We agree, loudspeakers are always the weak link in the chain. And yes i was talking about opamp (njm4580).
My point was more to give some perspective on how to mod the pad to adapt it to (current) drive capability of opamp: RoboDNA is not really a noob with electronic but i suppose he is self taught and could have 'holes' in his knowledge ( as i have, my knowledge in EE come from having had to take care of consoles in studios i worked in and my own circuits mostly, but i'm audio engineer first, not an EE).
Anyway, the console schemo are linked if someone is curious about it.
Jaddie, if you put an eye on it and you know how this kind of balanced ( impedance) output is called in english ( page 31) i would appreciate if you could give the name ( i already seen it named but forgot it: eg: it's used in ITI, Sontec and GML eq).
My point was more to give some perspective on how to mod the pad to adapt it to (current) drive capability of opamp: RoboDNA is not really a noob with electronic but i suppose he is self taught and could have 'holes' in his knowledge ( as i have, my knowledge in EE come from having had to take care of consoles in studios i worked in and my own circuits mostly, but i'm audio engineer first, not an EE).
Anyway, the console schemo are linked if someone is curious about it.
Jaddie, if you put an eye on it and you know how this kind of balanced ( impedance) output is called in english ( page 31) i would appreciate if you could give the name ( i already seen it named but forgot it: eg: it's used in ITI, Sontec and GML eq).
Thanks. Yeah, they did it cheaply. It's an unbalanced output if Pin 2 is used. The "balanced impedance" is a way to make it interface better to a balanced input, but nothing like a real balanced or differential output. Its disappointing they could afford to put in one more opamp that costs $1 in singles, and half that in quantity to make it really balanced. Between that and the lack of a monitor section, this is really a low-end board. Probably sounds fine, just under designed.
Well it's a 'real' balanced impedance output, there is no issue about this ( the balanced line receiver sees same impedance from pin2 and 3 so it is indeed balanced, as stated by Bill Wihtlock from Jensen transformers, dad of THAT 124X/1646 line drivers and receivers ).
And i'm not sure i would say 'real' or disapointing on that point ( however i agree about the other issue you raised): Massenburg or McNeal are on par with Neve , Toft or Porter... they know what they are doing and this circuit never bothered anyone when using gear using this kind of design/topology ( in fact Sontec, ITI or GML are considered as the creme de la creme regarding parametric eq - all are Twin T topology and all share this same ouput topology).
When you say 'real' balanced i only see transformers as being 'real' balanced. As much as i love transfo they are not the last word about transparency.
Driving differential can be really nice i agree too but as long as the line is balanced ( impedance) gains are marginal imho. That said if i can use the THAT driver/receiver i happily do, those are transparent, mimic transformer behavior, can be dc servoed, cost 1/10th of a good transformer.. what not to like about them?
They have one only drawback imho: no galvanic isolation. But at 1/10th the price of a Jensen or Lundahl...
And i'm not sure i would say 'real' or disapointing on that point ( however i agree about the other issue you raised): Massenburg or McNeal are on par with Neve , Toft or Porter... they know what they are doing and this circuit never bothered anyone when using gear using this kind of design/topology ( in fact Sontec, ITI or GML are considered as the creme de la creme regarding parametric eq - all are Twin T topology and all share this same ouput topology).
When you say 'real' balanced i only see transformers as being 'real' balanced. As much as i love transfo they are not the last word about transparency.
Driving differential can be really nice i agree too but as long as the line is balanced ( impedance) gains are marginal imho. That said if i can use the THAT driver/receiver i happily do, those are transparent, mimic transformer behavior, can be dc servoed, cost 1/10th of a good transformer.. what not to like about them?
They have one only drawback imho: no galvanic isolation. But at 1/10th the price of a Jensen or Lundahl...
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From the standpoint of balance, not a big deal, though you're up against component tolerances. A balanced line receiver can only achieve maximum CMRR when those impedances are exactly equal over a wide frequency range, and that's hard to do with 5% value parts. Yes, it can work, and does.
