Parallel dac to double sample, possible? - Page 1
Forum section: electronics / projects

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: October 31, 2024, 01:46:14 am
content: I came upon this articlehttps://www.diyparadiso.com/teksten/issue1-2/solnodac1.htmIt says, read bellow picture.Is this possible?If I double write the data in a dac and double the bit rate, will the image go from 22khz mirror to 44khz?

author: PCB.Wiz
author_rank: Super Contributor
timestamp: October 31, 2024, 02:07:33 am
content: Quote from: Kokoriantz on October 31, 2024, 01:46:14 amYou mean usetwo DACsand interleave the updates, then sum the DAC's outputs ?Sure, that can work, but you need to check the settling time / slew rates of the DACs for your application.It sems complicated tho, to double the parts count for a modest gain ?If their slew rate is good enough to permit interleave, perhaps they can be overclocked anyway, saving a DAC ? Or just buy a better dingle DAC ?

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: October 31, 2024, 03:28:57 am
content: Thank you for the reply.I don't understand how by giving the same output but alternated with 2 dacs can do anything different than one.I simulated by LT interleaving zero or repeat the sample but inverse value, the image is always mirrored to 22khz although the rate has doubled.

author: Georgy.Moshkin
author_rank: Frequent Contributor
timestamp: October 31, 2024, 04:10:02 am
content: Sample time at 22kHz is approximately T=1000ms/22000=0.045ms. I think that the idea is to run the second DAC with a T/2=0.023ms time offset. This means that the combined DAC output can change value at rate F=1/(T/2)=2*1/T which is essentially doubles sampling rate to 44kHz. In other words, while second DAC is synchronized with the first DAC, they change the output not simultaneously, but in an alternating order.

author: PCB.Wiz
author_rank: Super Contributor
timestamp: October 31, 2024, 04:39:16 am
content: Quote from: Kokoriantz on October 31, 2024, 03:28:57 amYou originally saiddouble the bit rate, which I took to mean alternative data values.If you do not add any extra information, you are correct, all you have added, is a t/2 delayed copy.

author: SiliconWizard
author_rank: Super Contributor
timestamp: October 31, 2024, 04:42:42 am
content: I for one have a hard time making sense of the linked "article". If anyone has better luck than I do.That said, a couple things have been said in this thread, so I'll be dealing with two different approaches when "paralleling" DACs:- Just paralleling, or rather effectively summing the outputs of two DACs running *synchronized* at the same sample frequency: that's something that is sometimes done and used to be sort of "popular" in older audio gear with the "early" 16-bit audio DACs which were barely OK. The idea is that of averaging: it increases the SNR (by a factor of sqrt(n), where n would be the number of summed outputs, so for 2 outputs summed, that would be by 3 dB at best; 6 dB for 4 outputs, etc). Note that it requires as good sync as possible between DACs, otherwise the benefits of averaging are lost and one can make things actually much worse in terms of SNR and distortion.)- Interleaving (as I suppose may be the intent here?) is something routinely done for ADCs, for instance in oscilloscopes at least past a certain sample frequency. As long as the delay between each ADC is fixed and with as low jitter as possible, then you effectively can multiply the sample frequency by n, the number of ADCs. Doing the same with DACs, though, would be ridden with pitfalls, as all DACs would be permanently summed, and if samples are interleaved, that's likely to give you much worse SNR. But you'd still need to properly interpolate samples - just duplicating the same value would make things worse.I don't see the point whatsoever, if the goal is to just increase sample frequency. You'd get much better results with a single DAC supporting double the sample frequency, and in the audio range, they are everywhere these days up to like 768 kHz. So, that makes no sense. But (and again I can't make good sense of it) in the article linked, it's possibly pretty old and written at a time when audio DACs were rare and expensive. But if the underlying idea is to find an idea for doubling the sample frequency without requiring interpolation (/ sample rate conversion), then I'm afraid it looks like a dead-end.

author: moffy
author_rank: Super Contributor
timestamp: October 31, 2024, 05:58:02 am
content: Repeating the same sample twice, but delayed 1/2 a sample period will increase THD for in band products, you are basically adding distortion to the signal. Oversampling requires interpolation to create those extra samples while minimising the distortion. Second, any mismatch between the two DACs gains, even if outputting just the same sample will contribute to a rise in THD.

