the effect of attenuation is the loss of intensity of signal.
loss. it goes away.
attenuation is a lossy process. information in the signal is literally absorbed and radiated away as heat. it cannot be reconstructed because it's gone.
it isn't an analog process that theoretically preserves the signal, it's an analog process that explicitly destroys a component of the signal.
but what if it wasn't...
okay, lets assume for a second that you have a signal with the same harmonic content as one of my previous examples, a high peak when viewed on a frequency spectrum chart, a low peak when viewed on that chart and everything else. these three parts of the signal represent the loud, quiet and "silent" parts of the signal respectively. unlike the previous example we'll let our noise floor for the silent parts be infinitely low. for now. so you start hooking up your perfect amps and pots in line and setting them all to 1% or so and listening. it's sounding pretty good at first, but once you get a few deep, you start getting white noise and clicks and pops and all kinds of craziness.
what the hell! all this equipment is theoretically perfect, why is there noise? it can't be coming from the perfect equipment!
it's not. it's coming from the medium. in our theoretical example all these amplifiers and pots are hooked up with conductive wire. the signal has to propagate through that wire from component to component. atoms of copper are being excited and losing their excitation in proportion to the signal. their state of excitation is being amplified over and over again. the noise is in the wire. by amplifying it over and over again you made it audible. you can't ever escape it. signed, listening to noise gang. come to my modular synth show.
okay, so now that the possibility of ever attenuating a signal without losing information is hopefully put to rest, lets turn to the digital attenuation of the signal in comparison.
level attenuation over the digital domain is also a lossy process. what's being misunderstood here is that the levels aren't being shrunk relative to each other, they're each being divided and the signal that's reconstructed by the DAC no longer contains the quiet parts.
just like those quiet parts were absorbed and radiated as heat by the resistor, the digital version of attenuation does away with the need for all that physics crap and simply deletes them from the stream.
if the levels were being shrunk relative to each other, you'd be compressing the signal like when you use the bitcrusher pedal for your guitar and there would be lots of harmonic distortion. but attenuation and compression are different processes and have significantly different results.
consider a quiet sound, your 1/64th volume signal. a sine wave. its encoded to represent 1/64th of the maximum level of the adc's input because when it was recorded, it represented 1/64th the maximum level of the preamp/microphone/whatever that was plugged into the adc.
is the quiet sine wave of lower quality than one that's using the full bit depth of the adcs output because it's intended to represent the maximum level that the adcs input saw from the preamp/microphone/whatever?
of course it isn't. it just wasn't loud.
and if your loud sine wave was electrically generated by a theoretical perfect function generator which contains no distortion or other sonic content before being sent to the adc, would it be more damaged if it's amplitude were divided by 64 before being decoded or if it were decoded and sent through a resistor whose value was chosen specifically to dissipate 63/64ths of it as heat in order to make it as quiet as the quiet sine wave?
of course it wouldn't.
to your last question, let me rephrase it into something I can agree with: you cannot possibly attenuate and then amplify in any way and hope to get the same signal back. It’s a lossy process which destroys the signal by design.
Ok, the analog attenuation part makes sense now I think. I assumed that an amp increases the amplitude of a signal, and that a pot achieved the inverse (i guess dividing the signal?) but it's not, it's effectively subtraction?
Back to my DAC/amp, realistically am I ever intentionally attenuating the analog signal in order to get it to a listening volume? Or am I only ever amplifying it? I think that's the main difference in my head. If I output my digital signal to the DAC at 100%, and then only ever amplify it to a listening volume, then there's no way for the signal to be attenuated at all, right?
is the quiet sine wave of lower quality than one that's using the full bit depth of the adcs output because it's intended to represent the maximum level that the adcs input saw from the preamp/microphone/whatever?
of course it isn't. it just wasn't loud.
No, yeah, that makes sense. I was thinking that, it didn't matter how strong the signal was, as long as the full sine wave was still present, then quality is preserved. So dividing it down to be a very small voltage, or amplifying it up to be super large, as long as the relative voltages of the signal are retained, we wouldn't lose any quality (is my, likely flawed, impression).
I think I just don't know how to think of analog signals. I understand frequency response in theory, but I can't talk about a signal intuitively in terms of frequency space like you do. Does perfectly amplifying a signal change its frequency response? You don't have to keep answering, at this point I'm just poking your brain lol.
if you use the volume control on your amp, you're attenuating the signal. that's assuming that the amplifier uses a pot as a voltage divider on the input rather than a gain control for the amplifier IC it uses. both are common, but if it's used as a gain control you're relying on the linearity of the pot to inform the gain of the amplifier IC and generally frequency linearity across wide gain ranges aren't fantastic which is why old stuff that used a bunch of discrete transistors instead of chips had set gain and varied the input signal level with an attenuating pot acting as a voltage divider. there are counterexamples. this is complicated.
if you don't use the volume control on the amp then there's no attenuation. the downside is that it's really loud. too loud for safety. too loud for comfort. sometimes too loud for the sustained operation of the equipment.
perfectly amplifying a signal does change its frequency response by adding noise picked up in the process of conveying that signal to and from the perfect amplifier. the noise is added because it's amplified. there's some ways to lessen the effect of this, some are effective enough to allow absolutely miniscule signals like those produced by a moving coil phono cartridge to be amplified to tremendous volumes. but there is always noise.
all this talk about amplifiers and volume control is wonderful, but have you considered the stuff that comes after the amplifier? your headphones are a great example: if you pan over to the right channel so that only the right headphone is making sound and slowly pull the headphone away from your ear you'll hear the frequency response change. the lowest notes will go away, then higher and higher pitched notes will become inaudible until all you hear are tinny high frequency sounds. that's the effect of attenuation of the sound wave propagating through the air. naturally in air, that attenuation acts as a high pass filter, reducing the volume of the lower pitched notes more than higher pitched notes.
