few MHz sample rate efficiently turning such uC into sinewave signal source with very high quality beyond typical 16..20 bit - something like SACD audio quality.2000 sine samples (single sine period) with 64 times oversampling will require approx 16KB table - 128 times oversampling will be 32K etc...Sine frequency can be changed by using different clock. There is possibility to use some tricks to reduce required memory and/or to increase sampled signal phase accuracy if signal is like sine - similar to tricks used in DDS/NCO.If uC is equipped with PWM then using high order delta sigma (multibit) very high quality signal reproduction can be created even with relatively low PWM resolution (<8 bit) but high sample rate - there is many possibilities to use digital resources and get high quality analog signal.To create PDM, open source software can be used for examplehttps://pcmdsd.com/Software/PCM-DSD_Converter_en.html

author: scopeman
author_rank: Frequent Contributor
timestamp: March 03, 2023, 03:44:44 am
content: Have you seen this? Looks like a great start to me:https://www.romanblack.com/onesec/Sine1kHz.htmSamW3OHM

author: jonpaul
author_rank: Super Contributor
timestamp: March 03, 2023, 04:29:32 am
content: Jim Williams 1970s app notes from National Semiconductor and Linear Technologyj

author: scopeman
author_rank: Frequent Contributor
timestamp: March 05, 2023, 05:54:17 pm
content: Looks like this is the ticket:Available in 1KHz -155 to -160dB Distortion level. A 10KHz version is slightly higher.https://viccc42.wixsite.com/uld-audioSam

author: Grandchuck
author_rank: Super Contributor
timestamp: March 05, 2023, 06:28:07 pm
content: There is also this:https://www.akitika.com/1kHzOscillator.htmlI have this one and it works well; however to verify its extremely low distortion specifications is a bit beyond my current setup. I am able (with REW and a Focusrite Scarlet Solo) to approach verifying that level of performance.

author: pandy
author_rank: Contributor
timestamp: March 08, 2023, 11:59:18 pm
content: If someone is interested then i've attached file (due limitations of upload extension is hex but in fact this is bin file with content of the 10 sine periods coded by delta sigma modulator like DSD) - file is small - 8192 bytes - technically it can be stored in some memory and after conversion from parallel to serial used to recreate sine with high quality - target sample rate should be around 6.5MHz to get 1kHz.Signal should have around 160dB SNR but everything depends on power supply and remaining circuitry. Probably small (8 pin AVR like?) uC with SPI should be able to play such sine.To generate file i used SoX fromhttps://audiodigitale.eu/repo/sox/- this SoX has added DSD converter, bellow commands i've used to create signal:Code:@sox --multi-threaded --buffer 131072 -S -V -D -r 1024 -t raw -e unsigned-integer -c 1 -b 32 -n sine_period.raw synth 1 sine 10 gain -n -0.0021125518979949800 stats statthis one create 1024 samples of the sinewave (10 periods) as 32 bit PCMCode:@sox --multi-threaded --buffer 131072 -S -V -D -r 44100 -t raw -e unsigned-integer -c 1 -b 32 sine_period.raw sine_period.wavthis one read previously created 32 bit PCM sinewave and store it as wav - this is optional step - wav is widely supported so generated sinewave can be easily inspected with help of the common audio editorCode:@sox --multi-threaded --buffer 131072 -S -V -D -r 44100 -t raw -e unsigned-integer -c 1 -b 32 sine_period.raw sine_period.datanother optional step - 32 PCM is converted to text file so it can be easily imported to for example spreadsheetCode:@sox --multi-threaded --buffer 131072 -S -V -D -r 44100 -t raw -e unsigned-integer -c 1 -b 32 sine_period.raw -t raw -e unsigned-integer -c 1 -b 8 sine_pdm.raw rate -s -a 2822400 sdm -f sdm-8another optional step - delta sigma bitstream is stored as bytes so it can be easily imported to common sample editor as normal 8 bit signal and later for example filtered to recreate sinewaveCode:@sox --multi-threaded --buffer 131072 -S -V -D -r 44100 -t raw -e unsigned-integer -c 1 -b 32 sine_period.raw sine_pdm.dat rate -s -a 2822400 sdm -f sdm-8optional - delta sigma bitstream is created as text file so it can be imported to for example spreadsheetCode:@sox --multi-threaded --buffer 131072 -S -V -D -r 44100 -t raw -e unsigned-integer -c 1 -b 32 sine_period.raw sine_pdm.dff rate -s -a 2822400 sdm -f sdm-8This one is non optional - delta sigma stream is created from 32 bit PCM and stored as DSD DFF file - after removing header (126 bytes) attached sine_pdm.bin was createdTo remove 126 bytes header i use small but nice hexeditor called XVI32http://www.chmaas.handshake.de/delphi/freeware/xvi32/xvi32.htmbellow script for hexeditor to cut first 126 bytesCode:ADR 0DEL 126EXIT

