Is this circuit possible? (reverse audio compressor...sort of) - Page 1
Forum section: electronics / beginners

author: jfharper
author_rank: Contributor
timestamp: March 14, 2025, 02:08:04 pm
content: When recording electric guitar via DI, the captured track shows a range of dynamic peaks. I would like to boost just the smaller peaks without boosting the large peaks causing them to clip...BUT, I don't want the entire signal squashed like how a compressor would handle it (i.e. increasing the signal to noise ratio) and changing the tone a bit which shows up later when I reamp and mic record...I'd like to try and match the tone I get when I old school plug directly into the tube amp and mic record while using headphones. Is there like a driver circuit or something I can build and place in between my guitar and the audio interface to do this? Sort of how low/high pass filters roll off frequencies, but this circuit would not be frequency dependent, more like small/low volume or amplitude dependent...but how would the circuit know which amplitude to boost and which not to boost. Just thought I'd ask to see if there was an idea that would point me in a direction where I can do more research. Thanks.EDIT: I've seen how offsetting specific value zener diodes in amp circuit are used to clip...this circuit would do the opposite, possibly detect the amplitude and boost if it were under a certain threshold, but if over the threshold...leave it be.

author: themadhippy
author_rank: Super Contributor
timestamp: March 14, 2025, 02:19:38 pm
content: Quotespeaker emulator?

author: timeandfrequency
author_rank: Frequent Contributor
timestamp: March 14, 2025, 03:07:51 pm
content: Quote from: themadhippy on March 14, 2025, 02:19:38 pmA speaker emulator is not a verycommon feature.But digging the Internet gives somefaint results.Don't know if it's easy to procure those old gears.Aforum entryon that topic.

author: themadhippy
author_rank: Super Contributor
timestamp: March 14, 2025, 03:41:42 pm
content: QuoteDo wot?Theirs plenty of commercial offerings ,wish i could recall the one we used,all i remember it was in a red box,but it did what it said on the tin,made the guitar sound like it was plugged into an amp without the annoying loudness on stage.

author: timeandfrequency
author_rank: Frequent Contributor
timestamp: March 14, 2025, 04:51:39 pm
content: Maybe that jfharper would be interested in the brand and model names of all these commercial offerings

author: themadhippy
author_rank: Super Contributor
timestamp: March 14, 2025, 06:12:09 pm
content: we call em emulators ,others seem to call em simulators,same samehttps://www.bax-shop.co.uk/guitar-amplifiers/speaker-simulator

author: SteveThackery
author_rank: Super Contributor
timestamp: March 14, 2025, 11:50:11 pm
content: This is confusing. The OP apparently wants two separate things: firstly, to boost some of the peaks (but not like a compressor does it); secondly, to achieve a particular tonal quality reminiscent of a tube amp.At least I think that is what he is asking. Am I correct,@jfharper? If so, it might be useful to discuss them separately.

author: Analog Kid
author_rank: Super Contributor
timestamp: March 15, 2025, 12:08:58 am
content: Sounds simpler than that to me: just a partial compressor that would boost all except the highest peaks, the ones that would cause clipping. Probably adjustable to control how much compression is done.Sounds do-able, though I have little idea how ...

author: dkfan9
author_rank: Newbie
timestamp: March 15, 2025, 12:56:14 am
content: This is a basic idea. Forum does not allow .qsch attachments but I can send you the sim if you have Qspice (or I can change the extension for upload, but I'm not sure on forum decorum regarding that practice). Circuit has a gain of 2 for inputs less than |2V| and gain of 1 above |2V|--the op-amp power is turned off via the comparator-->AND gate-->MOSFET chain. The 100n cap suppresses the power off spike. It's a proof of concept, not something you should actually try to implement as shown--you'd destroy the AND gate and MOSFETs.

author: jfharper
author_rank: Contributor
timestamp: March 15, 2025, 01:46:53 pm
content: Quote from: SteveThackery on March 14, 2025, 11:50:11 pmclose...I already have the tube amp tone I want from, well, the tube amp.When I DI record I get a track with various sized peaks that represent the dynamics. When I reamp, this track is played from the pc/audio interface to an opamp signal booster I made then to a reamp box, then to the amp/speaker for the tone. What get's lost a bit is the dynamic human or real to life dynamic analog tone that I get when I plug straight into the amp (i.e. guitar to amp to mic recording on speaker)...it's subtle, but it's an old school feel that I feel needs improvement. It's difficult for me to record old school with headphones on like this because I cannot react to the rest of the band playing in the room. My recordings come out way better if I hear without headphones, while I play the piece. That advancements to DI record and reamp later are amazing, but if the tone can be captured, IMO, better, I think this process would be there...for me at least.Quote from: Analog Kid on March 15, 2025, 12:08:58 am^this...and I would say since I don't want to compress anything, it more like boost only low peaks and yes adjustable would be perfect...that way I could subjectively choose if I "normalized" too far and back it off a bit to achieve the old school tone I'm after. It's really like an early 70s tone made from late 60s tube amps...it's a very 70s tone/feel. If the adjustments could be exponential or logarithmic (not sure if that's correct) but like the smallest peaks get booster more, while a threashold for the larger peaks would keep them untouched.