From the standpoint of driving the next device, the single-ended balanced impedance topology lacks 6dB of headroom relative to a real differential output. Not a big issue in this case, but driving pro gear, particularly vintage pro gear with max in/out at +28 or higher, it's a problem. And it's so cheap to fix.
Yes, the THAT chip combination is a win. I have lots of gear with them. And to me, they are more of a "gold standard". Though I do like my Jensens. I also work in some rather high RF environments, which changes the picture a lot.
"Balanced" means relative to a reference, often ground, and as respects impedance. There's nothing that's not balanced about a transformerless output if done right. A transformer can be isolated from ground, but it's still balanced with respect to the reference.
Yes, no galvanic isolation, but also (if you're not using Jensen or Lundahls), no transformer grunge either. There was a time when "transformerless" was a huge positive thing.
From the standpoint of driving the next device, the single-ended balanced impedance topology lacks 6dB of headroom relative to a real differential output. Not a big issue in this case, but driving pro gear, particularly vintage pro gear with max in/out at +28 or higher, it's a problem. And it's so cheap to fix.
Yes, the THAT chip combination is a win. I have lots of gear with them. And to me, they are more of a "gold standard". Though I do like my Jensens. I also work in some rather high RF environments, which changes the picture a lot.
"Balanced" means relative to a reference, often ground, and as respects impedance. There's nothing that's not balanced about a transformerless output if done right. A transformer can be isolated from ground, but it's still balanced with respect to the reference.
Yes, no galvanic isolation, but also (if you're not using Jensen or Lundahls), no transformer grunge either. There was a time when "transformerless" was a huge positive thing.
Can someone please explain to me why 3V out from a preamp is some sort of limit for an F5m input. I can crank my Iron Pumpkin preamp up to max volume at 8db gain setting into my F5m amp, then feeding into 4ohm 500W Aerial Acoustic Model 9's and I don't hear a bit of distortion, only the sound of my windows rattling.
Is there even a real "problem" to be solved here? Seems like 3V into the F5m is no limit at all in my experience.
I cranked up the volume to max and I'm getting clear and crisp audio that shakes my house and makes my wife frown and yell at me to "turn it down". The F5m loves it though.... my speakers seem totally un-phased by the level and I'm not playing gentle classical tunes, I'm playing Muete, Live in Paris bumpin' tunes. My stock built F5m kicks some serious butt at 25W output into 500W capable speakers. It sounds awesome!!!!
Makes me want to get up and dance 🕺
I love this amp! No limiters required.
Is there even a real "problem" to be solved here? Seems like 3V into the F5m is no limit at all in my experience.
I cranked up the volume to max and I'm getting clear and crisp audio that shakes my house and makes my wife frown and yell at me to "turn it down". The F5m loves it though.... my speakers seem totally un-phased by the level and I'm not playing gentle classical tunes, I'm playing Muete, Live in Paris bumpin' tunes. My stock built F5m kicks some serious butt at 25W output into 500W capable speakers. It sounds awesome!!!!
Makes me want to get up and dance 🕺
I love this amp! No limiters required.
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Hi,
It's been explained (and calculated) two time in this thread... one by Jaddie ( with acurate math and approach) another one by myself ( more coarse approach and 'in the ballpark' figures: cause i'm lazy and prefer to put people on track rather than doing things in place of them... gives a man a fish...blablabla).
To quickly answer a third time: desk output voltage is 10 to 14 db HIGHER output than what the F5M can cope with, because desk reference level is circa 0dbu ( a tiny bit lower than pro level reference which is often +4dbu).
As most amplifier ( except power buffer like eg an F4) an amp have voltage gain between it's input and the ouput stage driving the loudspeaker.
From F5M documentation we know the amp can output circa 25w into 8ohm and have a voltage gain of circa 16,5db.
From Ohm's law we can define the max ouput voltage delivered to the loudspeakers: U: square root(PxR), square root (25W×8R)= 14,14V rms.
This is equivalent to circa +25dbu but after a gain of 16,5db , which means a max input of circa 9dbu rms ( 2,18V rms). Those are approximate values, refer to Jaddie's post if you want more accurate...
