Maximum slew rate typically found in music/voice - Page 1
Forum section: electronics / general

author: HalFoster
author_rank: Frequent Contributor
timestamp: September 15, 2023, 05:57:40 pm
content: I came across a topic yesterday that had me thinking about the slew rates present in audio and the max/min required for processing equipment - specifically, how to fill in the blanks in the following statement:The highest slew rates are in the vicinity of <slew rate> and are frequently found in the waveforms generated by <instrument/voice/etc>.My first guess (admittedly a WAG) would be the initial hard strike of a cymbal or something along those lines. Does anyone have any specific numbers and/or references on this topic?Let me clarify that I am not referring to typical audio equipment's bandwidth - more like in an orchestra or concert in real life.Thanks,Hal

author: tom66
author_rank: Super Contributor
timestamp: September 15, 2023, 06:40:51 pm
content: Would it matter - if the human ear can't typically hear above 20kHz, the maximum slew rate is a function of frequency, since any energy above that frequency is going to be ignored by the low pass filter that is the human auditory system. There will be some audiophiles who claim that sampling above ~44kHz is necessary for one reason or another but AFAIK there's no scientific basis for those claims.I'd imagine it's possible that striking a cymbal could produce frequencies well above 20kHz and therefore slew rates well above what we would see from e.g. voice or wind instruments.

author: HalFoster
author_rank: Frequent Contributor
timestamp: September 15, 2023, 06:43:23 pm
content: Well, no in real life it wouldn't matter at all - in spite of what Audiophools might think. It was just curiosity on my part, nothing more.Hal

author: tggzzz
author_rank: Super Contributor
timestamp: September 15, 2023, 06:47:03 pm
content: The question is ill-formed. The slew rate is a function of frequency and amplitude.A 1kHz 100V signal has a slew rate 10 times higher than that of a 1kHz 10V signal.

author: BrianHG
author_rank: Super Contributor
timestamp: September 15, 2023, 07:00:06 pm
content: Quote from: tggzzz on September 15, 2023, 06:47:03 pmI was getting to that....What if you have a 1kv or 1 megavolt driver amp? What happens to the slew rate?

author: TimFox
author_rank: Super Contributor
timestamp: September 15, 2023, 08:08:45 pm
content: The reason why slew rate is important for audio is that when the waveform's slew rate approaches the maximum possible rate at any point in the circuit, the feedback around the amplifier can no longer correct for the distortion.Also, for a typical "op amp" circuit, the slew rate is determined by the maximum output current of the input differential stage flowing into the compensation capacitor across the second stagedV/dt=Imax/C,and the input stage's nonlinearity increases drastically as its output current approaches the maximum value.Similarly, feedback cannot correct for a deadband in the output stage ("crossover distortion").Without appropriate filtering before the input, an unwanted glitch or transient could cause a temporary excessive slew rate, which can interfere with the good stuff you want to hear until the amplifier recovers.

author: Benta
author_rank: Super Contributor
timestamp: September 15, 2023, 08:50:46 pm
content: Quote from: TimFox on September 15, 2023, 08:08:45 pmVery well phrased. It was a big audiophool issue in the 80s/90s carrying the buzzword "TIM" or "transient intermodulation distortion". The simple and universal fix is an RC filter on the input of the circuit as mentioned.

author: Smokey
author_rank: Super Contributor
timestamp: September 15, 2023, 08:58:29 pm
content: Musical instruments in the real world don't just create a single frequency. There are usually a complicated set of harmonics involved. The higher order harmonics can interact to generate lower audible frequencies sometimes. But like has been mentioned already, you can't hear above 20kHz so you don't actually care about anything higher than that in the absolute sense. If you were to digitize the resultant frequencies up to 20kHz, you could recreate what the real world thing "sounded like" to a human even if you don't have the full bandwidth representation of the original thing.So from that perspective it doesn't matter what happens above about 20kHz, even though some of that stuff is actually generating the frequencies you can hear.

author: spostma
author_rank: Regular Contributor
timestamp: September 15, 2023, 09:04:53 pm
content: I would simply write tool to analyze a number of typical professional audio recordings in WAV format.

author: mzzj
author_rank: Super Contributor
timestamp: September 15, 2023, 09:27:48 pm
content: https://www.renardson-audio.com/slew-tid.html

author: SeanB
author_rank: Super Contributor
timestamp: September 15, 2023, 09:32:27 pm
content: Pretty much most audio at some point or the other went through the venerable 741 opamp, or any of the clones, with the 1v/us slew rate it has. Works for low level audio, so pretty much should be a minimum, though better is not going to be easy to hear.

author: SiliconWizard
author_rank: Super Contributor
timestamp: September 15, 2023, 09:39:01 pm
content: Quote from: tggzzz on September 15, 2023, 06:47:03 pmObviously. To this I will add: a reasonable approximation for the minimum required slew rate would be to start with the classic relationship between bandwidth and rise time.For a 10%-90% rise time Tr, and a bandwidth BW, it is approx.:Tr = 0.35 / BWIf we take a typical "line" level for audio, amplitude in the order of 1V, so a p-p amplitude of 2V, we can approximate the 10%-90% section with 80% of the p-p amplitude, so 1.6V.Let's call the p-p amplitude App.The minimum required slew rate to accomodate a given bandwidth at the max amplitude given above would thus be (all values are in SI units, so App in V, BW in Hz and SR in V/s):SR = 0.8 * App / Tr = 0.8 * App * BW / 0.35 = 2.29 * App * BWWith a typical 20kHz BW for audio and the typical amplitude mentioned above, that yields: SR = 2.29 * 2 * 20e3 = 91600 V/s = 0.0916 V/µsAt +/-10V you have 10 times that, so ~ 0.916V/µs.Edit: Fixed a silly calculation mistake.We get a value within about 80% of the SR found classicaly from the derivative of a sine wave, which is perfectly normal since with the derivative method, you use the maximum of the signal's slope (the max of the derivative of the sine wave), while using the above rule of thumb, this is equivalent to using the slope of a line between the 10% and 90% points, considering sin(x) is relatively close to a straight line around the origin.(Finding the equivalence between the two formally takes a little while in calculation, but it is, within a factor.)As an extra exercise, you may consider what happens to a pure sine wave with the max frequency (= BW) if we limit the slew rate to the value calculated with the method above, rather than using the derivative at the origin.

author: wraper
author_rank: Unknown
timestamp: September 15, 2023, 09:48:54 pm
content: Quote from: tom66 on September 15, 2023, 06:40:51 pmSampling frequency above that is needed because 2x sampling frequency from Nyquist's theorem does not really work IRL because real low pass filters are not perfect and thus you want more margin. 44kHz sampling works good enough not because it works fine for 20kHz but because even though humans can hear high frequencies, they do it poorly. And neither actual sound contains much in that part of spectrum.

author: TimFox
author_rank: Super Contributor
timestamp: September 15, 2023, 10:10:58 pm
content: Quote from: SeanB on September 15, 2023, 09:32:27 pmThe 1 V/us slew rate of the 741 should be adequate for signal levels below, say, 1 V rms.Note that before you see the tell-tale straight line on the output waveform, the differential input stage has gone non-linear (large-signal behavior of differential pair).The decompensated 5534 op amp (Ccomp= 0) has a slew rate of 13 V/us, decreasing to 6 V/us with 22 pF, according to the 1994 Philips datasheet.(22 pF is needed for unity-gain stability, but 0 works for higher closed-loop gains.)

