Me: "How many audio facts do you want to demonstrate with a Null Test?" Dave: "Yes!" Really like the videos and the way you communicate the information!
Another great thought provoking video, Dave! I have always felt that pro audio power amps always run at their maximum capable output when turned on and the knobs on front were input adjustable controls, attenuators, for each channel of output. Most people I've discussed this with didn't know that was the case since the volume does go down/up when you turn those attenuators. My point is that most power amps run wide open when their powered on. The input attenuators on front let you set input sensitivity not output. I'm speaking of pro audio amps without DSP as they don't have the obvious knobs on front but have attenuators inside their software but work on the same principle. I assume most if not all power amps work on his principle.
Agreed. Most amps are fixed gain with front end attention. I have seen some cheap amps that have gains on the front. Old Tapco if I recall and that was a huge noisy bummer that would often distort
Talking about amp attenuators, personally I'd like to turn down the attenuator knob on amps a little bit so that the 0dBu (-20 to -18 dBFS) on the mixer's main volume meter corresponds to the loudest average SPL desired for the facility or event. This not only allows for a higher S/N ratio on signal lines but also ensures a healthier gain structure for the whole system, making full use of sampling bit depth of A/D and D/A converters and (at least theoretically) improves sound quality by reducing rounding error in computing process of the mixer.
I encountered this idea recently and I was completely baffled by it, but your video clearly shows how it may have come about. You don't mention compression, but if a channel is running through a compressor, then a similar thing would happen, I would assume, as gain is pre compression, and the fader is post?
Compressors can be pre or post fader and yes, compression will increase feedback and room sound while minimizing increases in peak volumes. But that is a separate adventure
Also if you bump the gain to get more level you should do it before you set aux and matrix levels so you don’t affect recording feeds, wedges, in ears etc. great stuff!
What sound does kitty cat make? Kitty cat goes meow. What sound does rat make? Rat goes OMG A FU KING CAT! EVERYBODY BACK IN THE WALL NOW! DROP THE CHEESE, CHARLOTTE, IT"S NOT WORTH IT! At least, that's what my books said when I was in day care. Yeah, it was back in the 70s when ''accelerated'' programs were ''accelerated'' by exactly what Disco was being ''accelerated'' by (cocaine). Picture it and then spend the rest of your life learning to cope.
Dave, I recently helped spec out and install a new x32 at our church and I could not have been remotely successful if you didn’t share so much information for free on your channel. This is yet another example where you explained something I was struggling with in easily understood terms. Thanks so much for sharing your knowledge and experience.
There's a simple way to put all this. Gain controls are always on the input of a component, or the beginning of a circuit, and attenuation (Volume Controls) are last. Gain controls can distort the the signal if pushed too far, and volume controls can't distort the signal. To further visualize the concept, a guitar amp is a great example. On a guitar amp, the gain and volume controls are usually right next to each other. You increase the gain to overdrive the tube and distorts the signal. Once you have the distorted sound you want, you use the volume knob to set how loud you want it. Now, I understand you would never want to do this on any other type of amp. I'm just using it because its easy to visualize how gain will distort a signal, and volume won't. You can do the exact same thing on your mixer, or any other piece of gear that has gain controls. Its just that on a guitar amp with tubes, the distortion sounds good. Too much gain on solid state gear sounds terrible. One other thing I feel worth mentioning is resolution loss when you are adjusting volume in the digital domain. Normally, this isn't worth spending too much time on if you are working with good quality files (CD quality on up). I DJ as a hobby, and 95% of the files DJ's work with are lossy (MP-3). If you are aver working in this type of environment, its a relevant factor that you should be aware of. It really can impact sound quality. Even on the highest quality MP-3, most of the musical information in the file has been thrown away during the transcode. If you stick to just using 320kbps files (best quality MP-3), the noticeable effect is minimal. However, if you start off with lower quality MP-3's, you can definitely have issues.Its just something to be aware of.
Hmmm, got to be careful with "always" Power amps tend to be fixed gain with attenuators on the front end. Compressors can have gain on the in and the out, or just the out. Digital consoles tend to have both a gain and an attenuator in the form of a digital trim, and a pad is a fixed attenuation switch which is often before the first gain stage. But in general, what you describe is often the case with consoles
Excellent video, Dave!! I Think this is a very important topic for every sound engineer. What about digital consoles? I mean, gain stage is an analogic preamp but faders are digital. Does this applies also to digital consoles? Thank U so much. This video is going to make me win a pair of beers in a bet 😁😁
Awesome on the beer win! As far as digital, I am remiss to generalize but theoretically it should be the same. That said, I will look into doing a test as I have not done so
The earliest compressors I know of were 'clippers' (I'm thinking broadcast AM radio), thus, when you referenced the approach of 'hearing adjacent instruments' I said ahh, just clip the mic pre, and found it interesting to hear you discuss the approach of compression (albeit crude) at the input preamp stage. Nicely covered Dave.