Desk is able to output at max +21,5dbu= circa 9V rms.
From there it's obvious the desk will saturate F5 input if a pad is not used to adapt max voltage to the range the amplifier input need.
The other factor you have to take into account is dynamic range of input signal: unprocessed 'natural' signal can have up to +20db peak over the reference level, 'processed' recordings are more in the 14db range of peak over reference level... So the amp will have to cope with potentially an higher dynamic range too ( than when you listen to a record).
Your final acoustical output level ( SPL) is more largely dominated by your loudspeaker's efficiency: the higher it is the less amplifier power you need.
If we take RoboDNA loudspeakers ( wharfedale Lyndon) their efficiency is given as 90dbspl/1w/1m.
Each time you double amplifier power you gain +3db ( from 1 to 2 watts=+3db, from 2 to 4=+3db, from 4 to 8watts=+3db, from 8 to 16= +3db, from 16 to 32watts= +3db).
RoboDna modified bias of his amp to 32w so we know the max ouput from his loudspeakers + amp at max power ouput will be = 90dbSpl + (3db+3db+3db+3db+3db) so 90 +15= 105dbSPL at one meter.
Now take a loudspeaker with an efficiency of 78dbSpl/1w/1m and able to cope 500w it will end up with 78dbspl+27db=105db.
Same acoustic level than the Lyndon with 32w amplifier...
Let's see a last example with a loudspeaker with efficiency of 108db/1w/1m. For it to ouput 105dbspl you'll need... 0,5Watts.
Conclusion: amp power doesn't mean anything without efficiency of loudspeakers used with... as well db without a clear reference added to them ( fs, u,V, SPL) are nothing more than a tool for comparison purpose. When you say max gain of 8db it doesn't mean anything. If you said max gain of +8dbu we know the reference ( 0dbu= 0,775v) and as such can define a known voltage of 1,95V rms...
One last thing, your loudspeakers are more threatned with a lower power amplifier than one which meets their max ouput power.
Why? Because a low power amplifier have more chance to clip than one with same max power that your loudspeaker max rating... or bigger!
It's amplifier clipping which destroy tweeters ( because clipped looks like a square wave, which shape is a halfway equivalent to DC...).
In fact if you want to be sure your amp never clip it should be more powerful than the loudspeaker it drive. Something like +6db more powerful.
Of course you'll have to be sure it never see it's max input level but a signal -6db lower.
For your 500w loudspeaker it would mean: 2kw power amp ( +3db to 1kw and another +3db to 2kw)... but which will never see higher input level than -6db of max input ( indeed ouputing 500w max).
This is the 'headroom' principle we talked about previously.
If you want to verify the math check previous message and links, you have all tools given to do ( maybe it miss an Ohm's law wheel but you can google it easily).
It's been explained (and calculated) two time in this thread... one by Jaddie ( with acurate math and approach) another one by myself ( more coarse approach and 'in the ballpark' figures: cause i'm lazy and prefer to put people on track rather than doing things in place of them... gives a man a fish...blablabla).
To quickly answer a third time: desk output voltage is 10 to 14 db HIGHER output than what the F5M can cope with, because desk reference level is circa 0dbu ( a tiny bit lower than pro level reference which is often +4dbu).
As most amplifier ( except power buffer like eg an F4) an amp have voltage gain between it's input and the ouput stage driving the loudspeaker.
From F5M documentation we know the amp can output circa 25w into 8ohm and have a voltage gain of circa 16,5db.
From Ohm's law we can define the max ouput voltage delivered to the loudspeakers: U: square root(PxR), square root (25W×8R)= 14,14V rms.
This is equivalent to circa +25dbu but after a gain of 16,5db , which means a max input of circa 9dbu rms ( 2,18V rms). Those are approximate values, refer to Jaddie's post if you want more accurate...
Desk is able to output at max +21,5dbu= circa 9V rms.
From there it's obvious the desk will saturate F5 input if a pad is not used to adapt max voltage to the range the amplifier input need.