author: gf
author_rank: Super Contributor
timestamp: October 31, 2024, 07:24:26 am
content: Quote from: moffy on October 31, 2024, 05:58:02 amIt is basically equivalent to 2x oversampling withlinearinterpolation, which still does not eliminate the image in the 22...44kHz region, and which also introduces an undesired frequency response roll-off in the 0...20kHz passband. A decent interpolation filter (almost brickwall) needs significantly more than 3 taps.

author: Georgy.Moshkin
author_rank: Frequent Contributor
timestamp: October 31, 2024, 02:12:27 pm
content: I've read the article once more and conclude that "bit doubling" has nothing to do with two DACs. They have used two DACs just because they thought that it sounds better. And then they increased the sampling rate from 44kHz to 88kHz to move undesired "mixing byproducts" to a higher frequency away from audible range to simplify the filtering and increase attenuation. (Remember some switching regulators advertise that their switching frequency is out if audio range, very similar thing). "Bit doubling" is probably the wrong term they have used for "word doubling". If there is an array of 16bit words sent to the DAC, then we'll need to send each word twice if sampling rate is doubled. So, this "bit doubling" is most likely resampling of audio stream from 44kHz to 88kHz by doubling words and not bits.Regarding paralleling two DACs, maybe they've achieved some improvement as it provided more current to drive the next stage. No comparison is provided though. DAC performance papers on multiple MCUs recommend high speed high impedance input opamps at the dac outputs.To sum up, "bit doubling" and using two DACs are two unrelated features of the device that was described in the linked article.

author: gf
author_rank: Super Contributor
timestamp: November 01, 2024, 05:40:16 am
content: Assuming that the DAC outputs a staircase, why should there be a difference at all in the resulting signal, between 44kSa/s, and 88kSa/s achieved by doubling each 44k sample?

author: Georgy.Moshkin
author_rank: Frequent Contributor
timestamp: November 01, 2024, 11:25:19 am
content: Quote from: gf on November 01, 2024, 05:40:16 amnoise is caused by the digital interface and DAC internal circuit which performs a lot of shifting and switching during conversion.By increasing the sampling rate we can increase the switching noise frequency. But then our audio will be played at double rate resulting in funny high-pitched sound. To get normal playback rate back each sample is doubled (resampling without interpolation).

author: David Hess
author_rank: Super Contributor
timestamp: November 01, 2024, 01:53:00 pm
content: It is similar to oversampling but I think it produces a cosine response like a sampling process does. The Tektronix 7D20 digital storage oscilloscope had a mode which operated like this to double its sample rate.If it is just repetition of samples without averaging, then it does not do anything useful.Update: Didn't Burr-Brown have an application note discussing doubling sampling in this application? I will try to find it.Update Update: I did not find it. I might have been thinking of deglitching the DAC output which may involve doubling the sample clock.

author: wasedadoc
author_rank: Super Contributor
timestamp: November 01, 2024, 01:53:13 pm
content: Quote from: gf on October 31, 2024, 07:24:26 amNo it is not "basically equivalent to 2x oversampling withlinearinterpolation". It is just sample repetition. No interpolation at all.

author: gf
author_rank: Super Contributor
timestamp: November 01, 2024, 02:31:04 pm
content: Quote from: wasedadoc on November 01, 2024, 01:53:13 pmExample: Original sequence of samples: ... 1 3 7 9 ... (44kSa/s)... 1 1 3 3 7 7 9 9 ..... sequence with duplicated samples (88kSa/s)..... 1 1 3 3 7 7 9 9 ... same, but delayed (2nd DAC)..... 1 2 3 5 7 8 9 ..... average of the above two sequences=> same as if you 2x upsample the original sequence ... 1 3 7 9 ... with linear interpolationEDIT: I'm referring to moffy's speculation: "Repeating the same sample twice, but delayed 1/2 a sample period".

author: Haenk
author_rank: Super Contributor
timestamp: November 01, 2024, 03:24:11 pm
content: I tried to follow the description, but only read about paralleded 1541 DACs. And those 1541s don't come with 88kHz bandwidth either, so what's that doubling nonsense?. It all kind of looks like audiophool nonsense.Putting the DACs in parallel will help smoothing out coversion errors (and might reduce noise), this is a known concept for decades (all top tier HiFi manufacturers used it for their DACs back then), still unsure about this article though...