Instead of over driving the gain on the console and risking damage to the console, wouldn't using a compressor to limit peaks in addition to raising background sounds be more practical? In my early years while mixing on someone else's console I frequently found channels 1 and occasionally 2 blown. They were where the kick and snare usually resided. :) I learned early the technique to getting a fatter kick and snare by overdriving those channels. I never approved of that approach. I invested in compressors early on.
This entire 18 minute video could have been summed up by just saying you take your mic level with the Gain knob, push the stupid Pre-Fader Listen button and make sure the Meter isn't staying in red. How hard can that be? By the way, that's incorrect to say that Faders can only take away level. Faders can add level by as much as 10db.
Or possibly you may have some assumptions that are incorrect and you missed an opportunity to learn something. And instead have held on to those incorrect assumptions. ttenuators, often refered to as faders and often linear slide but rotary aux sends are also attenuators, can only drop the level. Though people often are confused by the +6 or +10 and 0db labelling and think that because it says +10, it actually adds 10db. This is incorrect and one of the important aspects I am trying to point out. When arentuator ( channel fader) has a +10 capability (labelling) it actually means that there is a +10 gain stage somewhere after the fader. So the signal enters the fader and if the fader is at the max or +10, the signal is unchanged by the fader. When the fader is at 0db, the fader is dropping the signal 10db. Then, somewhere else in the console, the signal is "gained up" 10db. The 10db post fader gain is always there regardless of fader position.
0 dB is unity as in, what comes in is the same as what comes out. An attenuator will have 0dB at the max position. Like most amplifier inputs. But with faders and aux sends it's useful to be able to boost or cut. So 0 is 6dB or 10 dB down from max position. If 0 was at the top, then everything would either be at max or you would be losing dB through each fader stage for everything not at max
Educational and easy to understand, as always. Your videos are fun to watch even with a couple of decades of experience. This one was a great way to (partially) kill an old truth. I would like to see you test out the difference between standard 2-core mic cables vs. quad-cables. Is there an actual reason for us small folks to care or are the quad’s only for the larger stages?
Panning laws vary between gear depending on whether they are designed to maintain constant voltage, constant power, or a compromise between the two, very nice video!
Ha! I was waiting for someone to catch that! I was practicing my "erase unwanted objects in the vid" skills and when I masked that area I became a no hand zone. Thought it was fun to leave it
I often wonder if this observation that the gain control affects the mic's sensitivity is due to the fact they have a compressor on the channel and the gain control is also affecting the compression threshold?
This was awesome! So like you've been saying, bad if you push the preamp into clipping, it will alter your sound. Unless that's what you want. I hadn't thought about the fader at maximum being zero attenuation. Always thought zero on fader was strait through and there was +10 available. Wonder if this differs per manufacturer... Another great video, thanks Dave
@@DaveRat Unity gain was a mantra that takes a while to understand. If you want loud sound/music then everything has to be in balance, gain, EQ and compression (at different stages) and a human being with ears and understanding.
Hello. You likely answered this question about gain staging, but I just want confirmation, the fadder could end up at 0 db or -20 if it needs to be for a mix, but it is does NOT affect the quality of the signal so it's just a turned down version of the great quality signal and can't diminish it correct?
Yes, a fader just turned everything down. Equally. All the signal, all the noise, andt distortion. It does not add anything or change change anything. Gain does pretty much the same thing but it can have non linearties and has the ability to to distort the signal if too much gain is added, while a fader can not distort or gain the signal it can only reduce.
@DaveRat thank you for the clarification. Here's the case that brought the question, I gain staged a sm58 and a left and right for a pc music playback. The mains are powered meyer upjs so I have the main fadder at -20 since they are loud. The sm58 is for announcements and sounded great for the pa before feadback at 0db with some headroom. Now the playback is gain staged and is equally loud to the sm58 at -20 for a good balance. For some reason my boss says that the fadder needs to be at 0db is where it is its best quality. But I told him it doesn't affect the quality and it's a even balance to whoever does the announcements. Pretty dang loud. I try to explain playback has more width since it's all vocal, drums, etc down two channels that overpower a simple sm58 but he doesn't believe me. He then pushs it to 0db on the fadder and then turns the gains down. Basically turning down the gain structure to where there is no lights. He then said I don't know gain structure.