The other factor you have to take into account is dynamic range of input signal: unprocessed 'natural' signal can have up to +20db peak over the reference level, 'processed' recordings are more in the 14db range of peak over reference level... So the amp will have to cope with potentially an higher dynamic range too ( than when you listen to a record).
Your final acoustical output level ( SPL) is more largely dominated by your loudspeaker's efficiency: the higher it is the less amplifier power you need.
If we take RoboDNA loudspeakers ( wharfedale Lyndon) their efficiency is given as 90dbspl/1w/1m.
Each time you double amplifier power you gain +3db ( from 1 to 2 watts=+3db, from 2 to 4=+3db, from 4 to 8watts=+3db, from 8 to 16= +3db, from 16 to 32watts= +3db).
RoboDna modified bias of his amp to 32w so we know the max ouput from his loudspeakers + amp at max power ouput will be = 90dbSpl + (3db+3db+3db+3db+3db) so 90 +15= 105dbSPL at one meter.
Now take a loudspeaker with an efficiency of 78dbSpl/1w/1m and able to cope 500w it will end up with 78dbspl+27db=105db.
Same acoustic level than the Lyndon with 32w amplifier...
Let's see a last example with a loudspeaker with efficiency of 108db/1w/1m. For it to ouput 105dbspl you'll need... 0,5Watts.
Conclusion: amp power doesn't mean anything without efficiency of loudspeakers used with... as well db without a clear reference added to them ( fs, u,V, SPL) are nothing more than a tool for comparison purpose. When you say max gain of 8db it doesn't mean anything. If you said max gain of +8dbu we know the reference ( 0dbu= 0,775v) and as such can define a known voltage of 1,95V rms...
One last thing, your loudspeakers are more threatned with a lower power amplifier than one which meets their max ouput power.
Why? Because a low power amplifier have more chance to clip than one with same max power that your loudspeaker max rating... or bigger!
It's amplifier clipping which destroy tweeters ( because clipped looks like a square wave, which shape is a halfway equivalent to DC...).
In fact if you want to be sure your amp never clip it should be more powerful than the loudspeaker it drive. Something like +6db more powerful.
Of course you'll have to be sure it never see it's max input level but a signal -6db lower.
For your 500w loudspeaker it would mean: 2kw power amp ( +3db to 1kw and another +3db to 2kw)... but which will never see higher input level than -6db of max input ( indeed ouputing 500w max).
This is the 'headroom' principle we talked about previously.
If you want to verify the math check previous message and links, you have all tools given to do ( maybe it miss an Ohm's law wheel but you can google it easily).
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Update: I'm still processing all this great info. but it may take me a bit to catch up and report back since I"m doing this on my breaks/spare time 🙂 I'm looking more closely at a few items mentioned here... thanks again.
@krivium the docs show 35watts when the amp's mosfets are at 1.1amp to 1.4amp gain. I made a mistake when I said 25watt... sorry if that is where you got it from.
@krivium the docs show 35watts when the amp's mosfets are at 1.1amp to 1.4amp gain. I made a mistake when I said 25watt... sorry if that is where you got it from.
There is no issue at all! If you understood the mains points that is all that mater imho. It's not a race, everyone is different and we don't all proceed things at same rate, but it doesn't mater in the end how long it took...
As i said i'm a bit sketchy about math and won't pretend to be an EE... in the end with help from great people in here if you want the accurate math, you'll manage to have it explained ( at first you could read every article the great Nelson Pass wrote, manual of F5, the zen series of article,... it's all in there). And guy like Jaddie (or Pano which wrote the article about gain structure and is a moderator in here) and other members participating this thread ( and others members too) are always ready to help.
Diyaudio is a great place, i learned a lot from here and still do... and when i can i give back. That's the point: share what you know ( or think you do!) and learn from others. Such a good model of human interactions! Refreshing in time we live and ...it often bring back hope in humans kind to me. 😉
As i said i'm a bit sketchy about math and won't pretend to be an EE... in the end with help from great people in here if you want the accurate math, you'll manage to have it explained ( at first you could read every article the great Nelson Pass wrote, manual of F5, the zen series of article,... it's all in there). And guy like Jaddie (or Pano which wrote the article about gain structure and is a moderator in here) and other members participating this thread ( and others members too) are always ready to help.