author: SteveThackery
author_rank: Super Contributor
timestamp: November 01, 2024, 04:52:41 pm
content: Quote from: PCB.Wiz on October 31, 2024, 02:07:33 amI don't think there is anything in that article about using two DACs. The text that says "another one" doesn't mean another DAC, rather it means "another type of oversampling" (as opposed to the usual type mentioned immediately beforehand).The article seems to advocate running the DAC at 88kHz and outputting each sample (NOT each bit!) twice.That is how I interpret it.FWIW, I think the output from the DAC using that method - 88kHz with each sample output twice - will be identical to running it at 44kHz and outputting each sample once. In the first case the staircase steps will be 2/88000 of a second apart; in the second case they will be 1/44000 of a second apart. In other words, identical.

author: gf
author_rank: Super Contributor
timestamp: November 01, 2024, 07:24:26 pm
content: Quote from: SteveThackery on November 01, 2024, 04:52:41 pmExactly this was my point, too:https://www.eevblog.com/forum/projects/parallel-dac-to-double-sample-possible/msg5697477/#msg5697477EDIT: I don't know - maybe this kind of doubling helps with noise. But I fail to see how it can eliminate the unwanted 24kHz tone in the spectrum when a 20kHz sine wave is sent to the DAC. The 24kHz tone is an image of 20kHz in the 2nd Nyquist zone, and it is simply an integral frequency component of the staircase signal coming out of the DAC. It is not noise. If the analog reconstruction filter is not capable enough to suppress this image, then IMO only upsampling (oversampling)with a proper interpolation filtercan do it. And by "proper" I mean a filter with, say, 100 taps or so, which is quite computationally intensive. I have no idea how one could achieve the same with simple "bit doubling".

author: moffy
author_rank: Super Contributor
timestamp: November 01, 2024, 09:44:32 pm
content: I have a program I wrote a while ago for DFT analysis, so I ran the doubled samples with single samples through the DFT program, and for a single 1kHz tone the doubled samples had a widening of the base, not as sharp a peak, similar to what you get when you have clock jitter on the ADC clock. It was subtle but noticeable.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 01, 2024, 10:05:28 pm
content: As @Georgy pointed out, the bit rate and paralleling are not related. If one of the two dacs was delayed by half a period, a 16bit register would be needed and I don't see any on the picture. Even so, it would be just linear interpolation that only decreases the level above 8khz.I can't figure out how they got the spectrum in pink. Did they use 88.2khz cd ?When I look on the net about dac, I found so many wrong understood and preached since 4 decades. Many believe that the step shape is responsible for digital sound. Many think that by stuffing zeros to increase the bit rate, the image frequencies get chased away. Others give hard time to equalize the sinc function to 40khz. The lowest image frequency is 24khz and it is a dirty thing needs to be cleaned off.The pre and post ringing in OS dacs is 22khz and it multiplies with the incoming signal to generate the difference frequency at transient, no problem with that except if you are the Batman or the Catwomen to hear 22khz.

author: gf
author_rank: Super Contributor
timestamp: November 02, 2024, 12:41:55 am
content: For a "perfect reconstruction" in the sense of the sampling theorem, the images have to be elimininated. In practice "eliminated" means attenuated sufficiently. But that's a solved problem with today's technology. It was difficult with early 1980ies technology, though.I think one misunderstand is to blame the player/DAC for something that already happens on the recording side. It is the recorder where the bandwidth has to be limited to 20kHz with a very steep cutoff towards 22kHz. That's where ringing in the step response isunavoidablyintroduced.The player does not necessarily need to introduce significant additional ringing when it reproduces the bandwidth-limited signal, but it cannot undo the filtering that already happened at the recorder.EDIT: If you believe that the step response shape is responsible for the sound then you need an anti-aliasing filter with a much softer roll-off at the recording side and a correspondingly higher sample rate. 44100 is not enough then, but you rather need, say, 192k or more.