There are quite a few people that believe that the gain pot turns up everything and the fader somehow is smart enough to just turn up the instrument itself. That would be great but fader's nor gamepads have enough intelligence to differentiate. Both of them turn everything up or down What I believe this comes from is a misunderstanding and a confusion of the instrument versus noise If the game pot is very low and the fader is very high you will have a minimal amount of microphone and a maximum amount of noise If the gamepad is high and the fader is low you may have the same output level but you will have way less noise If the gamepad is really high and the fader is really low you may have the same level but the input may be distorting from the gain overloading. You want to set your game pot approximately in the range that is below distortion but not so low that the signal is close to the noise floor and not gained up enough. For most applications this is a nice wide range. For recording purposes and applications where absolute minimal noise is necessary then you would want your game parts as close to clipping as possible without clipping. There is one advantage to having your fader around zero which I show in the video is that the sensitivity of motion is the highest resolution around zero where you can move the fader quite a bit with very little change as you get to very low fader positions tiny movements cause major changes. So in a way your boss nd you are both right. If you put your fader at zero or where you want it like -20 and set your gain so that you get the level you desire All should be good
I know this is an old video, but I just found it. Very good! I wonder if you could talk about compressor out gain pots? How do they work? Sometimes I wonder if there's much point to them if you're only compressing things a little bit.
With the gain down, fader up or gain up, fader down I think getting a good fat non-clipping input signal is vital to getting proper insert input levels , recording levels etc, otherwise there tends to be a chase the gain competition with too much make up gain or boosting record send levels and therefore mixer background noise. Also I don't know how it could be done but a way of comparing digital console gains, their digital output attenuators (not faders) and faders would be interesting. There are ways in a Yamaha console to tie yourself in gain knots.
Correct input gain is critical to getting good sound. That and correct mic/speaker positioning probably gets you 90% to a good mix. Assuming your gear and cables aren't faulty of course.
I will look at doing the same test with digital. It should work as long as latency is locked in. And as far as gains, send hot and attenuate late to reduce noise while staying below clip to avoid distortion.
You explain things so well, all of these things are very important in understanding pro audio gear. Please continue these videos, really appreciate it 🙏
Thank you Dave for taking the time to make these videos, I look forward to seeing your next one on this subject. Im glad you got your body parts back!! :-)
Hi Dave I'm a big fan! I'm wondering what your opinion on the use of limiters in live sound? I don't like to use them, but someone recently challenged my beliefs and I'm wrong a lot so I'm investigating. I've seen guys use them on the channel, master and (most common) within the dsp. Thanks for the videos man!
Limiters are extremely useful when used correctly and absolutely necessary to get maximum system outputs and to protect systems from damage. Setting a limiter so that the audio does not get louder than a predetermined level is wonderful. Lims and great to deal with singers that scream, or certain guitar effects boost to super high or just to smooth out inconsistencies and for system protection. That said, most of the time most of the inputs should be not in limiting and limiting should usually only be audible if it's set up wrong or the system or input is being over driven. The limiter turns it down so ya don't get blasted, things distort or blow up.
When I was younger I was told to think of gain/volume like a water hose/faucet. Your water pressure is a set amount of max “power” that the hose can produce, and the faucet is how you adjust the amount of water. Essentially, you never change the water pressure (gain), you open and close the faucet (volume) instead. My understanding of the concept has become a lot more technical than that, but I always thought that was a good starting point.
Hmmm. If you had a water tank and a variable speed pump, that would be your gain. The higher the pump is set the more water pressure But the pump is running the water into a hose with a faucet, that is your fader or attenuator. With the faucet all the way open, the pump is in total control of the output. The faucet is only in control of off all the way up to, but never beyond the pumps current output.
Would the results of this null test be the same if a mic were used instead of the signal generator? When I first started doing live sound I was running into feedback problems and another engineer told me that taming feedback is all about input gain control-if you turn down the input gain and turn up the fader or compressor’s make-up gain (this is on a digital console), you can achieve the same volume level in the wedges without feedback. I’ve always sort of wondered if that were actually technically true, but I’ve been functioning as if it were true for years and it always seems to solve feedback problems-maybe I’m fooling myself? What was explained to me is that the input gain actually makes the mic more sensitive while the other stages don’t-which is apparently the common misconception. If the thing we’re trying to disprove is the input gain’s influence on mic sensitivity, I would be curious to see what would happen with a mic instead of the signal generator. If a test were done that way, would the Y-cable create a weird variable related to whatever’s happening between the preamp(s) and the mic?
Ahhh, the gain vs fader to fix feedback. Fact or fiction? Did a vid on exactly that topic. ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-chgovyDUxx4.html
Thank you and so cool! Let me know if I can ever be of assist. Also, Catbox 8 with 4 channel with both male and female XLR, is coming soon. It has a removable internal jumper that converts it into a catbox male and catbox female. Nearly done and just approved the metalwork
Some cheaper pre amps have a relavant amount of noise. Does the Signal to noise ratio get worse on different gain settings? That way you would get a better sound by pulling the fader up and setting the gain low? Or do preamps have their singal to noise sweet spot at around 0 dB?