Diyaudio is a great place, i learned a lot from here and still do... and when i can i give back. That's the point: share what you know ( or think you do!) and learn from others. Such a good model of human interactions! Refreshing in time we live and ...it often bring back hope in humans kind to me. 😉
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The calculations are correct for the data given. If you want to change the output power to 35 watts, fine, but that's all of a 1.4dB difference.
Here's why this isn't worth too much attention. The power rating is based on a resistive load, not reactive. All speakers are reactive. It's also based on an impedence of 8 ohms. No speaker is 8 ohms across the audio spectrum, they change, often radically, from 2 to above 16 ohms. And the calculations are based on a sine wave. Listen to tones much? Me neither. Music is different, with a different crest factor.
The concern is the mixer is referenced to a "Pro Audio" standard level of +4dBu, and the amp is meant to work in a consumer audio environment, so it will at some point be over-driven by the mixer, before the mixer overloads. Ideally it would be peachy if all the devices clip/overload points lined up, or you do pay a noise and headroom price. Ideally, but not mandatory.
The pad suggested makes the interface closer to "ideal".
You can't measure output power or distortion just by listening, though.
Here's why this isn't worth too much attention. The power rating is based on a resistive load, not reactive. All speakers are reactive. It's also based on an impedence of 8 ohms. No speaker is 8 ohms across the audio spectrum, they change, often radically, from 2 to above 16 ohms. And the calculations are based on a sine wave. Listen to tones much? Me neither. Music is different, with a different crest factor.
The concern is the mixer is referenced to a "Pro Audio" standard level of +4dBu, and the amp is meant to work in a consumer audio environment, so it will at some point be over-driven by the mixer, before the mixer overloads. Ideally it would be peachy if all the devices clip/overload points lined up, or you do pay a noise and headroom price. Ideally, but not mandatory.
The pad suggested makes the interface closer to "ideal".
You can't measure output power or distortion just by listening, though.
^^ that all makes sense to me... consumer audio = -10dBV = 0.316Vrms and pro audio = +4dBu = 1.23Vrms. I have not gotten around to check output voltage yet, but I'm expect the mixer's VU Meter is calibrated to +4dBu when master fader is at 0dB.
@birdbox Could it be your iron pumkin pre-amp is consumer level output? I'm not sure but let's say it's +20dB, then the input into the F5m would be 3.16Vrms which is pretty hot from my ( possibly wrong ) calculations. This would explain the party-levels your were getting no?
@birdbox Could it be your iron pumkin pre-amp is consumer level output? I'm not sure but let's say it's +20dB, then the input into the F5m would be 3.16Vrms which is pretty hot from my ( possibly wrong ) calculations. This would explain the party-levels your were getting no?
There would be no reason on earth it should be anything else. The specs state the output level. You don't need to measure, but you could if you want.
His issue:
1. doesn't seem to be a problem
2. isn't described with accurate objective data - subjective only
3. should probably be ignored unless it actually is a problem that needs to be solved. In other words, he seems happy with what he has, asked a question based on subjective observations, so we really can't address much.
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@jaddie Just to make sure I understand the faders; If 'in theory' the input on a channel is at 1Vrms with the gain at 0, and both the channel and master faders at full ( +10dB ), the output will be 1Vrms? Or, are you referring to an internal amplifier in the mixer which brings the level back up after all the faders?
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No it will be at 1v+20db ( +10++10). ( in other words it will saturate the console output as 1v= +2dbu)...
When both faders are at 0db marks, what you have post input gain trim is equal to what will be sent out of console (if you don't boost with eq). This is why it's commonly called 'unity gain' despite it sees a number of attenuation and boost inside the console... and as such it's not technically an UG.
When both faders are at 0db marks, what you have post input gain trim is equal to what will be sent out of console (if you don't boost with eq). This is why it's commonly called 'unity gain' despite it sees a number of attenuation and boost inside the console... and as such it's not technically an UG.