author: radiolistener
author_rank: Super Contributor
timestamp: November 02, 2024, 04:26:35 am
content: interleave DAC is possible, but it needs precise phase shift between their clock and output, which is hard to achieve. With phase shift error you will get distortion harmonics.for audio frequency its more easy and cheaper just use single DAC with higher sample rate.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 02, 2024, 04:40:40 am
content: Both, Over sampled and NOS+filter have problems.With NOS, the high slope eliptic filter 11th-13th order used, filters out the 24khz and also the transient of 12khz-20khz. Below you see a 9th order eliptic with its response, usable only Below 11khz for short bursts. In music, above 8khz, the sounds are perceived in percussion instruments as short bursts.For interpolated image frequencies filtering, it is necessary to have minimum 15 samples from the future and 15 from the past each is weighted by sinc function. When a pulse of 1 sample is applied, the filter will betray one by one its weighing coefficients while coming in and going out of the interpolator. As the sinc function alternate the sign on every sample, the 44khz becomes 22khz multiplying frequency. If a 16khz burst enters the filter, it will generate 6khz new frequency by intermodulation and the sound of the percussion instrument will be different. We, listeners cannot know it, as we don't know how the original sound is but the artist realizes that this isn't the sound of his instrument.

author: SteveThackery
author_rank: Super Contributor
timestamp: November 02, 2024, 10:32:54 am
content: Quote from: Kokoriantz on November 01, 2024, 10:05:28 pmI'm certain there aren't two DACs. I recommend you read the OP's attached quote again (I admit it is badly worded and ambiguous).

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 02, 2024, 11:50:09 am
content: Look at the picture all at the end. There are two ICs with quality capacitors. The doubt is that, each IC needs 2x7 capacitors but here each have 1x7. Do they share it?

author: SteveThackery
author_rank: Super Contributor
timestamp: November 02, 2024, 01:55:32 pm
content: Quote from: Kokoriantz on November 02, 2024, 11:50:09 amWhat picture? Where is it?

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 02, 2024, 02:15:28 pm
content: Here.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 02, 2024, 02:17:36 pm
content: Here.

author: gf
author_rank: Super Contributor
timestamp: November 05, 2024, 07:56:16 pm
content: Quote from: Kokoriantz on November 02, 2024, 04:40:40 amA filter is a linear operation. Itdoes not createany new frequencies. Only frequencies already present in the input signal are passed through with frequency-dependent attenuation and phase shift.[ The images, whicharenewly created frequencies, are not introduced by filtering, but when the DAC emits a rectangular pulse with a width of one sample interval for each sample. The spectrum of the resulting staircase then contains the images in the 2nd, 3rd, ... Nyquist zone. ]QuoteBut then my conclusion would be that 20kHz bandwidth (44.1kSa/s) is simply not enough. But this cannot be solved on the player/DAC side alone. Already the recording must be done with higher bandwidth and sample rate, and the player, amplifier, and speakers must have a higher bandwidth, too. I also foundthis article, saying that recordings done with a high-bandwidth microphone at 192 kSa/s were prefered by listeners.Quote

author: Someone
author_rank: Super Contributor
timestamp: November 05, 2024, 09:49:46 pm
content: Quote from: Kokoriantz on November 01, 2024, 10:05:28 pmMust be, just doubling the output sample rate with a simple interpolator (zero order, linear, cubic) retains the images around the original sample rate. As you noted it would require a sophisticated interpolation filter to suppress that to the level they showed.

author: langwadt
author_rank: Super Contributor
timestamp: November 05, 2024, 10:05:09 pm
content: Quote from: Kokoriantz on November 02, 2024, 04:40:40 amdoes it come with a time machine?

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 05, 2024, 10:16:28 pm
content: Filter do create its own resonant frequency to dissipate accumulated energy. With digital filters, the signal enters and goes out in time domain as on bumpy road which makes it vibrate at Nyquist frequency. Apparently, much effort has been thrown to minimize this pre-post resonance but the linear phase shift type has always got the upper hand.I tried on simulator to deal with the image frequencies as double side band modulated by the ADC. So I demodulate synchronous way first the output of the DAC by multiplying with the Nyquist, 20khz becomes 2khz and remodulate it with Heartley single side band modulator. I do get rid of the images but as the modulator multiplies back, the DC component remained after the signal burst, generates decaying Nyquist frequency similar to post resonance. 2 week work for nothing.

author: gf
author_rank: Super Contributor
timestamp: November 05, 2024, 11:07:55 pm
content: Quote from: Kokoriantz on November 05, 2024, 10:16:28 pmIf you feed a (say) 10 kHz sine wave in, you get a 10 kHz sine wave out. Nothing else.I mean a continuous sine wave. Not an impulse or burst. These are not sine waves. They have a continuous spectum, containing an infinite number of frequencies.Even if a sinc impulse appears to oscillate at say 20kHz, it has a continuous, FLAT spectrum from 0...20kHz, and no frequencies beyond 20kHz. There is no dominant frequency peaking out in the Fourier spectrum.If you want less ringing/overshoot/preshoot you need a filter with a soft roll-off (like Gaussian or Bessel). Any filter with a sharp cut-off (analog or digital) suffers significantly from overshoot.