Thank you again for being such an amazing educator. I have sort of worked backward when doing musical theatre sound where I set the amps wide open, mixer outputs at unity, the faders at -10, then run the gains up to a good audible level, but way short of clipping (and get decent headroom). Where this breaks down firstly is noise, but usually not too bad from mic packs. -It's when I'm tracking to a DAW to have students practice mix, or when I make a board mix, the levels are way low. I have since been attenuating the amps more, but we are blessed with pretty great noise floor these days. Lighting is the new hum coming from the stage...
Yeah, there are many paths to get good results. Things like having too much PA for the venue or low to medium volume level shows, will inspire reducing amp levels to keep gains up and faders in a reasonable position. Generally it is better to send hot and reduce the level later in the signal chain to optimize noise floor, as you started doing.
Ha, I've heard the whole background noise (spillover I call it) will increase with gain pot. I saw this "problem" is actually coupled with the mains fader (aka Gas) position/amp attenuator gain structure than with the fader position. The most contested example is where do you set the cgain for High hats since using VU meters to get gain got you a hot mic in the mix. I saw so many touring engineers who turn hh to - 15 with the fader, instead of the more common turn the gain down and mix at 0. I mixed on a soundcraft venue in the 90s. The deal with the high hat in my opinion should carry over to today: I took the perspective that the pfl led / vu were both c weighted with different attack /release times. The problem is actually with the higher frequencies being represented in VU due to log /power issues were not represented properly by either vu display. So the proper way to set the hh gain was based on what sounds blended volume for unity gain setting.
No reason to make things so complex and there is not much magic. Gain and faders do the same thing, they bring everything up or down. But, too low gain will add hiss and too high gain will distort. Once that sound is established, the fader just increases or decreases it. I just did another video on gain and balancing the four goals of gain structure: Reduce noise Avoid overload Avoid faders near the extremes Establish visually desirable fader positions
Another 10 out of 10 video! Where do you find the time to run a company, experiment with tech, and make videos for us shmos? Not sure if this is the right time/place, but how bout a video on the "dangers" of UNDERpowering a speaker?
Just came across your channel. Excellent content. You mentioned that it's impossible to clip with an attenuator, which peaks my curiosity. I'm new to audio, but I have done alot of research. I've been trying to record my crotch rocket motorcycle exhaust and have been having troubles with distortion and clipping. I first tried the zoom h1n, then upgraded to the zoom h4n pro. Still no dice. So I recently acquired a shure sm58 and a xlr cable. I assume I will still experience clipping, and my research pointed me towards something like a shure -25db or greater attenuator. Would this solve my issues? I was also looking at a 3.5mm attenuator cable, and try and pair it with a lavalier and the zoom h1n for the same project. Have you any experience with 3.5mm attenuator cables? One last question I had. I recently discovered this week about the new 32 bit recorders. And I was reading that there is no gain to set on them, and that the 32bit eliminates clipping. Is this true? And if so, would I be better off upgrading my current equipment and buying the new zoom h4n essential, and possibly the zoom h1n essential or the zoom f1/f3? Your guidance would be very much appreciated 🙏. Thank you
Yes, an inline attenuator will help the zoom grab the sound clean. But... You also need to make sure the mic itself is not clipping. You will need a mic with a high max spl and keep the mic away from direct puffs of air/exhaust as the puffs will move the diaphragm too much. Good luck! Lmk how it goes
@@DaveRatThank you for replying back. I was going off of an article from the shure website, and it said the sm58 could handle a max of 180db spl. The article says a space rocket taking off is 180db, and I can't imagine my motorcycle being louder then that haha. They also said the recording device would clip before the sm58 would, and they recommended using an attenuator with it. This article was my reasoning for choosing this setup. I'm planning on lining the inside of the tail, (underneath the rear seat) of my motorcycle with some sound proof foam. Then I was going to get a cheap pelican case, and use the block of foam out of it, and cut to fit slots for both the h4n recorder, and sm58 microphone to fit into. I would be placing both devices under the rear seat, in the tail section. My hope is the sound proof foam will help cut out wind noise, and also muffle the exhaust sound a bit. What size attenuator would you recommend? Should I go with a -25db ? Or a -50db to be on the safe side? The feedback is greatly appreciated. I linked the article I was referring to in case you were curious to read it. service.shure.com/s/article/can-a-dynamic-microphone-handle-really-loud-sounds-maximum-spl?language=en_US
For the attenuator, 10DB is 10 times the power so 20 dB is 100 times the power and 24 dB is like 300 times. So a 24DB attenuator is 1/300th That should be sufficient but the way to tell is to make sure that when you hold the mic near the tailpipe and you set the level on your h4n that the input level is not turned way way down. You want it so that it's like at 50% or higher setting gives you a nice level below clipping. I would put the h4n under your seat and run and mic cable to the microphone placed down by the tailpipe somewhere. To keep wind noise reduced I would put a big foam windscreen on the microphone The biggest fluffiest one you can find and then I would put something like a dog collar cone around the mic with the point of the cone pointing forward and the mic pointing backward inside of it. Then I would take that mic cone windscreen combination and mount it with the mic pointing backwards somewhere maybe behind your seat as close as you can get it to the tail pipes. I would avoid putting any kind of box around the microphone because that will screw with the sound. If you still get too much wind noise then find a bunch of open cell foam And loosely wrap the mic into a big Tootsie roll in it And then put the dog cone over that. Cool cool let me know how it goes
@@DaveRat Thank you, your advice is very much appreciated. I will have to give that a try when I get everything together. I still need to aquire a xlr cable, and some sound proof foam for this project. I'll have to say though, I wasn't so much concerned about the wind noise, as I am of the sound coming from the exhaust. This motorcycle is pretty loud, and when the engine gets up into the higher rpms, it literally hurts your ears if you're standing near the bike lol. I'm not sure how many decibels it is. I was going to get a cheap decibel meter, but everything I seen only goes up to 120db. And I'm certain the sound level from the exhaust exceeds that 120db at times. I'll just have to experiment and see what works. Your video on microphone placement was very insightful, so I'll have to experiment with the microphone placement. Thank you, much appreciated.