author: Nominal Animal
author_rank: Super Contributor
timestamp: September 15, 2023, 11:18:54 pm
content: Slew rate must be at least \$2 \pi f V_{pk}\$ for a sinusoidal signal of frequency \$f\$ and amplitude \$V_{pk}\$ to be reproduced perfectly. This is well documented for amplifiers.This is because the voltage of the signal with respect to time is$$V(t) = V_{pk} \sin \left( 2 \pi f t \right)$$and the slew rate is the magnitude of its derivative with respect to time,$$\left\lvert \frac{d V(t)}{d t} \right\rvert = \left\lvert 2 \pi f V_{pk} \cos\left( 2 \pi f t \right) \right\rvert$$which reaches the extrema when \$\cos\left(2 \pi f t \right) = \pm 1\$. Therefore, the maximum slew rate is \$2 \pi f V_{pk}\$.For \$f = 20000 \text{ Hz}\$ and \$V_{pk} = 1 \text{ V}\$, the slew rate must be at least \$125663.7 \text{ V/s} = 0.126 \text{ V/µs}\$.

author: gf
author_rank: Super Contributor
timestamp: September 15, 2023, 11:56:47 pm
content: Quote from: Nominal Animal on September 15, 2023, 11:18:54 pmAnd I also don't see why the relation tr = 0.35 / BW would apply here, which is rather the 10%...90% risetime of the step response of a Gaussian filter with a given -3dB bandwidth. This waveform is not a sine wave, but the integral of a Gaussian bell.

author: HalFoster
author_rank: Frequent Contributor
timestamp: September 16, 2023, 12:21:03 am
content: Quote from: BrianHG on September 15, 2023, 07:00:06 pmThat... is very true. Instead of slew rate the term should have been rate of change or rather DV/DT. I was thinking of how to get to a usable slew rate requirement and skipped a term. Mea culpa.

author: HalFoster
author_rank: Frequent Contributor
timestamp: September 16, 2023, 12:29:29 am
content: So a clearer question would be: In any average live musical or voice performance what would be the highest DV/DT of a natural waveform encountered and what instrument/voice would be the most likely to generate it. Naturally ignoring any random interference patterns or such.I was looking for an answer like "A <some type of cymbal> often generates a DV/DT of 200 V/uS on initially being struck." or somesuch.I really wasn't looking to get too complicated with this, I just thought that someone, somewhere, would have done this research and might have published it somewhere.Hal

author: tggzzz
author_rank: Super Contributor
timestamp: September 16, 2023, 12:53:26 am
content: Quote from: HalFoster on September 16, 2023, 12:21:03 amSince slew rateisdV/dt, you haven't changed anything - the question is still ill-formed.The necessary slew rate and dV/dt both depend on the waveform's amplitude

author: T3sl4co1l
author_rank: Super Contributor
timestamp: September 16, 2023, 03:44:52 am
content: Cymbals emit audio waves, not voltage; the question isn't meaningful. Through a transducer system (microphone and speakers), the voltage and therefore slew rate or dV/dt is proportional to the peak level, which is arbitrary: it might be 3V at line level, it might be down in the noise floor (<1µV?) if turned down/muted, or it might be thousands of volts on a powerful loudspeaker system (or some specialty types like electrostatic speakers).The question you meant to ask, is frequency response, and that is easily answered: the ear is sensitive to up to 20kHz or so, typically, so that's all that we care about. Sound sources can go higher, but there isn't much point in preserving that information for human consumption at least.And since such an oscillation can be full-scale, the slew rate still depends on line level or whatever, but several to low 10s of V/µs is typical for such equipment.Tim

author: TimFox
author_rank: Super Contributor
timestamp: September 16, 2023, 02:42:05 pm
content: A serious suggestion for the mathematically inclined to answer the original question:1. Assume the "music/voice" to be analyzed is on a conventional audio CD.2. The 16-bit digital waveform, sampled at 44 sec-1, goes through a reconstruction filter (something like sinx/x) to give the audio output.3. Determine the code that gives you a maximum 1 kHz sine wave and define its reconstructed output as the reference level.4. Put the full-scale digital square wave (zero and max on alternating samples) into the same reconstruction filter and compute the slew rate at the output.5. Determine the code that gives you a 20 kHz square wave and do the same.

author: Nominal Animal
author_rank: Super Contributor
timestamp: September 16, 2023, 06:05:51 pm
content: Let's expand what TimFox described.Conventional audio CD's contain metadata (including error correction), and uncompressed pulse-code modulated 16-bit stereo (two channel) data sampled at 44100 samples per second. The maximum slew rate is therefore 2¹⁶ = 65536 quantization steps per one 44100'th of a second, or 65536×44100 = 2890137600 steps/second ≃ 2.89 quantisation steps per nanosecond.If the audio CD data contains an alternating sample sequence (-32768, +32767, -32768, +32767, ...), per Nyquist-Shannon theorem, it should be reconstructed as a perfect 22050 Hz sine wave with maximum amplitude. This leads to the \$2 \pi f V_{pk}\$ (here, \$138544 \, V_{pk}\$ per second, or \$0.1385 \, V_{pk}\$ per microsecond) minimum slew rate required for both the incoming circuitry before the ADC and the output amplifier circuitry after the DAC.Does the DAC have more stringent slew rate requirements? Slewing from rail to rail in a single sample period is \$2 \, V_{pk}\$ in one 44100'ths of a second, or \$88200 \, V_{pk}\$ per second, which is less than the aforementioned reconstruction limit. This means that with a theoretically perfect brick-wall low-pass filter, the \$0.1385 \, V_{pk}\$ slew rate suffices, but this slew rate is \$1.5707\$ times (\$\pi/2\$) as fast as just slewing from rail to rail in a single sample period. This affects our choice of DACs, as just being able to slew from rail to rail in a single sample period is not sufficient; it needs to slew basically 1.5707 times rail-to-rail range, in a single sample period.If we did a Fourier analysis of the error spectrum at different (higher than necessary) DAC slew rates, we'd find a faster slew rate does push the error noise somewhat higher in the output spectrum, which makes it easier to filter out this particular error using analog circuits. Note that we're still assuming a brick-wall low-pass filter at 22050 Hz for reconstructing the highest-frequency components in the audio signal, though.For the ADC, there is no "slew rate" requirement as such, if we assume instantaneous sampling in time, or integration of the input signal for the duration of each sample. Existing ADCs differ, but their frequency response is known, and can be compensated using filters prior to the ADC. The main problem is that for perfect signal capture, we'd again need a brick wall low-pass filter at 22050 Hz, and AC coupling (i.e. rejecting the DC component). In practice, human hearing does not go below 10 Hz - 20 Hz, so signals below 10 Hz can be rejected also, giving us a brick-wall 10 Hz to 22050 Hz band-pass input filter requirement.Here we get into the realm that will forever weird out Audiophōles. If we replace the exact DAC with an oversampled dithering DAC, we can push all quantization noise to higher frequencies, so that we don't need a brick wall filter; a more realistic low-pass filter can reconstruct our original signal perfectly.Looking updelta-sigma modulationshows how these are done in practice, noting that the intermediate steps (within the modulation scheme) do require much higher clock rates than the sample rate used. It is also useful to note that single-bit delta-sigma modulation ispulse-density modulation, exactly.It is also useful to understand what kind of voltages are used to process and transfer audio signals. The most common standard isline level, which has \$V_{pk}\$ of 1.414 V at 0 dBu (decibels unloaded) in "consumer" devices, and 1.095 V at 0 dBu in "pro" devices, with signals clipped to somewhere between ±1.5V and ±2.0V. Thus, an initial assumption of \$V_{pk} \approx 1.4 \text{ V}\$ for consumer line-level audio that does not clip, is sensible.Combining all of the above, and a rough estimate of \$f = 20 \text{ kHz}\$ for the highest frequency we humans care about, we can say that usingline levels, the maximum slew rate needed is \$2 \pi f V_{pk} \approx 0.2 \text{ V/µs}\$.To understand the range in slew rates we should consider, let's consider superhuman hearing that can detect components up to \$f = 25 \text{ kHz}\$, and a fully clipping signal with say \$V_{pk} = 2 \text{ V}\$. The slew rate we get for this is \$\approx 0.31 \text{ V/µs}\$.Because we are talking about stereo audio, however, we do need to consider the one oddity abouthuman hearing: time discrimination. Humans can detect audio signal time separation down to 10 µs, which corresponds to 100 kHz. That is, because of the exact mechanism of human hearing (which is very much a spectrum analyzer, rather than time-domain sampling), humans can detect much smaller time delays than the maximum frequency they can hear. For engineers, the time-domain discrimination for changes in the spectrum detected in each ear is 10 µs. In turn, this does mean that even though 20 Hz .. 20 kHz bandwidth per ear suffices, we may need much higher bandwidth to properly represent 3D audio effects, because of the out extreme time-domain discrimination ability!Which also explains why 192 kHz and audio sampling, even when band-filtered to say 10 Hz ... 20 kHz, can produce superiorstereo/3D audio experience. Of course, that only really applies when the speaker configuration matches the microphone configuration, preferably a human head acoustic model with earlobes and all.It also turns out that most 3D effects do not rely on high time-domain discrimination at all, but more on spectrum shaping; basically, our earlobes and the shape of our head causes sound spectra to be filtered differently based on their direction, with the time-domain separation being just "fine tuning" on top of that. You can investigate and experiment on this further by looking into the open sourceOpenAL library.