author: moffy
author_rank: Super Contributor
timestamp: November 06, 2024, 01:15:57 am
content: Quote from: gf on November 05, 2024, 07:56:16 pmThe big difference between the 44.1kHz and 192kHz sampled signals is the reconstruction filter. With the higher sample rate it is much easier to maintain good linear phase in the passband than with a necessarily much steeper filter for the lower sampling rate. I remember a story from Bob Pease about a fellow who designed a crossover for a speaker and complained that he had the spectrum quite flat, but Bob asked about the phase response, which in the end was where the problem lay.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 06, 2024, 04:26:27 am
content: I simulated an upsampling.The 16khz input gets by the interpolating filter at entrance and exit, multiplied by the Nyquist 22khz. The FFT graph shows 6khz and 38khz components.Then I upsample to 192khz. Now, all the undesired are frequencies go above audio range.I think the interpolated transient will still remain.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 06, 2024, 09:24:36 am
content: I came to the evidence that 20khz frequency is too high for 44.1khz sampled. The reasonable frequency is 16khz. After all any personal above 20 do not hear above. Commercial way is not good, specially online reviewers pretend to hear 0.5db difference at 20khz at an age of over 30, such proposition can only convince long time audiophile.I simulated the output of a DAC filtered by a lowpass 16khz 5th Bessel, bellow are the bursts of 0.5ms for 20khz and 16khz.Edit: Butterworth not Bessel.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 06, 2024, 11:38:04 am
content: I found a similar claim of paralleled dacs on DIYAUDIO.Post in thread 'Building the ultimate NOS DAC using TDA1541A'https://www.diyaudio.com/community/threads/building-the-ultimate-nos-dac-using-tda1541a.79452/post-914726Post in thread 'Building the ultimate NOS DAC using TDA1541A'https://www.diyaudio.com/community/threads/building-the-ultimate-nos-dac-using-tda1541a.79452/post-914899

author: gf
author_rank: Super Contributor
timestamp: November 06, 2024, 02:11:08 pm
content: Quote from: Kokoriantz on November 06, 2024, 04:26:27 amNot sure what you are doing - attached are my results from 4x upsampling from 44.1 to 176.4 kSa/s.No 6 kHz tone in the FFT spectrum, only the expected images at N*44100 +- 16000 Hz, where N is an integer.After applying the interpolation filter, the images are strongly attenuated.Code:% 16 kHz sine wave signal @ 44.1kSa/st = [0:44100-1]'/44100;x = sqrt(2)*sin(2*pi*16000*t);% plot spectrumfigure 1fs = 44100;f = [0:length(x)-1]'*fs/length(x)-fs/2;win = kaiser(length(x),18); win /= mean(win);plot(f/1000,fftshift(20*log10(abs(fft(x.*win)/length(x)))))grid onylim([ -140 0 ])% xlim([0 fs/2/1000])xlabel("kHz")title("16 kHz sine wave")% up-sample 4x by zero insertionx = 4 * [ x'; zeros(3,length(x)) ](:);% plot spectrumfigure 2fs = 44100*4;f = [0:length(x)-1]'*fs/length(x)-fs/2;win = kaiser(length(x),18); win /= mean(win);plot(f/1000,fftshift(20*log10(abs(fft(x.*win)/length(x)))))grid onylim([ -140 0 ])% xlim([0 fs/2/1000])xlabel("kHz")title("After 4x up-sampling (zero insertion)")% apply interpolation filter[n,Wn,beta,ftype] = kaiserord([20000 22000],[1 0],[0.0001 0.0001],fs);h = fir1(n,Wn,ftype,kaiser(n+1,beta),'noscale');x = filter(h,1,x);% plot spectrumfigure 3fs = 44100*4;f = [0:length(x)-1]'*fs/length(x)-fs/2;win = kaiser(length(x),18); win /= mean(win);plot(f/1000,fftshift(20*log10(abs(fft(x.*win)/length(x)))))grid onylim([ -140 0 ])% xlim([0 fs/2/1000])xlabel("kHz")title("After 4x up-sampling + interpolation filter")