OK, the most crucial thing here is signal to noise S/N. It is important to understand that all electronics create a certain, fixed amount of electronic noise. If you input a low signal into any component, then you will have to amplify it at some later point and this will also amplify the electronic noise that has come along with the signal from upstream stages of the signal path. Therefore, in order to minimise the proportion of accumulated noise at the output, the signal level must be maintained as high as possible without clipping through every single stage of the audio system, so that the total amplification is as low as necessary. This is known as gain structure and is one of the most important concepts in all audio systems. However, you do need to make artistic mixing choices but in doing so, you should be trying to apply this principal as much as is appropriate. Generally, the 1st thing a source signal like a mic will hit will be a mixer input channel preamp. The signal from the mike will vary, some singers are loud and others soft & some mics are less or more sensitive than others and mics are a lower signal source than say a keyboard. The channel input gain should be set so that when the signal from that source is at it’s loudest. The channel clip light may flash occasionally because clip lights generally come on a little below the actual level of clipping, so the occasional flash is just a warning. Generally, analogue gear is pretty tolerant of clipping and just goes into a non linear rang of amplification, while digital gear can be extremely intolerant, having almost no headroom before hard clipping. You get to know your gear after a while. Anyway, at this point of 0VU we have what’s called unity signal. Now then, the next control is the input channel fader. Depending on the mixer, you may have groups, so you would for example assign vocal input channels to a vocal group channel. In this case there should be a group output meter. As this group channel is receiving combined signal from several input channels, the combined channel faders act as the groups input gain control. You will know that the combined group input signal is at unity when the group output meter matches the group output fader setting. If the group fader is -20dB then you want to adjust all the input channel faders to an artistic mix and at the same time meet that -20 dB target on the group meter. Therefore, setting the group fader to 0 dB will meter the group output 0 dB (without clipping) or setting the group fader to -50 dB will show -50dB on the group output meter. This ensures you will have no unhappy accidents. Now, the groups will be assigned to the main output buss which will also have an output meter, and the output buss’s input gain control is the sum of the sum groups. So again, whatever the output fader’s are set on, is the target for setting the group mix. So, you find your artistic mix and move all the group faders up or down until you meet you output target in exactly same way you did for setting the group target levels. Now you have setup a gain structure that maintains unity signal through the whole board, in that when the mixer main out is set at unity, that is zero dB, the output meter reads zero dB. You may have other gear in the signal path, such as FOH EQ, system EQ, DSP and then Amplifiers. The point is that you want to have setup the system inputs and outputs so that a unity signal from the mixer is maintained through the whole signal path right up to the amps, but the amp adjustments are a special case. Setting up the amps. You need to decide on a maximum target volume and be able to measure that with a standard pink noise. Sent from the desk at unity level. A word of warning, you don’t want to actually do this adjustment at unity as that is likely to damage your hearing. If you decide say 95 dB SPL is your maximum target read at 1m from a speaker. Then do the settings 75 dB SPL with a -20 dB pink noise signal from the mixer output. This makes the whole system predictable and ensures you audience hearing is protected and noise compliance is met. Some amps may have a gain selector switch on back, allowing you to reduce or increase amplification gain appropriately. Using the least gain possible, allows you to minimise signal attenuation at the amps input (volume control) and this is the last stage in setting gain structure to achieve the best possible S/N result. That is the cleanest most pure sound your system can produce. There are other things than can improve sound reinforcement fidelity but that’s a much bigger discussion and the answer is as always “ Well it depends”.
I remember being told ages ago that it sounded best to run power amps at full, and even that it wore out a power amp more quickly to run it lower. I always wondered, and I kinda felt like it might be BS but I tried to stick to it "just in case". Good to hear some common sense and proper info!