author: nctnico
author_rank: Super Contributor
timestamp: September 16, 2023, 06:15:41 pm
content: Quote from: tom66 on September 15, 2023, 06:40:51 pmThe problem is that -like with any filter- you'll see / hear distortions happen at much lower frequencies. Brickwall filtering at 20kHz gives nasty effects as well. So yes, for excellent audio quality you'll need to sample at much higher frequencies. Some of the higher end audio amplifiers have bandwidths up to 200kHz or more in order to have the lowest phase shift in the audio band. In the end a CD is pretty bad where it comes to frequency and dynamic range.Quote from: Nominal Animal on September 16, 2023, 06:05:51 pmNo! You don't need a higher samplerate to phase shift a signal by a small amount. In case of audio you need a higher samplerate /bandwidth to preserve the frequency/phase response better. There is a slight difference there..

author: Nominal Animal
author_rank: Super Contributor
timestamp: September 16, 2023, 09:12:36 pm
content: Quote from: nctnico on September 16, 2023, 06:15:41 pmThat statement makes no sense.First, phase shift of a discretized signal is not exactly the same as delaying it, because the wave still starts at sampling window boundary, at best at a sample boundary, in the reconstituted signal. As it is the leading edge of a wave packet that is detected, the sinusoidal phases start at zero degrees, and shifting it to nonzero causes a leading error in the reconstituted signal, which can be audible as a high-frequency "click"-type noise.That is, it is true that for a band- or low-pass limited signal, the higher time resolution does correspond to greater phase resolution, mathematically. For a discretized signal, phase shift is problematic at the leading wavefront; and the leading wavefront is what is involved in the time discrimination here.Second, phase resolution in a discrete signal is solely determined by the sample rate \$f_s\$ and sinusoidal component frequency \$f\$: \$360° \, f / f_s\$. At the Nyquist-Shannon limit of half the sample rate, there are only two possible phases, 0° and 180°, because the discrete signal consists of alternating values. At quarter the sample rate, there are four possible phases, and so on. Phase resolution in a discrete signal is therefore always inversely proportional to frequency.Quote from: nctnico on September 16, 2023, 06:15:41 pmThat's notexactlyit, though: for mono audio, none of this matters.We only need a higher samplerate to preserve phase resolution of the difference of the signal to each ear at effective frequencies.Consider a stereo audio signal$$\begin{cases}L(t) = 0, & t \lt 0 \text{ or } t \gt 1 \\R(t) = 0, & t \lt \tau \text{ or } t \gt 1+\tau \\L(t) = (1 - t)^2 \sin\bigr(2 \pi f t\bigr), & 0 \le t \le 1 \\R(t) = (1 + \tau - t)^2 \sin\bigr(2 \pi f (t - \tau)\bigr), & \tau \le t \le 1+\tau \\\end{cases}$$where \$\tau\$ is the delay on arrival for the right channel, \$t\$ is time, and \$f\$ the fundamental frequency. For such a signal with sharp leading edge or peak, it is \$\tau\$ that has the 10 µs resolution, even though our hearing is limited to approximately \$20 \text{ Hz } \le f \le 20000 \text{ Hz}\$ or so (as if we had 50 µs \$t\$ sampling intervals).If we stored our audio as \$S(t) = \bigl(L(t)+R(t)\bigr)/2\$ and \$D(t) = \bigl(L(t) - R(t)\bigr)/2\$, only the latter (difference) would need the higher time resolution (sample rate, with both having the same bandwidth). To reconstitute, we'd need to use the higher time resolution for both channels, and apply \$L(t) = S(t) + D(t)\$ and \$R(t) = S(t) - D(t)\$.For a discrete audio signal, phase is an usefulmathematicaltool, though. For example, the peak sensitivity for human hearing is a bit below 4000 Hz; let's approximate it as 3675 Hz, or 1/12th of the CD audio sample rate of 44100 samples per second. This means that at that frequency, the phase resolution of CD audio is 360°/12 = 30°. If we produce two sinusoidal signals at 3675 Hz (each full wave taking about 272 µs), one for each ear, both starting at zero phase, but one minutely delayed, humans can generally discriminate at down to 10 µs difference. That corresponds to a 360° × 10µs / 272µs ≃ 13° phase difference at 3675 Hz. Thus, CD stereo audio does not have sufficient phase resolution at 3675 Hz to match human hearing.Yet, phase does not convey the correct underlying concept, and may lead to problems –– for example, the case above, initial samples in a phase-shifted wave packet leading edge, that tripped nctnico too.Humans are born with about 3500 hair cells in each ear, with each cell having a bundle of 50 to 100 hairs,stereocilia, which sense a specific frequency range. We can think of each ear as a spectrum analyzer with about 3500 channels (with each channel consisting of a few dozen frequency samplers within its range). Our brain can detect down to 10 µs delays between the activation of a pair of corresponding channels in each ear. Therefore, instead of pure sinusoidal signals, it is better to use the concept ofwave packets–– those with a steep leading edge or peak, for arrival time discrimination. In some sense, the "spectrum analyzer" does 100,000 spectrums per second; but this isn't exactly correct either, as most changes are detected at much, much lower rate.Overall, we're deep into human physiology andpsychoacoustic modelhere.

author: nctnico
author_rank: Super Contributor
timestamp: September 16, 2023, 11:43:48 pm
content: Quote from: Nominal Animal on September 16, 2023, 09:12:36 pmYou are forgetting the actual sample values / resolution. You can sample a sine wave at a random phase and still end up with a sine wave. If your DAC resolution is infinite, then your phase resolution will also be infinite. And this holds true up to (notat) the Nyquist-shannon limit.