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 06, 2024, 07:05:43 pm
content: What happens to a burst of 16 cycles of 16khz pass through this filter?

author: gf
author_rank: Super Contributor
timestamp: November 06, 2024, 09:47:44 pm
content: Quote from: Kokoriantz on November 06, 2024, 07:05:43 pmIt depends on the shape of the envelope whether its spectrum happens to be bandwidth-limited to 0...20 kHz or not.If it is bandwidth-limited, the above filter will not change it noticeably, because it only affects frequencies > 20 kHz.If the bandwidth happens to exceed 20 kHz, then you cannot put it "as is" on an audio CD, and the question of how the player's reconstruction filter would alter the signal becomes meaningless. The question then becomes, how does therecorderalter the signal to make it fit into the 0...20 kHz bandwidth?

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 06, 2024, 11:50:48 pm
content: The 16khz entering the bumpy Nyquist rate vibrating coefficients for interpolation will get modulated to be transformed into 6khz+38khz. The same modulation while going out.Transient instruments, as percussion, piano, strings, when reproduced, sound different than the original. We, the listeners, we don't realize it but the musician does that it sounds a different instrument.I found on you tube several examples how the pre and post ringing generate different sounds by switching linear phase to zero phase types.Here is an example.https://youtu.be/efKabAQQsPQ?si=8m-YEIMPdW0mmKsmOr else, on continuous note instruments, the digital filter is perfect.I want to remind that we give all the trouble to attenuate the image frequencies, to avoid intermodulating by the speaker's nonlinearty and generate audible frequencies. If the remedies are generating other modulating frequencies, is to replace the problem with another one.At this stage, I am convinced that 44.1khz is good for 16khz -3db.

author: moffy
author_rank: Super Contributor
timestamp: November 06, 2024, 11:58:08 pm
content: Quote from: gf on November 06, 2024, 09:47:44 pm@gf, sorry to intrude but could you plot the phase as well as the amplitude for the filter response? Thanks

author: gf
author_rank: Super Contributor
timestamp: November 07, 2024, 07:42:41 am
content: Quote from: moffy on November 06, 2024, 11:58:08 pmThe unwrapped phase response is a straight line. This is a linear phase FIR filter, having a constant group delay. What else would you use as interpolation filter for upsampling in the digital domain?Btw, in the digital domain you also had the opportunity to pre-compensate the inherent sin(x)/x frequency response roll-off of a traditional DAC, and/or pre-compensate the phase of the final analog reconstruction filter with a digital allpass filter.

author: gf
author_rank: Super Contributor
timestamp: November 07, 2024, 12:48:07 pm
content: Quote from: Kokoriantz on November 06, 2024, 11:50:48 pmYou must be doing something wrong. Filtering is convolution (in the time domain), not modulation. Convolution does not create new frequencies.QuoteThe spectrum of many instruments does extend into the ultrasound region. So yes, then itcannotbe reproduced exactly with 44.1kSa/s digital audio. I never objected that. I also pointed to an article which also claimed that 192 kSa/s was preferred by listeners, and the most prominent example in the article was a drum track, which definitively did contain frequencies >20kH.The question of whether the reproduction still sounds "good enough" when the bandwidth is cut to 20 kHz is not an EE question, but a psychophysical one.The EE question would be: Can a signal with a (say) 0...40 kHz occupied bandwidth be sampled at 44.1 KSa/s and then reconstructed almost perfectly? And the answer to this question is clearly "no".QuoteIt's a very interesting demonstration, but IMO not a good example for the desired use case, as it mainly shows the impact of group delay in the context of mixing the filtered channel with other channels. This does not apply to a CD player, where both channels (L+R) are always filtered equally and therefore have same group delays.QuoteAgain, a filter does not generate new frequencies.QuoteFor what purpose? For the player's reconstruction filter?In practice, you need of course some headroom below Nyquist. So 22.05 kHz are definitively not feasible. But I consider 20 kHz (-0.1dB) @41.1 kSa/s well feasible with today's technology and sufficient oversampling.Keep the following difference in mind:The antialiasing filter on the recordermust modifythe input signal if it happens to contain components >20 kHz. This modification can of course make an audible difference. And it may also make a difference,howit alters the signal in order to limit the bandwidth.But the player's reconstruction filter does not need to do that. It is never confronted with with useful frequencies >20 kHz. They are not present on the CD, because the recorder has already filtered them out. So all it has to do is pass through 0...20 kHz unaltered and suppress the unwanted images sufficiently. It does not matter how the reconstruction filter at the player would hypothetically alter the original analog signal when it exceeds 20 kHz. This signal is never seen by the player.