Fun fact: The BBC used to take the term "fader" literally. On a BBC mixing desk, to reduce the level of a channel in the mix you would push the fader up the board, away from you... Up = more fade = quieter!
In Broadcast, that isn't always a bad thing, especially if you've got post backstop prefader aux sends where a slight unintentional nudge of a fader could result in something undesirable happening. That is much less likely if you have to pull on a fader to open up a channel. Where it gets really interesting is if your main board has "BBC Type" reverse faders, and also a secondary desk that is "conventional"!
@@kevinwood4592 Hi and congratulations....... I have, I think, have a pretty good understanding of my way round a mixing desk. However "post backstop prefader aux send" had me stumped. So, in good t'interweb form, I googled it, and whaddya know...... Drum roll...... A Google whack! A truly rare beast these days. So, perhaps you would be kind enough to fill in this gap?
Interesting, I did the test using pink noise which puts out energy at all frequencies in the audio spectrum. Using a microphone would give us some of those frequencies depending on the signal but nowhere near as much total data as the pink noise. What advantage for benefit in the test would be achieved by using less information provided by a microphone?
Well, quite possibly none 😅 I just wondered if somehow they might behave differently. There's no good reason they should so it's mostly just plain curiosity. So many people ask this question, and always in relation to the real life situation of mics on a stage.
Okay I will ponder that. The console doesn't really know whether it's a mic or other type of signal so it wouldn't have a way of doing with them differently. Since a fader is just a reduction of the exact sound after the gain pot. The only way high gain vs low gain would be different would be for the gain pot to sound different at different gains. It is good to question whether tests are relevant. I guess if the question of whether a console deals with mics differently than other signals would also mean that all the specifications would be questioned. Freq response Distortion Maximum output Are all determined using test signals, and if a console dealt with mics differently that test signals which are basically non musical music, then none of the specs would relevant. But also with a console it is fairly easy to test whether it deals with mics differently than signals. Just set up 2 channels to null as I show in the video. One channel high gain low fader and the other channel low gain Hi fader. Once you have a null with test signal, just swap to a microphone or guitar or DI box or whatever you wish on the input and see if the channel's still null. And then let me know what you figure out!
Ok I shall report back! When I next get a chance to have a play around. I don't have enough parts to make the y-split, so may do it by duplicating then phase reversing the input to two channels on a digital desk. Theoretically the results should be the same, but let's find out!
Theoretically the results will be more accurate as you eliminate different mic pre. But you also eliminate testing the mic pre as a factor. But the question is not whether a mic pre is less or more linear with a mix than a test signal, so electronic split is good
I'd venture to say that ALL volume controls are faders - unless they're part of the amp's feedback circuit. Only then do they actually control the gain of the amp itself. That might be found on the input channel gain pot. Good video as always.
Great stuff,,, stuff so many guys don't understand.. so well layed out in this video. Gain, SNR, over driving, knowing what the gain stage vs attenuator knob does.....Love it! So good, I shared it out..
Lolz! I feel dumb. Never connected that fader/attenuator was only reducing the level coming from the preamp. I guess as you said the +6 or +12 marking by the fader was kind of making me think it also boosted the signal in the board. Thanks!
Hi Dave! could you please show the difference between speaker generation and how newer transducer reproduce a way better signal then the old wood box with JBL speakers. still to this day i have people come talk to me on gigs about how good A7 and 4530 where sounding at discos. I believe there memory is flawed and the newer technology reproduce a way more accurate signal
Interesting. All the good old days when everything was better, but mainly because our rates were less refined and our memory is flawed. That said, there ire some sonic differences between the big high efficiency cabinets powered with less amp and the newer smaller cabinets powered by more watts. I will ponder a punch, dynamics, efficiency and freq response vid. Cool cool
This is a great video and clarified many of my questions about gain and clipping. I never really understood what was happening when a signal goes into clipping.
What about frequency response of OP AMP's ? If you will regulate feedback loop (in electronic cirquit) then frequency response will rise or fall - that said some basic of electronics. Gain is not only voltage pump, in OP AMP happen more stuff than only voltage leveling, let's not forget about that. We don't know how OP AMP is built and steering in every console, so it is possible that the feedback in the monitor effect is based on some other variables. next topic is how GAIN itself is realized. In Yamaha LS9 you have physical gain jumps every about 20 dB (if I remember correctly) and everything between is software magic. Approximation for digital trim is another variable in that process... You will get that information from mixer serviceman, can't get that knowledge from marketing brochures or even selling manager specialist. So it should be as simple as you said but no, it is more complicated, esspecially if you are working on many equipment, you can find that every machine works it's specific way. I would not say that "experienced sound engineers" as some crazy guys just because they "feel" something that can't explain right now...sometimes the secret is hiding in the patented technology of that gear developer and we can't get informations how it is built and how exactly it works, but we feel that something works "that way" althrough it is not so obvious if we think literal way like "gain=voltage pump". Cheers. Thanks for that video.