author: CatalinaWOW
author_rank: Super Contributor
timestamp: September 17, 2023, 03:24:06 am
content: All of this discussion is wrapped around amplifier design and the effect of slew rate on distortion etc.I have had an interest in slew rate for a different reason. I transcribe old LPs (and other recordings) to .mp3 for convenience in listening and to avoid the problems controlling dirt, wear and other things in mechanical media. I am not an audiophoole, and in fact with my old ears don't need particularly hi-fi. But would like to remove the ticks, pops and clicks from the vinyl. The automatic click removal tool in Audacity actually does yeoman work in removing the highest amplitude clicks. But if it's parameters are adjusted to get rid of all of the audible clicks it has a noticeable negative impact to the sound, even to my pedestrian ears.You can manually go through the recording and eliminate the rest, but it is incredibly tedious and if the record is particularly noisy it can take many hours. Hence I started to write my own automatic click detector. Slew rate is an obvious metric, so many clicks have a slew rate easily within the recording equipment's capability, but out of family for the music on the record (and probably all music, since the stuff I am listening to has cymbal strikes and other similar sounds). But I never came up with a meaningful result. Partly because I found that the psychoacoustic stuff was too complex for me. You could have two clicks with identical time traces and one would be painfully audible and another inaudible even with careful listening depending on the sounds surrounding them in time. Many clicks that were objectionable had lower slew rates than "real" sounds.Long diatribe, with the only point being that slew rate may have other impacts than on the purity of reproduction.

author: mzzj
author_rank: Super Contributor
timestamp: September 17, 2023, 07:06:08 am
content: Quote from: HalFoster on September 16, 2023, 12:29:29 amSee my link above earlier.. " The prize for greatest slew rate found goes to a single peak from 'Year 3000' by Busted, which hit 150mV, equivalent to the slew rate of a full level sine wave about 10kHz, i.e. five times greater than Baxandall's figure, and equivalent to a level of 2.5V/µsec for a 100W power level."https://www.renardson-audio.com/slew-tid.html

author: David Hess
author_rank: Super Contributor
timestamp: September 17, 2023, 01:13:28 pm
content: Quote from: wraper on September 15, 2023, 09:48:54 pmBefore they used oversampling ADCs and DACs, 8th order elliptic filters sort of met the needed specifications, however anything modern uses oversampling ADCs and DACs and digital filtering so the antialiasing requirements are easily met now. The aliasing filter problem does not apply to modern 44.1kHz equipment.Quote from: gf on September 15, 2023, 11:56:47 pmIt still mostly applies, just like the 0.35 rule works for oscilloscopes with their bandwidth filter engaged which is decidedly not a Gaussian response. I tested this once with a Tektronix 7A22 set at various bandwidths and 0.35 matched exactly, as I suppose Tektronix intended it to.The fastest slew rate will be from a combination of frequencies so a step response, which has a comb of high frequencies, into the bandwidth filter will give a good approximation of the highest slew rate.

author: gf
author_rank: Super Contributor
timestamp: September 17, 2023, 01:52:36 pm
content: Quote from: David Hess on September 17, 2023, 01:13:28 pmYes, 0.35 seems to be the tr*BW product of traditional Tek scopes. An ideal Gaussian step response has actually 0.332 (which is of course pretty close).[ Edit: An ideal Gaussian filter is not realizable anyway. In the analog domain, approximations like "Gaussian to -6dB" seem to be common, and in the digital domain the infinite extent needs to be truncated/windowed, so that we end up with an approximation, too. ]My point was that the calculation 1.6*Apeak/tr = 1.6*Apeak/(0.35/f) leads to 4.57*f*Apeak, which underestimates the maximum slew rate of a sine wave Apeak*sin(2*pi*f*t), whose maximum slew rate is actually 2*pi*f*Apeak= 6.28*f*Apeak.

author: gf
author_rank: Super Contributor
timestamp: September 17, 2023, 03:04:33 pm
content: Quote from: Nominal Animal on September 16, 2023, 06:05:51 pmA DAC emits a pulse with a particular shape for each sample. In order to avoid non-linear distortion, the pulse shape must be the same for each sample (except for differently scaled amplitude, of course).However, the only pulse shape which enables "perfect reconstruction" is a Dirac delta pulse, i.e. in order to obtain an ideal, undistorted reconstruction (after applying the brickwall filter), the DAC would need to output a Dirac delta pulse for each sample (which requires infinite slew rate, of course).For any other pulse shape, the output of the DAC can be considered as if it were the above mentioned stream of delta pulses, convolved (in the continuous-time domain) with the DAC's pulse shape. The consequence is that any pulse shape other than Dirac delta leads to linear distortion, or in other words, it affects the frequency response. The convolution with the pulse shape acts as a filter.The linear interpolation between the samples you are considering above corresponds to a (hypothetical) DAC which emits a triangular pulse with FWHM=1/fs for each sample. This results in a sinc²-shaped lowpass frequency response, whose roll-off approaches 20*log10((sin(pi/2)/(pi/2))^2) = -7.84dB at fs/2 (or for for CD audio at 44.1kSa/s, it would be about -6.3dB at 20kHz).If you want to compensate this frequency roll-off in the digital domain before sending the data to the DAC, then you need additional dynamic range headroom for up to 2.47x higher amplitudes being sent to the DAC (implying the need for a 2.47x higher slew rate for the triangular pulses as well).BTW, I guess that many are not aware that even the commonly used rectangular pulse shape with a pulse width of 1/fs (a.k.a. first-order hold) introduces a sinc-shaped frequency response roll-off (in the amount of e.g. -3.1dB at 20kHz, for CD audio at 44.1kSa/s).

author: Karel
author_rank: Super Contributor
timestamp: September 17, 2023, 03:43:29 pm
content: Please read this article:A Method for Measuring Transient Intermodulation Distortion TIMhttps://www.ka-electronics.com/images/pdf/Leinonen_Otala_Curl_TIM_Measurement.pdf

author: newbrain
author_rank: Super Contributor
timestamp: September 19, 2023, 02:10:45 pm
content: Quote from: Nominal Animal on September 16, 2023, 09:12:36 pmNope. Sampling and reconstruction simply do not work that way.The theory says that any band limited signal can be reconstructed perfectly, and that include its phase; any reasonably implemented sampling system (CD) is able to reproduce the phase shift with inaudible accuracy.Please check this very old but still extremely valid video, it's absolutely worth watching the whole thing, but the relevant part starts at 20.56.https://youtu.be/UqiBJbREUgU?feature=shared&t=1256

author: Zero999
author_rank: Super Contributor
timestamp: September 19, 2023, 02:42:12 pm
content: Wow, there's a lot of interesting, yet irrelevant information in this thread.It's true, the original question isn't very well phrased, but it does have merit. An amplifier does not have to have its full power bandwidth extend all the way up to 20kHz. Look at a loudspeaker cabinet and note, the tweeter has around a tenth of the power rating, of the woofer. That should be enough to give you a clue. Most of the energy in music, can be found at lower frequencies. An LM741 run at +15/-15V will have more than sufficient slew rate to output an audio waveform with an amplitude of 20V peak-to-peak. There are other reasons not to use the 741.

author: TimFox
author_rank: Super Contributor
timestamp: September 19, 2023, 03:27:02 pm
content: Quote from: Zero999 on September 19, 2023, 02:42:12 pmThe 741 putting out 20 V pk-pk will push its slew rate limit.Minimum0.3 V/us rating corresponds to only 4.8 kHz at 20 V pk-pk.The actual slew rate limit is like clipping: you don't want to hit it, since the feedback around the circuit will fail and no good will come of it, and stuff happens as you approach it.As I pointed out above, the input differential stage in a typical op amp is reasonably linear at small amplitudes, but at larger current outputs (the maximum current into the compensation capacitor defines the slew rate) it is less linear and distortion increases.