author: jfharper
author_rank: Contributor
timestamp: March 15, 2025, 02:00:53 pm
content: Quote from: themadhippy on March 14, 2025, 02:19:38 pmQuote from: timeandfrequency on March 14, 2025, 03:07:51 pmQuote from: themadhippy on March 14, 2025, 03:41:42 pmQuote from: themadhippy on March 14, 2025, 06:12:09 pmMy idea is different from an emulator I think, it's done around the DI phase, to boost the low peaks of a recorded guitar signal. Plus, the trouble I'm finding with commercial products, is they are expensive for me and don't do exactly what I want...so I figured learning and DIY a circuit might get me to my goal...which is preserve the old school human guitar signal played out a tube amp/speaker tone mic recorded, but with the convivences of modern tech (i.e. DI/reamp), without discernable processing presence...(hope that made sense).

author: jfharper
author_rank: Contributor
timestamp: March 15, 2025, 02:07:15 pm
content: Quote from: dkfan9 on March 15, 2025, 12:56:14 amThanks, yes, please if you can send the QSpice, I'd like to study that circuit in the sim...I'm not sure the proper way to transfer via this forum. Let me know if you need something from me.

author: themadhippy
author_rank: Super Contributor
timestamp: March 15, 2025, 02:13:43 pm
content: QuoteIts wot those emulator/simulators do.No they dont sound 100% like a fender and a pair of 56's,but they can get pretty close.Unfortunately there not that diy friendly as there doing fancy computery stuff inside to modify the sound.

author: SteveThackery
author_rank: Super Contributor
timestamp: March 15, 2025, 07:02:02 pm
content: Another possible approach?A compressor with a user-adjustable in/out curve is a very common plug-in for a digital audio workstation app on a PC. You can adjust the dynamics in the time domain (e.g. to adjust the attack and decay times) as well as basic amplitude mapping.Have you considered doing some of your processing in the digital domain?

author: jfharper
author_rank: Contributor
timestamp: March 16, 2025, 03:31:02 pm
content: Quote from: SteveThackery on March 15, 2025, 07:02:02 pmYes, but I wanted to save that as a last resort and use it as sparingly as possible to preserve the original analog tone.I’m curious if this circuit would be more like a normalize function done on a captured waveform…I’ll have to investigate that and see if there are analog circuits for that.

author: dkfan9
author_rank: Newbie
timestamp: March 16, 2025, 03:38:14 pm
content: Quote from: jfharper on March 15, 2025, 02:07:15 pmI've attached as .txt. Change the extension to .qsch after downloading and it should work, let me know if it doesn't. Note for translating to real circuit: I used Schmitt triggers with 0V hysteresis to replicate comparators.