Good stuff and fun to test. On the Yamaha console, doing the same out of polarity signals with differing gain/fader settings and seeing if they cancel, would be interesting. I personally believe all can be measured. And yes there are things we feel or hear or think we hear that we just have not figured out an easy good way to measure yet
The analogy I always used to explain this was gains are water pumps and faders are spigots/valves. I use gains to calibrate various inputs to the same level for the board and the faders for the use case. Much the same as rural homes on well water require a stronger pump than a city home on a water main does, but the taps are turned up and down, from fully open to fully closed, as needed. As far as calibration I mentioned, I think of it as adjusting for different temperature scales. Some inputs come in using Centigrade and others using Fahrenheit (e.g., -10 dBV and +4dBu).
I'm new to PA systems, but I might disagree with his analogy of the PAN pot. If it's in the middle, both channels L/R have the same max level. If it's halfway left, it Will have half(or logarithmic half) level on Right channel, but the Left Channel still receives Maximum level. I hope someone corrects me if this is wrong.
A pan pot is basically two faders where the center position is 3db, 4.5, db or 6db down for both left and right, depending on the pan pot type. That's worth checking out as well. As you turn the pan pot it fades left up and right down and visa-versa when turned the other way. If you connect a mono PA to the left output, you can use the pan pot as a fader by panning to the right. Or connect the pa to the right and pan left to fade.
@@DaveRat Thanks for confirming this. That's what I meant. As soon as you turn pass 12 o'clock, let's say to the right, the right channel will Remain at max, and the knob technically become a Max-Zero Fader for the left channel.
Not exactly. The right is max only when the knob is turned all the way to the right and as you turn to the left, the right begins to fade down to between -3 or -6 when you reach 12 o'clock. And fades gradually faster as you continue to turn
Regarding "gain adds more bg noise and fader does not", I think that this sentence is for more complicated setup than you presented. This sentence will be definitely true if you insert a Gate to signal chain. Gate will be after Gain pot so if you increase Gain, you're like lowering Gate threshold for background noise. But if you raise fader - all sound that was filtered will be still filtered by Gate and only that filtered sound will be increased. But anyway, I really liked those examples, especially with clipping gain example!
Gates and comps will totally change everything. The belief that by some magic, a fader knows to turn up the actual instrument while a gain pot is a big scoop that turns up everything including the instrument is somewhat widely believed. I am guessing this is due to people clipping the input gain which in effect creates a bad sound limiter, in effect reducing the max level of the instrument while increasing the background noise. Then, turning down the gain and turning up the fader would in effect increase the instrument and reduce the background noise. But, without clipping, that whole flawed concept falls apart. Basically, just don't clip the input and all is good
Super cool! I always wondered if there's any truth behind the gain versus fader sounds. I'd assumed it was just hearsay since I never seen someone prove it, but I never thought about how a gain pot could add clipping and distortion that a fader would not.
Yes and also, super low gain and high fader will add some hiss as well. But the actual sound of the audio should be the same as long as you stay away from the extremes
Dave: Would this still be the same if the gain pot actually controlled gain rather than just attenuating the input signal? For that matter, are there really gain pots that control gain?
Gain pots do control gain. The gain pot determines how much gain(boost) to the signal and the attenuators ( channel, group and output faders and rotary aux sends) determine how much to reduce that signal. What I am showing is that a gain pot does not sound different than an attenuator
My brain was like, "now I know how to remove the vocals to create an HD karaoke track" .... The other way the background is raised with the gain is when the live sound system has a compressor inserted on the channel, and the sound guy is confusing the compressors influence (greater influence) than the gain pot.
True, a compressor would impact it. But I don't believe the flawed belief is based in compressors on channels. I've met some smart techs and engineers that grasp the interconnect wiring and operation of gear but lack the electronic understanding, that believe this oddity.
Great explanation Dave. It made me think of something the sound tech who mentored me told me once. When discussing gain staging and fader placement, he talked about how he knew many techs who put all the faders at 0 and adjusted the gain for volume from there. Saying that because of voltage, having the faders above or below -/+5 altered the audio in a negative way, and as you mentioned apparently kept leakage from instruments around out. Does this have an bearing on what you are speaking about here?
There is nothing wrong with adjusting gain to get the faders in a good position. I just did another gain structure vid for the member's side. It will be public at some point
iPod feeds into dj mixer to gain then ch fader then out to volume output which does route to FOH mixer which goes to there gain then fader to the bus to output which in that case went to powered pa every thing has a volume option to twiddle with. So let’s hope that iPod volume was right or all the rest is a mess ha ha ha Dave. Just did a video featuring you will be live probably first Friday of next month. Party on sir. Love you to do a vid on the dorrough 1200 in practical application. I love that thing. Oh and remember anytime you ever want to sell your DBX analyzer be sure to let me know I have still not found one of those anywhere
love your videos. Usually for my own musical adventures. This video went into a separate issue I've been very curious to learn more about. Cancellation while I've noticed this playing on different systems and having people over to my house where after hours of trying different settings someone will show up with their amps that do not go louder then the ones I have set up but I hear dramatic differences in all noise. Aside or in addition to this I've been contemplating cancellation towards other applications like sonar, radar, wifi and millimeter wave singles. With absorption, attenuation, reflection agaisnt standing waves and modulated frequencies. Your phase example was exactly what I wanted to see. Thank you. Of course your follow up with the exactness required all but eliminates the use of frequencies as a shield against frequencies with frequency hoping and the numerous adjustments / modulations possible.