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To put it another way, there is no such thing as "unity gain" inside an analog console. There can't be. A fader is a loss-only device, with minimum loss at the top of its travel. The "0" point of a fader is typically 10dB of loss, which then needs to be made up for with a 10dB gain amplifier. If you have a fader followed by a master they'll both have 10dB of loss for a total of -20dB, which needs to be made up for by amplifiers.
If you look at the gain chart I posted, unity gain would be a straight horizontal line at zero. Nothing like that there.
Furthermore, there is no need for unity gain because sources vary, and in your case, the destination (the power amp) has different level requirements.
The concept of unity gain can be abandon, both as an advantage and a desire, as difficult as it is to accept.
^^ I was wondering if the mixer has built-in amplifiers after the faders, or if you are referring to the external F5m amp.
I was able to use a digital meter fast enough to detect AC at 1kHz and 100Hz test sine waves which I played FROM YOUTUBE. The focusrite scarlett 2i2 outputs 1.7Vrms for th 1kHz and 1.5Vrms for 100Hz. When I generated the 1kHz and 100Hz waves from LogicProX I get over 4Vrms with all Logic's faders to -12dBFS. It seems playing from the browser ( mac ) lowers to voltage quite a bit which explains my lower volume levels. I"m thinking maybe using Spotify or other media player would help. ( would be nice to be able to play youtube at a decent level though ).
I'll connect to the output of my mixer next to see how the VU meter/faders are calibrated.
I was able to use a digital meter fast enough to detect AC at 1kHz and 100Hz test sine waves which I played FROM YOUTUBE. The focusrite scarlett 2i2 outputs 1.7Vrms for th 1kHz and 1.5Vrms for 100Hz. When I generated the 1kHz and 100Hz waves from LogicProX I get over 4Vrms with all Logic's faders to -12dBFS. It seems playing from the browser ( mac ) lowers to voltage quite a bit which explains my lower volume levels. I"m thinking maybe using Spotify or other media player would help. ( would be nice to be able to play youtube at a decent level though ).
I'll connect to the output of my mixer next to see how the VU meter/faders are calibrated.
Jaddie is refering to amplifier within the console, not your power amp.
It's a bit difficult to apprehend at first but there is many ways to amplify a signal: you can amplify voltage, current or a mix of both.
In a mixer you rarely needs to have power amplifier, mainly you'll need voltage amplifier because loads seen by those amplifier are high impedance ( relatively). Those tasks are done easily by an integrated opamp as the njm4580 we talked about previously. Their overall design is not really different than a power amplifier ( it's an over simplification but it's not wrong most of the time) but the ability of their ouput transistors to deliver current are tiny ( wrt a power amplifier) as the load they will see are relatively 'light' ( high impedance). We ask them mainly to do voltage amplification ( and a bit of current but nothing to compare with a power amp).
For a power amplifier ( for loudspeaker) the load seen is low impedance, as such you need a lot of current to drive them. As such we will need some components with high current capability, like the irfp240/9240 you probably have as ouput stage.
It all comes down to Ohm's law: U=RI.
Let's see your console output stage: p31.
Locate ic502a/b. Those are the opamp driving the outside world.
Both are connected as inversing amplifier ( the input is done through the negative input node). In this case we know the gain is equal to r506/r900. 62k/20k: 3.
To know the gain in db you have to do 20Log(3)= 9db.
So that last stage have a gain of circa 9db. This is more or less in line with what you see with the marking on your fader: there is 10db gain availlable past the 0 mark...
Now how much current is it able to output can be seen in the njm datasheet: take a look page 3 at graph named 'maximum ouput voltage vs load resistance'.
You'll see that below approx 1k ohms the both curve have knees which means it can't gives enough current to be linear. Above 2k those curves looks like straight lines and as such are linear...
Hence my previous question to Jaddie regarding his pad input impedance: if it had been below 1k the amplifier would have had issues to drive it. 😉
Luckily, line are known as bridging connections: you need low output impedance driving high input impedance. As such you don't need power but a voltage. Most of the time input impedance of line gear is 10kohms or more. Use ohm law to see how much current is needed at 21,5dbu to drive a 10kohms input. Then try with 8r input impedance... it'll be obvious where the issue is.