author: donlisms
author_rank: Frequent Contributor
timestamp: March 16, 2025, 06:39:17 pm
content: I'm not sure if I completely understand what you want to do, mostly because I'm not sure how you define "smaller peaks." But I'll assume you mean exactly what you say -- there are peaks, some stronger than others, and you want to bring up the lesser ones. The question remains of what to do with the parts of the signal that arenotpeaks; the technique you use will most likely have some impact on those, too, one way or another.A straightforward compression technique will reduce the large peaks, but the compressor also usually provides a way to apply "makeup gain," which simply raises the entire signal. So... the larger peaks are reduced by compression, makeup gain puts them back where they started, but also brings up all of the lower signal levels. The lesser peaks, below the threshold, will be increased by the amount of makeup gain.In other words, in relative terms, what you describe is what a compressor already does.If you're getting tonal changes and other artifacts when you apply a compressor, there is a better-than-fair chance that your settings are too aggressive. The threshold and gain reduction ratio are critical (don't overdo it!), but so are the timing parameters of the detector. Excessive gain reduction, and/or an inappropriate threshold level, makes horrible things happen to the sound quality; in general, what I would describe as "rubbery," but there are other badnesses.For timing of the detector, the setting of the attack is most important; it can drastically change the effect of the compressor. If the attack is very quick, the compressor will respond to the very quick transients, not affecting the overall signal level very much. If the attack is very slow, the compressor will respond as more of an automatic gain control, like riding the fader manually. In between is a wide variety of effects that would apply to a wide variety of musical material and the goals. It's very important.Release is also important, and usually goes along with the attack time, both kinda fast or both kinda slow.Because almost all compressors can only reduce gain, and usually give control over makeup gain at the output, there is no reason to assume compression will bring the signal into the clipping. Avoiding that is a natural part of the workflow.There is a related notion that you might consider. It's traditional to record the dry signal, or as dry as possible -- as early in the signal chain as possible -- so that you can apply as much processing as you want to the recorded signal, try different things, undo, etc. If you record the fully processed signal, that's what you have to work with. Some of the influences there can be managed, but some cannot -- there are things you can't "pull back out" of the signal. So in general, engineers like to have that straight performance, and then take whatever time is necessary to try different things with it. If the sound in the studio is Just So Perfect, then record them both! Maybe it will be fine, and that's a win.Compressors are harder to set than they look at first glance. "One knob" compressors exist, and are intended to make it easier, but they lack actual Human Brain Power, which is a remarkable thing. My basic bottom-line suggestion is to make a dry test recording, and spend as much time as you can exploring compressor settings, to really own them, and see if you can get what you're after.

author: SteveThackery
author_rank: Super Contributor
timestamp: March 16, 2025, 10:55:58 pm
content: Quote from: jfharper on March 16, 2025, 03:31:02 pmYou seem to think that transforming the signal into the digital domain will somehow adversely affect the "analog tone". This is completely wrong, provided the digital-to-analogue conversion has sufficient sampling rate and depth. Indeed, professional audio processors and amplifiers use digital signal processing tocreatevarious "analog" tones.I raised the possibility of processing in the digital domain as a possible different approach, although I don't necessarily advocate it because it might not be a convenient solution. But I can assure you that losing the "analog tone" is not a reason to avoid it because, done properly, you won't lose anything.

author: Analog Kid
author_rank: Super Contributor
timestamp: March 16, 2025, 11:27:54 pm
content: This still sounds, feels and smells to me like ... an audio compressor. An adjustable one.Because what does a compressor do? It brings up lower-level signals to make them more or less equal to higher-level ones, right?In the maximum case all signals are brought to more or less the same level.Obviously, that's not what the OP wants.If it's adjustable, then you can select the "depth" of operation, in other words, how low a signal you want boosted to the same level as a high-level signal.Other than that, it would preserve the original waveform. (And here I would definitely be in favor of an all-analog chain, with no digital processing, despite assurances that this won't adversely affect the sound.)I think the OP's use of the phrase "smaller peaks" has misled us somewhat: I think what they mean is really just "smallersignals". So no need for any fancy-schmancy peak-detection stuff. (The OP can confirm or deny this, of course.)

author: CaptDon
author_rank: Super Contributor
timestamp: March 17, 2025, 01:00:10 am
content: Investigate so called companders. Compressor expander in a single rack mount unit. Many wireless mic systems use built in companders. During my broadcast career we used the CBS Laboratories Volumax and Audimax in series. These are way over your price range but can be simulated in the digital world. Also, as a guitar player myself I have access to a custom built multiband compressor. It compresses peaks based on the frequency where they occur without effecting everything else!!! No pumping and sucking of the low level signals in the bands outside of the band where the peak occurred!!! Think of it like a 31 band EQ with the compression occurring only at slider where the frequency of the peak occurred with instant attack and release! Kind of like owning 31 frequency dependent compressors!!

author: Analog Kid
author_rank: Super Contributor
timestamp: March 17, 2025, 01:17:21 am
content: Can someone supply a circuit that would do simple compression and be adjustable? Analog only?Seems like it should be pretty simple (but unfortunately over my head, otherwise I'd contribute here).Something that senses the overall level, perhaps at the several-microsecond level, and uses that to adjust the gain of a following stage, with the adjustment level adjustable.Seems like a couple-few op amps would do the trick ...