author: Zero999
author_rank: Super Contributor
timestamp: September 19, 2023, 05:47:40 pm
content: Quote from: TimFox on September 19, 2023, 03:27:02 pmThat's more than good enough, considering the bandwidth of music. You'll only get 20V peak to peak, at low frequencies, up to a few hundreds of Hz, not at 4.8kHz.As far as not hitting the slew rate vs clipping is concerned. The distortion from limited slew rate doesn't sound as bad as clipping. The effect is more similar to when the sound has travelled through a wall, or ceiling than clipping.Having a wider bandwidth op-amp is good, but primarily because negative feedback is maintained at higher frequencies, which minimises the distortion due to the non-linearities in the output stage, rather than the slew rate.

author: magic
author_rank: Super Contributor
timestamp: September 19, 2023, 06:11:10 pm
content: Quote from: Zero999 on September 19, 2023, 05:47:40 pmQuote from: mzzj on September 17, 2023, 07:06:08 am100W into 8Ω being 40Vpp. Not so good for the old 741.Quote from: Zero999 on September 19, 2023, 05:47:40 pmCorrection of output stage distortion has nothing to do with slew rate, it only depends on loop gain at given frequency.

author: Zero999
author_rank: Super Contributor
timestamp: September 19, 2023, 09:40:04 pm
content: Quote from: magic on September 19, 2023, 06:11:10 pmThat's a moot point because the 741 is not a power amplifier and can't handle the current either.QuoteThat's true.

author: magic
author_rank: Super Contributor
timestamp: September 20, 2023, 05:46:18 am
content: But it can handle the voltage swing. Or at least a quarter of it (sorry, 100W is 80Vpp, not 40Vpp).Which means that over 0.6V/μs is needed to reproduce some real world CDs at 20Vpp full scale.Which also means that 741 likely wasn't used to produced these CDs, maybe for good reason

author: Zero999
author_rank: Super Contributor
timestamp: September 20, 2023, 09:39:28 am
content: Quote from: magic on September 20, 2023, 05:46:18 amI don't know where the figure of 0.3V/µs taken from, but the datasheet says 0.5µ/s.https://www.ti.com/lit/ds/symlink/ua741.pdfAlright I take the point, if can't quite be able to produce all real world CDs, just most of them. There are a couple of outliers.I'm not advocating the 741 audio, just point out that an audio amplifier doesn't have to have sufficient slew rate to output a 20kHz sine wave at the maximum output voltage.

author: TimFox
author_rank: Super Contributor
timestamp: September 20, 2023, 02:23:50 pm
content: 0.5 V/us istypicalThe onlyminimumvalue I could find is 0.3 V/us, which I used and noted asminimumNational LM741A datasheethttps://www.mit.edu/~6.301/LM741.pdfOf course, no one seriously recommends the 741 for modern audio, since there are reasonably-priced devices with roughly 10x the slew rate.TL071/2/4: 8 V/us min, 16 V/us typNE5532: 9 V/us typ

author: David Hess
author_rank: Super Contributor
timestamp: September 20, 2023, 03:23:39 pm
content: Quote from: TimFox on September 20, 2023, 02:23:50 pmSignetics designed a number of high slew rate bipolar parts at that time, but just a couple the NE series survived. Designers who were desperate used the LM318 but it relies on emitter degeneration so has high noise. National had the still popular LM833 at 5 min and 7 typ V/us ; I never understood how they achieved that and low noise but suspect the published schematics are simplified and do not show how the transconductance was reduced. Motorola had a bunch of high slew rate 741 replacements which are still available.I suspect the JFET parts were the most common 741 replacements for audio because their low input bias current and current noise allows higher impedance networks.

author: TimFox
author_rank: Super Contributor
timestamp: September 20, 2023, 04:20:55 pm
content: I went looking for the seminal article by W G Jung et al. in 1979 about slew-induced distortion, and found this in a free location:https://web.archive.org/web/20190303175101/http://pdfs.semanticscholar.org/cdbd/3ffcb276aabaf6cccd9b6bf067ee665c82d9.pdfBefore the 5534 series came out, there were some good high slew-rate products from Harris (later Intersil), that I believe used dielectric isolation or some such construction technique.e.g.https://pdf1.alldatasheet.com/datasheet-pdf/view/66679/INTERSIL/HA-2505.html

author: Nominal Animal
author_rank: Super Contributor
timestamp: September 20, 2023, 11:22:52 pm
content: Quote from: newbrain on September 19, 2023, 02:10:45 pm1. Shannon-Nyquist only applies exactly if you have infinite precision, i.e. exact discrete samples. Quantizing the samples limits the accuracy. Discretized quantized signal is limited in both bandwidth and phase resolution, and is limited in the resolution at which it can represent the wave packet arrival time.2. Human time-domain signal arrival time discrimination between each ear is not exactly a phase shift; the ear does not work like that. You do need to model it as the arrival of a wave packet to understand exactly how it occurs. Specifically, it requires the leading edge of the incoming wave packet to be sufficiently steep and sufficiently high maximum amplitude; in a Fourier transform, this is not a single-frequency event.You can easily simulate this, if you actually wanted to. Just generate a suitably shaped pulse as I described earlier at a higher sample rate. Generate several variants, delaying each by a one additional leading zero sample, then pad with lots of zeros to a suitable size. Decimate to CD audio sample rate and quantize to 16 bit samples, then resample back to original sample rate, and compare the initial and final waveforms.For example, using the aforementioned waveform using \$f = 3675\text{ Hz}\$ at 176400 samples per second (4x CD audio rate), with four variants; containing one, two, and three leading zero samples compared to the first (corresponding to 5.7µs, 11.3µs, and 17µs delays), using Python 3 and SciPy:Code:import scipyimport scipy.signalfrom math import sin, pifrom sys import stdout, stderr# Original discrete waveformwave = []for i in range(0, 12001):x = i / 12000.0wave.append( (1.0 - x)*(1.0 - x)*sin(500.0 * pi * x) )# Construct four samples, each delayed by one sampleorig1 = scipy.array((*36000) + wave + (*47999), dtype=scipy.float64)orig2 = scipy.array((*36001) + wave + (*47998), dtype=scipy.float64)orig3 = scipy.array((*36002) + wave + (*47997), dtype=scipy.float64)orig4 = scipy.array((*36003) + wave + (*47996), dtype=scipy.float64)# Resample by 1/4audio1 = scipy.signal.resample(orig1, 24000)audio2 = scipy.signal.resample(orig2, 24000)audio3 = scipy.signal.resample(orig3, 24000)audio4 = scipy.signal.resample(orig4, 24000)# Quantize to 16 bit precision (but no clipping)for i in range(0, 24000):audio1 = round(32768.0 * audio1) / 32768.0audio2 = round(32768.0 * audio2) / 32768.0audio3 = round(32768.0 * audio3) / 32768.0audio4 = round(32768.0 * audio4) / 32768.0audio1 = scipy.array(audio1, dtype=scipy.float64)audio2 = scipy.array(audio2, dtype=scipy.float64)audio3 = scipy.array(audio3, dtype=scipy.float64)audio4 = scipy.array(audio4, dtype=scipy.float64)# Resample back to original rateresult1 = scipy.signal.resample(audio1, 96000)result2 = scipy.signal.resample(audio2, 96000)result3 = scipy.signal.resample(audio3, 96000)result4 = scipy.signal.resample(audio4, 96000)for i in range(0, 96000):stdout.write("%5d %9.6f %9.6f %9.6f %9.6f %9.6f %9.6f %9.6f %9.6f\n" % (i, orig1, result1, orig2, result2, orig3, result3, orig4, result4))Note thatscipy.signal.resample()uses the Fourier method, i.e. it is the mathematically correct resampling method. Since it assumes the signal is periodic, we pad it with large number of zeroes before and after.Each output line contains the sample index, and original and final sample amplitudes (-1 to +1) for each of the four waveforms.Observe the first wavelength prior to the pulse, and two full 3675 Hz wavelengths at the beginning of the pulse (at sample 36001/36002/36003/36004), i.e. samples 36050 - 36099:Code:Original Delay +1 Delay +2 Delay +3Index Initial Final Initial Final Initial Final Initial Final35950 0.000000 -0.001446 0.000000 -0.001074 0.000000 -0.000089 0.000000 0.00091835951 0.000000 -0.000985 0.000000 -0.001470 0.000000 -0.001077 0.000000 -0.00007735952 0.000000 0.000092 0.000000 -0.001007 0.000000 -0.001465 0.000000 -0.00106835953 0.000000 0.001165 0.000000 0.000072 0.000000 -0.000994 0.000000 -0.00146335954 0.000000 0.001591 0.000000 0.001147 0.000000 0.000090 0.000000 -0.00100035955 0.000000 0.001085 0.000000 0.001573 0.000000 0.001165 0.000000 0.00008135956 0.000000 -0.000092 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-0.376217 -0.492363 -0.492399 -0.599564 -0.600150 -0.696540 -0.69735436094 -0.254780 -0.253986 -0.376775 -0.376245 -0.492363 -0.492386 -0.599564 -0.60014636095 -0.128468 -0.127896 -0.254780 -0.254004 -0.376775 -0.376233 -0.492363 -0.49237736096 -0.000000 0.000031 -0.128468 -0.127899 -0.254780 -0.253998 -0.376775 -0.37622136097 0.128425 0.127915 -0.000000 0.000037 -0.128468 -0.127901 -0.254780 -0.25398836098 0.254609 0.253873 0.128425 0.127922 -0.000000 0.000030 -0.128468 -0.12789936099 0.376395 0.375873 0.254609 0.253871 0.128425 0.127917 -0.000000 0.000022The 0° and 180° phases are correctly positioned, but the very first half wave – that which triggers the stereocilia, and humans being able to discriminate down to 10µs differences in the triggering time of same frequency bands between ears – is deformed. There is also quite a lot of quantization noise, and a spurious pre-pulse wave with significant amplitude (0.005, with quantization units of at 0.000030 or so). Essentially, the pulse leading edge is "smeared".If you examine the full output of the above Python script, you'll see that after the first wave or so, the output wave tracks very closely to the initial signal. The error or difference between initial and final signals are concentrated around the beginning of the wave packet, and this is what affects the signal arrival time difference discrimination in human hearing.