It's a bit difficult to apprehend at first but there is many ways to amplify a signal: you can amplify voltage, current or a mix of both.
In a mixer you rarely needs to have power amplifier, mainly you'll need voltage amplifier because loads seen by those amplifier are high impedance ( relatively). Those tasks are done easily by an integrated opamp as the njm4580 we talked about previously. Their overall design is not really different than a power amplifier ( it's an over simplification but it's not wrong most of the time) but the ability of their ouput transistors to deliver current are tiny ( wrt a power amplifier) as the load they will see are relatively 'light' ( high impedance). We ask them mainly to do voltage amplification ( and a bit of current but nothing to compare with a power amp).
For a power amplifier ( for loudspeaker) the load seen is low impedance, as such you need a lot of current to drive them. As such we will need some components with high current capability, like the irfp240/9240 you probably have as ouput stage.
It all comes down to Ohm's law: U=RI.
Let's see your console output stage: p31.
Locate ic502a/b. Those are the opamp driving the outside world.
Both are connected as inversing amplifier ( the input is done through the negative input node). In this case we know the gain is equal to r506/r900. 62k/20k: 3.
To know the gain in db you have to do 20Log(3)= 9db.
So that last stage have a gain of circa 9db. This is more or less in line with what you see with the marking on your fader: there is 10db gain availlable past the 0 mark...
Now how much current is it able to output can be seen in the njm datasheet: take a look page 3 at graph named 'maximum ouput voltage vs load resistance'.
You'll see that below approx 1k ohms the both curve have knees which means it can't gives enough current to be linear. Above 2k those curves looks like straight lines and as such are linear...
Hence my previous question to Jaddie regarding his pad input impedance: if it had been below 1k the amplifier would have had issues to drive it. 😉
Luckily, line are known as bridging connections: you need low output impedance driving high input impedance. As such you don't need power but a voltage. Most of the time input impedance of line gear is 10kohms or more. Use ohm law to see how much current is needed at 21,5dbu to drive a 10kohms input. Then try with 8r input impedance... it'll be obvious where the issue is.
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What level were the generated tones? Not the output, but the tone level in dBFS? You have to be VERY specific and careful about this.
Actually, you can hear clipping distortion using a piezo tweeter driven with a ~400Hz sine wave.
https://www.prosoundtraining.com/20...ccurate-gain-structure-using-a-common-device/
The piezo won't respond to the low frequency tone, but clipping creates high frequency harmonics that can be heard easily.
The voltage produced by the device at the clipping level can be measured.
To measure nominal power, square the voltage and divide by the resistance, for example, 14Vx14V=196/8ohms=24.5 watts.
https://www.prosoundtraining.com/20...ccurate-gain-structure-using-a-common-device/
Langston Holland:
"You can hear the change due to clipping about 1/4 dB after you first spot flattening on a scope. The RS model # for it is 273-0074 and you can order it online. I'd get several because there are huge manufacturing tolerances on these things, thus some will squeal louder than others at the onset of clipping harmonics."
Art
A sine wave is a single frequency with no harmonics. Some harmonics are added by the nonlinearities in speakers (actually a lot), but clipping shoots the harmonic content way up very quickly. A sine wave does not mask even 3rd haromic well, and with hard clipping, harmonics go on forever...well, not really, but way up to 9th and more.
Music is very, very different. First, it's not a constant level, so you can't ever clip continuously. Music also is not a single frequency, it always has a dense spectrum capable of masking distortion products. Distortion audibility is also time dependant, the longer it exists, the more audible. 1ms of distortion is hard to hear. 1 second, very easy. Music transients are short, filled with spectrum, and so clipping distortion on music is far more difficult to hear than a continuous sine wave.
Notice what birdbox posted:
And youtube perform 'leveling' so it's in no way accurate.
If you want to calibrate your console to your soundcard Robodna, generate a -18 or -20dbfs sine at 1khz into logic and adjust input gain until you read on console meter (+2 for -18dbfs) or (0 for -20dbfs).