author: CaptDon
author_rank: Super Contributor
timestamp: March 17, 2025, 01:16:28 pm
content: I don't have the schematics anymore but during the radio broadcast days we built several iterations to use with our R.P.U. (Remote Pick Up) Marti VHF transmitter. One was built using a gain of wheat bulb and photocell but it was too slow for the fast peaks. We changed it to LED but the LED brightness to voltage ratio was far to non-linear for good results. Along the bulb/photocell idea was the old Raytheon 'Raysistor' integrated device. Our store bought LA3 compressors used electroluminescent with a photocell in a sealed package. While fooling around on the bench with our ideas we were able to get beyond infinity to one where a 3dB increase could generate a 10dB decrease on the output. We used the bulb/photocell idea to combine an in phase signal with a 180 degree inverted signal. There are so many different brands of high quality rack mount expander/compressor/limiter combo devices for serious recording and many under $200.00 U.S.D. Orban 422/411, Samson Opti-C and other versions, DBX with sharp knee and over easy switch selectable options. The 'plug-in' digital clients are too numerous to count and if they are good enough to be used by the big guys they should be good enough for the home hobbyist guitar player. Trying to 'preserve' the low level nuances while reducing the peaks is also a choice of the guitar being used as well as the players technique. Even the gauge of the plectrum/pick will effect the creation of excess peaks!!

author: Buriedcode
author_rank: Super Contributor
timestamp: March 17, 2025, 04:42:21 pm
content: Quote from: jfharper on March 15, 2025, 01:46:53 pmWhat you're struggling with is recording and capturing your tone. Sound engineers struggle with getting musicians to perform their best - and you're right, often its better to let them play "organically", wiothut headphones, and to hear the loud tone of the amp in the room - whilst also capturing that sound without clipping, or artifacts. There's an entire industry and libraries of books written about the subject.Given we live in the golden age of cheap high quality recoridng, and storage is essentially "free" (no tape anymore!) I would record the dry signal from the guitar, the output from the amp, DI, *and* the mic'd speaker of the amp. That way you can pick which one you like, or mix, or even better - re-amp the dry signal that you've played whilst it was plugged into a real amp, using a different amp.I am kind of confused because amp emulators, speaker emulators, impulse responses and general digital modelling of amps - including speaker cabinet, mic, mic placement and even room size/type - have been around for at least two decades. And now ther are free options of things like Guitar Rig, and Amplitube that do all of this for you. How "good" these sound is entirely subjective, but even the cheap or free versions are more than good enough, its just very difficult to knowwhere to startsince you have control over absolutely everything.Quote from: jfharper on March 15, 2025, 01:46:53 pmA limiter will simply soft clip the large amplitudes, a compressor will amplify the lower amplitudes, and can leave the large ones untouched, but often sounds a bit odd.Again, as others have mentioned you appear to have mixed several topics into one - a tube amp (or any guitar amp) is not simply "a bit of compression", it's alotof factors, which include compression, but also EQ, distortion (even clean amps have distortion otherwise it sounds awful) and dynamics, and lets not forget the speaker cabinet arguably has more of an effect than the amplifier itself. Also what you are describing reallyisa compressor, but perhaps one set with a 1:2 ration and fast attack.Things like "tone", "feel" and "70's tone, or 60s tone" are entirely subjective and cannot be translated to a schematic. It sounds to me like you enjoy the tone you get from your (tube) amp, and believe that to be due to its compression and "feel", and so assume that one could create a circuit to reacreate this. But as I said earlier, there are many factors that go into how your guitar sounds through an amp, including things like room size (resonances and room reverb really do make a hell of a difference) and compression or devices that change dynamics would be a very small part of that sound.Now, it is true that "tube" amps seem to have a partcularly quality just as they are on the verge of breakup, a sort of compression - and it was thought that solid state amps lack this, but that was based on very old solid state amp that sounded crap. Plenty of more moden solid state amps have this. And I will sympathise with you any frustration with trying to get that tone you have in your head, onto tape (well, hard drive) because whilst we have access to much better equipment these days - even the cheapest modern multieffects would blow away the best studio gear from the 90's - its more difficult because there are so many things you can tweak its hard to know where to start.

author: Buriedcode
author_rank: Super Contributor
timestamp: March 17, 2025, 04:54:49 pm
content: Quote from: Analog Kid on March 17, 2025, 01:17:21 amYou mean.. a compressor? A quick google yields quite a few shcematics.A good start:https://sound-au.com/compression.htmhttps://sound-au.com/project92.htmhttps://generalguitargadgets.com/how-to-build-it/technical-help/schematics/has some schems for compressors for guiatr/bassNote, a compressor for guitar/bass is a different application that one for HiFi (since guiatr is certainly not high fidelity) or vocals. And are used in different ways, so it depends on what you plan on using these for. Film production? Live music recording (where you just want to prevent clipping at all costs)? Or to fatten up a clean guitar solo?