author: T3sl4co1l
author_rank: Super Contributor
timestamp: September 21, 2023, 01:14:15 am
content: Quote from: Nominal Animal on September 20, 2023, 11:22:52 pmSure, but suppose we have a quadrature sampled signal, i.e. 4 samples per cycle, so that Nyquist is happy enough. If the signal is a sine wave, then the samples are A sin(wt + ph), A sin(wt + ph + pi/2), etc. Depending on ph, either one pair of samples, or the other, is close to zero (ph ~ pi n / 2), and thus d(sample)/dph is large (~A) and the others are near +/-A and d(sample)/dph ~ 0; or both are similar (ph ~ pi (2n + 1) / 4) and all four slopes are comparable (d(sample)/dph ~ +/- A sqrt(2) / 2 in each slot). In either case, we have a total of two samples' worth at full gain, and thus somewhere on the order of 0.5-1 LSB of phase resolution, depending on how you want to count it (assuming A is ~full scale).I'm not sure what the complaint is here--?There might be complications due to mixing of frequencies, and practical signal chains, but those are issues of any practical signal chain, and if we had ideal brick wall filters, the above analysis can be approached arbitrarily (indeed it doesn't need to be quadrature, it could be, er, trisature so to speak, or anything strictly less than Nyquist condition if we don't mind that quantization noise becomes a larger fraction of phase noise for some samples; but that too doesn't really matter since we'd then be considering arbitrarily long windows of samples, as is necessary to resolve a signal arbitrarily close to Fs/2, and then phase is still perfectly recoverable in the presence of quantization noise; the N=4 case above is simply where it's evident within a ~minimum of samples). So in a real system with modest oversampling and whatever quantization level dictating overall bandwidth, things come through just fine.Mind, I'm not interested in getting into psychoacoustics here, and probably neither was the OP. At the very least, such a topic shouldn't be taken as a given, and must be explained and justified.For what little I know of it, it's my understanding the ear is particularly well tuned to correlations of specific kinds, like impulsive sounds (edges, short wavelets of many frequencies), and to -- given some assumptions about the expected spectrum of familiar sounds -- the comb-filtering effect of delays between ears, diffraction around the head, and reflection/diffraction within the (external) ear shape itself; and by extension, the effect of ambient (room or object) reflections. Which, I believe, the ears give a little front-back discrimination for applicable sounds, obviously mostly left-right, and very little up-down; which makes sense for a species expected to have evolved on a grassland to forest edge environment.Tim

author: gf
author_rank: Super Contributor
timestamp: September 21, 2023, 08:05:31 am
content: Quote from: Nominal Animal on September 20, 2023, 11:22:52 pmI think the main problem is that the the original signal violates the sampling theorem.It does contain frequency contents >= 12kHz.Then the reconstructed signal is not supposed to be equal to the original one.However, when I compare the tabulated Delay+1 and Delay+2 values (with one sample offset), the differences seem to be in the order of magnitude expected with 16-bit quantization.QuoteI think this is mainly the difference between the original signal and the band-limited original signal (filtered with a zero-phase brickwall filter). If the original signal was already band-limited to < 12kHz per se, then I think you would not see such a big difference.

author: Nominal Animal
author_rank: Super Contributor
timestamp: September 21, 2023, 09:47:28 pm
content: Quote from: gf on September 21, 2023, 08:05:31 amI don't see any way of low-pass filtering the original signal that would not "smear" the attack phase of the wave packet.Another way to put it, I guess, is to say that while humans cannot typically sense sounds above 20 kHz or so, our time discrimination (arrival time difference between each ear) is such that components above 20 kHz are involved, with the time discrimination ability measured to be on the order of 10 µs.The reason I keep talking aboutwave packetsis because it much better matches the physical operation of the stereocilia and hair cells. It is the time difference between activation of the corresponding hair cells in the ears that has the 10 µs time discrimination ability.Now, how would you model that as a simple frequency response? I don't know exactly how. It is much easier to do using wave packet model, where if you do an infinite-width Fourier analysis of the waveform, you indeed do have frequencies above the range at which humans can hear a sinusoidal continuous signal. It should not be a surprise that a pure frequency domain representation cannot capture a mixed frequency-domain (stereocilia and hair cells, that respond to a narrow band of frequencies) and time domain (band activation time difference in the auditory brainstem) sensors.The uncontested facts are that while a typical human ear can sense acoustic waves within 20 Hz to 20 kHz or so, and differences as small as 10 µs between initial arrival time in each ear. The peak sensitivity is a bit below 4000 Hz, which means that at that frequency, the time discrimination is on the order of one 25th of the wavelength. Another sensitivity peak is around 1000 Hz, where the time discrimination is on the order of one hundredth of the wavelength.The question seems to be whether CD audio can capture that or not.My understanding is that in optimal conditions (i.e., volume set depending on background noise levels, background noise having no sharp spectral peaks) mono CD audio does convey basically all perceivable information to a typical human (if we exclude dynamic range changes, like an orchestra playing some parts very loud, and some parts very quietly; human hearing, effectively, has an automatic gain control relative to the surrounding noise floor, if you will). For stereo, it does not convey all the information needed for the human auditory senses to perceive the exact direction of specific types of sounds, "smearing" their directionality. The reason for this belief is the physical structure of the audio sensor, having stereocilia waving in a fluid that in groups trigger nerve cells, with groupwise arrival time differences from each ear processed in the auditory brainstem (via the cochlear nucleus). It is neither purely time-domain, nor purely frequency-domain apparatus, but a mix.I've probably messed up my explanations and my math above –– Idomake errors often ––, but the above paragraph is basically what I've been trying to convey.The end result, in my opinion, is that 24-bit quantization allows a larger dynamic range, and 192 kHz sample ratebettercaptures the stereophonic information human hearing can detect, and is not just audiophoolery or done because it makes filtering and quantization noise shaping easier.