Excellent video Dave, I think I commented on a previous video of yours about this subject and I've always found it really interesting. When I started out I definitely had some placebo effect where (before monitors were being taken into consideration) it seemed that if a vocal mic was feeding back, you could reduce that feedback by pushing the fader up the same amount as you pull the gain down. It's very easy to fall into these traps working by yourself in a club setting and with other people repeating the same information. I really really really want one of those tiny micro wedges! Is there a chance these would be buyable at some point? ;) Have a good weekend!
With some recording preamps/gain stages, they would add harmonics/saturation even before clipping as I understand it. Do most live sound boards have super clean preamps, and not saturate before clipping?
The live sound pre amps vary but in general added coloration is not sought out. That said, pee amps that still sound good when clipped are more desirable.
I feel that mic amps sound ' faster ' or ' punchier' when they're cranked. In my mind, the slew rate is faster the further the amp has to throw the signal, it must be faster if it is achieving more voltage in the same time?
Hmmm, using that analogy, would it not be easier to move something not very far but very quickly and to move it farther, it would more difficult and slow down? Meaning gain turned down would be quick responsive and turned up would be sluggish? Not unlike an electric car off the starting line, super punchy but at 80 miles per hour, hit the gas and all the punch is gone. Regardless, the analogy does not hold well. There is no mass to electrons and also it's pretty easy to test and hear differences using a null test. The primary factors are staying above the noise floor and below clipping. Followed by having the fader in a usable place around the zero db mark or where you visually want the fader to sit
Thanks for this. I was always told that proper gain structure would result in better signal to noise ratio and now I know that that’s only true if we compare clipping to not clipping.
Higher gains do give better signal to noise ratio. But if you count undesirable distortion as noise as well. Then a gain somewhere below clip and above hiss becomes optimum
Killer demo! I love the way he can show you these minute differences (gain pot versus fader) and explain how you can get into the areas where they aren't equal!👍
16:40. You clipped your left hand.. 😂 jk. Great stuff! I work mainly in recorded audio but there’s plenty of crossover with live sound. I’m also a sucker for null tests. Its the part of audio that is scientific enough to prove what we think we hear isn’t necessarily correct. Thanks for the videos!
Very interesting Dave! And power amps deal with lots of wattage and amps, whereas a mixer is voltage? With an active level control on an amplifier I like to run it at +0 boost -0 cut, and use the mixer to set what needs more reinforcing. Kinda the same with a non-active amplifier but I turn the amp to max (10 or all the way up if it has no numerals) and again use the mixer to set what needs more reinforcing.
I may be wrong, but I find a bit of truth in the “gain brings everything else up” argument when using condenser microphones. I feel like the amount of gain effects the microphone directional response and sensitivity
Hmmm, gain does bring everything up or down. And also the output of the gain stage goes into a fader that brings everything that comes out of the gain stage up or down. So, with the fader all the way up, you hear purely the gain stage. As you bring the fader down you still hear purely the gain stage but at a lower level.
Loved this, when setting up a new venue would you be looking at the amps (gain boost) or wattage then calculate how hot the system would run and how much you'd need to hit a specific decibel rating? The system I run is well setup but haven't had to choose and set one up from scratch yet. Talking Mid size venues around 1500 seats :)
In analog audio gear, an attenuator typucally refers to a passive resistive slide or rotary potentiometer that reduces the level of a signal. In digital gear the function is emulated to respond in the same way.
3:53 thank you thank you thank you thank you. I'm a mechanical engineer with amateur electrical experience. I've got a Mackie 8 channel board and was looking for a simple explanation between gain and fader. You knocked it out of the park when nobody else does.
If you draw an imaginary dotted line around the combination of mic and preamp then you could say that it does increase the sensitivity of that "circuit block", but then you could draw the same line around every other gain stage in the system and say the same thing.
😅 🤷♂️ one of the most respected and recognized recording and mixing engineers in my country stopped treating me or communicating with me because on twitter a long time ago I said that input gain and output level were not the same because they do not give the same results when not behaving with the same characteristics in their performance. I said that we had a daily example with guitarists and their behavior in the use of their effect pedals. I simply had to record a video and demonstrate it. This video is key. Once again, grateful. 👏☕