author: jonpaul
author_rank: Super Contributor
timestamp: March 17, 2025, 10:17:45 pm
content: Eventide OmnipressorAudio cyclopediaj

author: dkfan9
author_rank: Newbie
timestamp: March 18, 2025, 01:06:57 am
content: In addition to the links buriedcode posted above, I came across this one, similar to many of the ESP LDR-based circuits but a different take:https://www.freestompboxes.org/viewtopic.php?t=32709

author: Analog Kid
author_rank: Super Contributor
timestamp: March 18, 2025, 01:19:37 am
content: O.P.: how about trying one of these circuits (fromreply #23), likethis one?Uses a lowly LM358, could be breadboarded in 15 min.You could at least see if it's even in the ballpark, as they say.

author: jfharper
author_rank: Contributor
timestamp: March 18, 2025, 01:21:31 pm
content: Quote from: Analog Kid on March 16, 2025, 11:27:54 pmConfirm

author: jfharper
author_rank: Contributor
timestamp: March 18, 2025, 01:38:16 pm
content: Thanks to everyone for posting this GOLD information. Definitely points me in a direction to research. Will take me some time to process (no pun) all while real life testing during recording. Thanks again and I'll post back with questions or success, if achieved.

author: Buriedcode
author_rank: Super Contributor
timestamp: March 19, 2025, 01:47:26 am
content: No problem, alas, whilst there is a wealth of info on the web, and in forums - I've found the only real way to get a "decent" recording that sounds like what you hear in the room, is to test lots of things. Its frustrating when there are a ton of variables, but worth it, because you quickly learn "yeah, that sounds crap" and so, what works, and what to avoid.

author: donlisms
author_rank: Frequent Contributor
timestamp: March 19, 2025, 04:42:25 am
content: Let me add one more idea that might no have been mentioned.I often use (at least) two compressors in series when I want certain effects, e.g. really rock-solid output levels from a widely varying input level. The first is set with a very low threshold, a modest ratio, and the longest attack and/or release time I can get - several seconds. The goal is to bring the signal into an overall average level. It still has its peaks and dynamics; this is more of what we might call “AGC.”The second compressor is then set for a more easy-to-adjust control of the peaks, and finishing touches.The end result can be better than what you get with one compressor, and yes - lots of experimenting and fiddling is required. In software, it’s easy to chain maybe four or five compressors, each with different timing characteristics and goals.In theory, all of it can be done with one very complex detector, controlling one VCA, but such a detector likely doesn’t exist, or at least isn’t available for my system. So using compressor chains lets me do it one step at a time.No amount of time spent exploring exactly what the compressor settings do, what they sound like when “just a little” or “way too much,” will be wasted.

author: CaptDon
author_rank: Super Contributor
timestamp: March 20, 2025, 09:57:10 pm
content: In the digital domain like when broadcasting from a remote location you can use 'pre-delay peak detect' where the signal is going through a delay of 10 milliseconds or so but the compressor sees the pre-delay and can decide how to correctly control the peak ahead of time. Something similar was used when recording vinyl. The cutter head would move farther inward to allow physical groove space for the delayed upcoming peak.

author: themadhippy
author_rank: Super Contributor
timestamp: March 20, 2025, 10:34:31 pm
content: problem with introducing any delay whilst playing live is yer ears and fingers will be outa sync and the brain gets confused.

author: Analog Kid
author_rank: Super Contributor
timestamp: March 21, 2025, 01:42:40 am
content: Might be good for players who tend to rush, though.(just kidding.)

author: RJSV
author_rank: Super Contributor
timestamp: March 21, 2025, 04:05:49 am
content: Yeah, I'd also prefer less delay, maybe 2 milliseconds max.My setup always has a stereo output, sometimes a digital reverb has the left and right outputs. A reverb, plus an octave pedal sometimes. I also purchased the Schober Organ tape reverb unit.Found out that I'm strong on technical, but no artist ! That meant selling my rig, and obtaining a bunch of YOGA mats, using the headphones for guitar licks but not ready to consider it 'ART'...not like a Hendrix or a Johnny Cash.Also did home-built Leslie (tm) tests.Ended up preferring the Fender style SPRING Reverbs...Rick-Jack(Send PM, I can give info on Chord short-cut (Hendrix), and my own 'RAKE' chord invention).

author: Buriedcode
author_rank: Super Contributor
timestamp: March 21, 2025, 08:34:54 pm
content: Regarding delay: It's true that we percieve delay much more acutely whilst playing instruments (hand to ear) as opposed to noticing lip sync issues (eye to hear), but 2ms... sound travels about 70cm in that time. It's nice to keep delays low because they all add up, but 10ms is really the minimum I've seen on multieffects, USB interfaces or (digital) wireless systems.