author: CatalinaWOW
author_rank: Super Contributor
timestamp: September 21, 2023, 11:52:11 pm
content: Or saying the same thing as Nominal (I think), in somewhat different words. Our brain is not just a simple RMS meter connected to a microphone and front end amplifier. There are at least two types of sensors and literally hundreds or thousands of each type. Non-linear responses are clearly evident and there is evidence of temporal variation on time scales ranging from under a second to decades. Our experience of sound is the brain combining and interpreting all of those different inputs.Reconstructing the pressure variations in the ear canal produced by the original sound source is the goal of all audio systems, but a simple measure of performance measured in the 20-20k band at some amplitude is a necessary but not necessarily sufficient criteria.Unfortunately at our current level of understanding we can't really define a better measure, and so how it sounds is all we can do. And that is subjective and also subject to social pressures. For most of us CD quality sound is more than good enough, and for a somewhat smaller subset of us streaming quality digital audio is just fine. But I believe there are some who really do hear defects in CD quality sound, and many more who have convinced themselves they can for a variety of reasons.

author: NiHaoMike
author_rank: Super Contributor
timestamp: September 22, 2023, 04:42:24 am
content: Those who use equalization/ "room correction" will want more than 16 bits playback capability, since by the very nature it operates, it will reduce some frequencies so that they'll end up with less than 16 bits of resolution if you start with just 16 bits. Starting with 24 bits (or even 20 "real bits"), there will be room for some pretty aggressive adjustments and still have at least 16 bits of resolution left over for every frequency in the range.

author: metebalci
author_rank: Frequent Contributor
timestamp: September 22, 2023, 06:11:09 am
content: When reading the posts, I was thinking there has to be some amount of studies done in this topic (I have read some amount of psycho/technical acoustics and auditory neuroscience papers many years ago). Here is the abstract of one I quickly found from 2007:"Misalignment in timing between drivers in a speaker system and temporal smearing of signals in components and cables have long been alleged to cause degradation of fidelity in audio reproduction. It has also been noted that listeners prefer higher sampling rates (e.g., 96 kHz) than the 44.1 kHz of the digital compact disk, even though the 22 kHz Nyquist frequency of the latter already exceeds the nominal single-tone high-frequency hearing limit fmax∼18 kHz. These qualitative and anecdotal observations point to the possibility that human hearing may be sensitive to temporal errors, τ, that are shorter than the reciprocal of the limiting angular frequency  ~ 9us, thus necessitating bandwidths in audio equipment that are much higher than fmax in order to preserve fidelity. The blind trials of the present work provide quantitative proof of this by assessing the discernability of time misalignment between signals from spatially displaced speakers. The experiment found a displacement threshold of d≈2 mm corresponding to a delay discrimination of τ≈6 μs."http://boson.physics.sc.edu/~kunchur/papers/Audibility-of-time-misalignment-of-acoustic-signals---Kunchur.pdfSo it looks like 20 kHz is obviously not enough to replicate natural listening experience.I dont think enough is known about how this works (in terms of neural coding, early processing in the brain etc. at least it was the case ~10y ago) so the only way is doing experiments and this gives a starting point. It is not much ungrounded to move to 96 or 192 kHz.

author: tggzzz
author_rank: Super Contributor
timestamp: September 22, 2023, 07:16:22 am
content: The concept of "degregation of fidelity" misses the elephant in the room.Consider recording an orchestra in a concert hall. Where exactly should the microphones be placed? Conductor's podium? First row of the audience? Centre, right left? Centre of the audience? With or without a complete audience?The concept of a single place for the microphones is, of course, too simplistic. But it does highlight the point that therecannot bea single "correct" sound - it is all a choice made by the recording engineer.Back when I had ears and CDs were new, I and a friend did A-B comparisons between a CD and vinyl. His setup was good, but not audiophool. We could tell a difference, but we could not tell which was which nor which was better.

author: metebalci
author_rank: Frequent Contributor
timestamp: September 22, 2023, 07:25:22 am
content: Quote from: tggzzz on September 22, 2023, 07:16:22 amWhat about binaural recording ?

author: tggzzz
author_rank: Super Contributor
timestamp: September 22, 2023, 08:37:52 am
content: Quote from: metebalci on September 22, 2023, 07:25:22 amQuestion in the second paragraph: whereexactlywould you put the dummy head?

author: metebalci
author_rank: Frequent Contributor
timestamp: September 22, 2023, 11:05:57 am
content: Quote from: tggzzz on September 22, 2023, 08:37:52 amWhere I would be sitting. I understand the difficulties, I was just trying to say for some situations I think the answer is simple but particularly for large environments yes I agree. I am actually amazed how good they are recording and mixing.

author: tggzzz
author_rank: Super Contributor
timestamp: September 22, 2023, 11:53:58 am
content: Quote from: metebalci on September 22, 2023, 11:05:57 amAnd where would you be sitting? Why there and why not elsewhere?Depending on where you are sitting, you will hear the music differently. Which "different" is "right", and why are the other positions "not right"?

author: metebalci
author_rank: Frequent Contributor
timestamp: September 22, 2023, 02:29:59 pm
content: Quote from: tggzzz on September 22, 2023, 11:53:58 amI am not talking about subjective difference (recording each instrument individually and creating a virtual mix for each location is another topic), naturally each seat in a concert hall can be different. I hardly remember the lectures and there are probably changes but a hall is probably optimized acoustically for certain locations, at least these are the points measurements are probably made. I guess in any environment even on a street, one would want to face the sound source to not have a major imbalance and want to be in a proper distance depending on the volume. I dont think finding a reasonably proper location is a question here. Just putting a dummy head there does not resolve all issues as far as I know, that is a bigger problem, if not binaural records would be much more common. However, binaural records are much better at stereo imaging at least to my experience, which is not surprising since all others are to some extent is mixed for an artificial space (since as you said for a large space there are a lot of mono/stereo different types of microphones). As far as I remember, it is also not only hearing from two ears, but also movements of head improves the subjective experience, that is partially also why spatial audio wants head tracking (eg airpods).

author: CatalinaWOW
author_rank: Super Contributor
timestamp: September 22, 2023, 02:35:17 pm
content: I think the where you are sitting is being either over thought, or strong evidence that more reproductive precision is required. One on end of the argument, any seat is fine as long as you would be happy with that seat in a live performance. On the other hand, if you want recordings from a bunch of different locations so you can make a seat selection for your next live performance the answer gets complicated.