author: bozidarms
author_rank: Regular Contributor
timestamp: March 09, 2023, 11:53:15 am
content: Victor`s 1kHz is Nr.1 - can hardly be better.THD: -135dB -- 0.0000178%6dBV = 2Volt(eff) = 5,6Volt(s-s) = 8,2dBu

author: Grandchuck
author_rank: Super Contributor
timestamp: March 09, 2023, 03:31:13 pm
content: Thanks Boz! How did you achieve that low noise floor?

author: bozidarms
author_rank: Regular Contributor
timestamp: March 09, 2023, 05:03:11 pm
content: Hi GrandchuckQuoteThat is one RTX6001 Audio Analyzer and 1 Million-point FFT length

author: Martin72
author_rank: Super Contributor
timestamp: May 14, 2023, 06:30:45 pm
content: Only a few more components and then mine is ready.I still have to equip the notch filter board.

author: Martin72
author_rank: Super Contributor
timestamp: May 17, 2023, 09:53:30 pm
content: Oscillator board now complete assembled(except connections), for the notch filter board I thought I already have all the parts, but today I recognized that the OPA209 chip got the wrong package...Nevertheless, the other parts will be soldering tomorrow.I´m curious about the oscillator, will test/adjust it soon.

author: Martin72
author_rank: Super Contributor
timestamp: May 19, 2023, 09:27:17 pm
content: Notchfilter assembled...The package on the pcb is for SO8, a real small one - But the opa09 I got from mouser is even smaller..SO8 Variants are out of stock at the moment, so I must wait.Or assemble the existing one somehow on the board...

author: RoGeorge
author_rank: Super Contributor
timestamp: May 19, 2023, 10:06:33 pm
content: Worth a try, the smaller one might fit. Its middle 2 pins can touch their respective PCB pads. If not, bend them sideways a little, so they can be soldered directly to the middle 4 PCB pads. If the marginal pads are too close or touching the middle PCB pads, put some Kapton tape underneath the corner pins, then use thin wires to solder the corner pins to their corresponding PCB pads.

author: Martin72
author_rank: Super Contributor
timestamp: May 19, 2023, 10:41:56 pm
content: Hi RoGeorge,Got the same thoughts and when everything will go wrong (delivertime too long), I´ll try this out.Will be a nice surgery under the andonstar microscope..

author: RoGeorge
author_rank: Super Contributor
timestamp: May 20, 2023, 01:26:51 am
content: Indeed, under the microscope it's doable.Once I've soldered a chip same size as yours 4 times (the chip, a 16bits ADC for a thermocouple, was not working apparently), only to realize after replacing the chip for the 4th time that all the previous chips were in fact OK, just that there were reflections on some data lines while controlling the ADC.The shoddy soldering in the pic is the 4th one, where no care was spent for the look, believing at that moment that all the chips were bad anyway. It's been years since then, and never properly rework the soldering.