author: gf
author_rank: Super Contributor
timestamp: November 07, 2024, 01:01:33 pm
content: Quote from: gf on November 06, 2024, 09:47:44 pmAttached is an example of a burst with a "soft" envelope, where 99.9% of the power happens to reside in frequencies < 18 kHz. So I think this burst can be considered as properly bandwidth limited for sampling and reconstruction at 44.1 kSa/s.OTOH, with a rectangular envelope, this is no longer granted.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 07, 2024, 01:34:10 pm
content: Very interesting example.I will experiment by myself, so I will buy an ADC analog to I2S, SRC4190 sample rate converter board I2S in/out and I'll make a I2S tda1387 DAC.ADCs I read must use analog low pass filter for ant-aliasing. I don't think the 16khz can be so bandwidth limited. Bellow is that of pcm1802.It is highly probable that the ADC's digital filter generates pre and post ringing, then, OS will not harm any further.

author: gf
author_rank: Super Contributor
timestamp: November 07, 2024, 02:58:26 pm
content: Quote from: gf on November 07, 2024, 01:01:33 pmAttached is also the spectrum of a 16 cycle burst with rectangular envelope.The 99.9% quantile of spectral power is now at ~30 kHz.Problem with this kind of signals is that the actual bandwidth isinfinte. Using the 99.9% quantile of the spectral power as threshold between significant and negligible frequency components was an arbitrary choice of mine. I have no idea which amount is really audible.

author: gf
author_rank: Super Contributor
timestamp: November 07, 2024, 04:14:10 pm
content: Quote from: Kokoriantz on November 07, 2024, 01:34:10 pmJust a note: For a highly oversampling ADC like this one, the role of the analog antialiasing filter is not the same as for a non-oversamlping ADC. Here, the analog filter just provides anti-aliasing for the sampling at the significantly higher oversampling rate (64*fs or 128*fs), but it does not yet do the full filtering for the output sample rate fs. This is done by the decimation filter.

author: gf
author_rank: Super Contributor
timestamp: November 08, 2024, 01:53:08 pm
content: @Kokoriantz, yet another point where a misunderstanding might exist:Which bandwidth-limited continuous-time signal is represented by the discrete sequence of samples ...0 0 0 0 1 0 0 0 0... ?What do you think? What is the shape of the continuous-time signal between the points?

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 08, 2024, 03:59:04 pm
content: I learned to use a single pulse of 1 sample duration to generate a full spectrum signal instead of white noise.

author: gf
author_rank: Super Contributor
timestamp: November 08, 2024, 05:34:22 pm
content: Quote from: Kokoriantz on November 08, 2024, 03:59:04 pmThe samples ...0 0 0 0 1 0 0 0 0... represent a sinc impulse in continuous-time. See attached diagram, which also shows the sampling points. The diagram is of course truncated, since the horizontal extent of sinc is infinite. This impulse is theonlybaseband signal which isbandwidth-limitedto sample_rate/2, and which gives you discrete-time samples ...0 0 0 0 1 0 0 0 0... when you sample it. Its spectrum is continuous and perfectly flat in the 0...sample_rate/2 region, and zero beyond sample_rate/2 (brickwall).Your rectangular impulse is not bandwidth-limited. It does contain frequency contents beyond sample_rate/2.

author: BennoG
author_rank: Frequent Contributor
timestamp: November 11, 2024, 07:47:45 am
content: The signal path just released a video about thishttps://youtu.be/rVJwcZavr9U?si=_eK_RwFhKyHS3ejK&t=581Benno

author: Someone
author_rank: Super Contributor
timestamp: November 11, 2024, 10:58:05 am
content: Quote from: BennoG on November 11, 2024, 07:47:45 amNot really the same, that assumes you already have the higher sample rate signal available.

author: Kokoriantz
author_rank: Frequent Contributor
timestamp: November 11, 2024, 11:10:11 am
content: Here, the purpose is to double the sample/second conversion as you would put 2 x192khz in sequential parallel to get 384khz.