author: TimFox
author_rank: Super Contributor
timestamp: September 22, 2023, 02:35:40 pm
content: Aficionados of different concert halls know where the best seats are for serious listening.In some opera houses, the music lovers know about cheap seats with lousy sight lines but good acoustics.I was told about these seats at the Bayerische Staatsoper, but I only got there once (an excellent seat 1/3 back in the orchestra, dead center left-right).Flying back from Japan on a long flight, I had an interesting chat with a professional acoustics engineer who had carefully surveyed several European halls for this purpose.Unfortunately, we could not compare notes about my home city (Chicago), where I prefer the front of the first balcony at Orchestra Hall.

author: CatalinaWOW
author_rank: Super Contributor
timestamp: September 22, 2023, 02:43:09 pm
content: While those aficionado approved seats are undoubtedly good I would be willing to bet that folks with preferences for subsections of the music, brass or woodwinds as examples, have preferences that are subsets of these, or even outside of them.

author: nctnico
author_rank: Super Contributor
timestamp: September 22, 2023, 03:10:43 pm
content: Quote from: CatalinaWOW on September 22, 2023, 02:35:17 pmMost halls will have reflections that cause peaks & valleys for certain frequencies. This makes recording using an artificial head almost a fruitless exersize. One of my mentors who previously had worked at the R&D department for an electronic organ manufacturer, had an interesting story about this subject. At some point they wanted to build a digital church organ. So they took a good quality recorder and made recordings of a real church organ. They reproduced the waveforms in great detail but the organ they build sounded like crap. After some head scratching they decided to go recording again but this time at different positions. And with those recordings, they managed to design an electronic church organ that sounded like an actual church organ.

author: tggzzz
author_rank: Super Contributor
timestamp: September 22, 2023, 03:48:00 pm
content: Quote from: CatalinaWOW on September 22, 2023, 02:35:17 pmIt is an accuracy vs precision issue.I am noting that accuracy in a concert recording is an improper thought, but precision could be attained. Audiophools witter on about accuracy, whereas you have sensibly noted that precision could be used to choose a seat. (Or conversely that the sound engineer chooses the seat for you).Yes, I am well aware that sometimes test equipment's precision is more important than its accuracy. One company I worked at produced an attenuation meter with a 0.001dB resolution but only a 0.1dB accuracy.

author: TimFox
author_rank: Super Contributor
timestamp: September 22, 2023, 03:59:45 pm
content: Quote from: CatalinaWOW on September 22, 2023, 02:43:09 pmOf course, brass aficionados have their own preferences.Once, by accident, I ended up at an outdoor opera production sitting immediately in front of the viola section, which gave me a different impression of Mozart.One thing that may distinguish "audiophools" (a common insult hurled about in this platform) from genuine music and audio lovers is if they attend actual live concerts of acoustic music, or not.

author: Nominal Animal
author_rank: Super Contributor
timestamp: September 23, 2023, 08:51:07 am
content: Quote from: tggzzz on September 22, 2023, 07:16:22 amNote that I used it specifically in the context of conveying information. I do not actually believe that perfect reproduction is always the best choice.When one needs to convey specific information, simplification and detail control is key. This is why emergency vehicles have that undulating sound (trivially recognized, as it does not occur in nature) interspersed with shorter noise or tones (to help humans perceive the direction the sound is coming from; it is very difficult to do if the sound is continuous), and why we use diagrams instead of photographs to convey visual data series.Quote from: tggzzz on September 22, 2023, 07:16:22 amFurthermore, should the recording even reproduce a specific listening point, or a physically possible listening position?An interesting similar choice occurs in 3D rendering and stereoscopic visuals. If you choose the view frustrum to match the typical viewing distance, the rendered image acts like a window to the rendered scene. However, you can increase the field of view, which lets the human view more of the scene at once than they could in real life; and for stereoscopic images, increasing the interocular distance enhances the depth perception.One of the related tricks we have all seens in horror movies is when the camera zooms in while moving away, or zooms out while moving closer in, causing a field of view and depth scaling change that humansrelate to. (Something similar happens to ourperceptionin specific situations (danger!) due to hormonal surges, even though our eyes cannot zoom in or out per se.)Quote from: TimFox on September 22, 2023, 03:59:45 pmI use the term for those who believe their audio equipmentcan haveneeds certainje ne sais quoithat can befeltbut notmeasured.To me, live concerts are more of a social event – I'm not just listening to the music, but participating in it as part of the audience –; but I tend to listen to certain pieces of classical music simply because of the effect it has on my mind. In live concerts, there is always an interaction between the performers and the audience, which is a big part. Even with acoustic music and no vocals, the interplay between the audience and the musicians and the venue (especially large halls intended for music) is a huge part of the experience for me. Thus, I do not believe live music vs recordings is related to audiophõlery.

author: tggzzz
author_rank: Super Contributor
timestamp: September 23, 2023, 09:10:43 am
content: (uncontentious points snipped)Quote from: Nominal Animal on September 23, 2023, 08:51:07 amHaving started taking stereoscopic pictures in 1984, and possessing many interesting vintagestereoscopic glass slides, I'm sensitive to the modern use of "3D"There's a complementary point to interocular distance that everybody encounters when watching a stereoscopic slideshow or movie: there is a sweet spot. Obviously it is better to sit on the projection axis rather than to the right or left. Less obviously, sitting closer to or further from the screen causes the depth axis magnification to change. The cause is simply the angle subtended at the eyes by the homologous points in each image: larger angle means increased depth magnification.No doubt some like "superstereo depth"; I don't!

author: TimFox
author_rank: Super Contributor
timestamp: September 23, 2023, 04:21:56 pm
content: "It don't mean a thing if it doesn't have that certainje ne sais quoi" -- Peter SchickeleI just got back from hearing Riccardo Muti conduct the Chicago Symphony Orchestra in Stravinsky'sFirebirdsuite in front of a full audience.The social aspects of the event were not primary.

author: strawberry
author_rank: Super Contributor
timestamp: September 23, 2023, 05:20:23 pm
content: slewrate is bandwidthlimit for 20kHz is for 1 harmonic signal , I thinksignal will be delayed when moveing up in frequency spectrum20kHz low pass filter will subtract high harmonics from signal20kHz vs 100kHz sound a bit cleanerif 20kHz low pass is ~3..6db/octave then there might not be audible differencethere is reason why MP3 sounds ok on cellphone but not on stereo system

author: TimFox
author_rank: Super Contributor
timestamp: September 23, 2023, 05:47:03 pm
content: Slew rate is relevant to distortion, which may increase with signal level non-linearly.Harmonic distortion and intermodulation distortion are different ways to quantitatively measure the same non-linear behavior.3rd-order intermodulation products will not be affected by a low pass filter, since they are close to the two test-tone frequencies.Bandwidth is a measure of a linear system.

author: tggzzz
author_rank: Super Contributor
timestamp: September 24, 2023, 09:05:42 pm
content: Quote from: strawberry on September 23, 2023, 05:20:23 pmSlewrate is bandwidthplus amplitude.There is a reason that bandwidths are often divided into two distinct categories: "small-signal bandwidth" and "full-scale bandwidth". The former is where slew rate limiting doesn't occur.

author: TimFox
author_rank: Super Contributor
timestamp: September 24, 2023, 09:12:06 pm
content: The latter is often called “full-power” bandwidth on data sheets.

author: tggzzz
author_rank: Super Contributor
timestamp: September 24, 2023, 10:49:37 pm
content: Quote from: TimFox on September 24, 2023, 09:12:06 pmSigh. Quite right.Brain failure on my part; shouldn't post when too tired.