if you pull it away in time with the beat of the music it acts like a bpm synced filter effect or a wah pedal (depending on your age).
the distance from the driver to your ear has an effect on the sound you hear. is that distance correct? everyones heads are a little different...
how about the ears themselves, are they clean? it's gross to think of, but wax acts just like a pair of earplugs.
and what about your own brain. i'm not even close to versed in psychoacoustics but just imagine agent smith from the matrix saying "you think that's music you're listening to?"
there's nothing wrong with wanting good quality equipment and for it to perform as close to perfect as possible but at some point we're picking only the most perfectly ripe, unblemished, peak season tomatoes to make salsa.
if you don't use the volume control on the amp then there's no attenuation. the downside is that it's really loud.
Ahhh, I think this is the part I was missing. So I should think of an amp as "injecting" a fixed amount of energy to the signal, way more than I need, and then the volume pot attenuates it back to a comfortable volume. That makes sense since we've established that pots attenuate, which necessarily destroy the signal. I still had it in my head that the amount of energy used to amplify the signal was proportional to the volume knob position.
I didn't know how air affected frequency response, but that makes sense.
As for how clean my ears are, I'm completely deaf, so who cares?
jk 😁. Thanks for the talk, I learned some things!
No problem. It’s nice to talk about this stuff. If you want the skinny on amplifiers:
There’s two parts: the volume control and the amplifier. The volume control sits in between the input and the amplifier circuit itself. The amplifier circuit can be thought of as a fixed multiplier of whatever signal goes in. If there’s x50 gain then it’ll make an input of 20 into 1000 arbitrary units(tm). An input of 2 would be made into an output if 100au and you choose between those two inputs or any number you like with the volume control, acting as an attenuator before the amplifier starts multiplying the signal.
To look a little closer, the amplifier takes a big dc voltage and modulates it based on the small input voltage. If it was a tube amplifier, all the dc voltage gets put on the plate of the tube, the input is sent to the grid (a literal grid of wire in between the plate and cathode) and electrons jump through the grid to the cathode in proportion to the grid current. If the grid current is a song, then the massive amounts of electrons gathered at the high voltage plate will jump across to the cathode in proportion to it and if an enterprising person were to put a transformer and speaker in between the cathode and their path to ground those electrons could be used to move it back and forth!
Of course, they only do that in a vacuum and when heated up, so all that takes place inside an evacuated glass tube with those parts crammed into it next to a light bulb filament to make them toasty.
Point is: you’re not even getting the same electrons!
Your solid state amplifiers are doing the same thing but with transistors instead of vacuum tubes. Feed the input into the base of the right kind of transistor and it’ll let more or less voltage move in the direction of its arrow.
You get a few of the input electrons with a transistor, but it’s mostly electrons from the big dc voltage.
Ahh ok, that makes more sense. I think I never saw the connection between a transistor and an amp until now. Using a small signal to modulate a different, larger signal. Or like a relay.
the effect of attenuation is the loss of intensity of signal.
loss. it goes away.
attenuation is a lossy process. information in the signal is literally absorbed and radiated away as heat. it cannot be reconstructed because it's gone.
it isn't an analog process that theoretically preserves the signal, it's an analog process that explicitly destroys a component of the signal.
but what if it wasn't...
okay, lets assume for a second that you have a signal with the same harmonic content as one of my previous examples, a high peak when viewed on a frequency spectrum chart, a low peak when viewed on that chart and everything else. these three parts of the signal represent the loud, quiet and "silent" parts of the signal respectively. unlike the previous example we'll let our noise floor for the silent parts be infinitely low. for now. so you start hooking up your perfect amps and pots in line and setting them all to 1% or so and listening. it's sounding pretty good at first, but once you get a few deep, you start getting white noise and clicks and pops and all kinds of craziness.
what the hell! all this equipment is theoretically perfect, why is there noise? it can't be coming from the perfect equipment!
it's not. it's coming from the medium. in our theoretical example all these amplifiers and pots are hooked up with conductive wire. the signal has to propagate through that wire from component to component. atoms of copper are being excited and losing their excitation in proportion to the signal. their state of excitation is being amplified over and over again. the noise is in the wire. by amplifying it over and over again you made it audible. you can't ever escape it. signed, listening to noise gang. come to my modular synth show.
okay, so now that the possibility of ever attenuating a signal without losing information is hopefully put to rest, lets turn to the digital attenuation of the signal in comparison.
level attenuation over the digital domain is also a lossy process. what's being misunderstood here is that the levels aren't being shrunk relative to each other, they're each being divided and the signal that's reconstructed by the DAC no longer contains the quiet parts.
just like those quiet parts were absorbed and radiated as heat by the resistor, the digital version of attenuation does away with the need for all that physics crap and simply deletes them from the stream.
if the levels were being shrunk relative to each other, you'd be compressing the signal like when you use the bitcrusher pedal for your guitar and there would be lots of harmonic distortion. but attenuation and compression are different processes and have significantly different results.
consider a quiet sound, your 1/64th volume signal. a sine wave. its encoded to represent 1/64th of the maximum level of the adc's input because when it was recorded, it represented 1/64th the maximum level of the preamp/microphone/whatever that was plugged into the adc.
is the quiet sine wave of lower quality than one that's using the full bit depth of the adcs output because it's intended to represent the maximum level that the adcs input saw from the preamp/microphone/whatever?
of course it isn't. it just wasn't loud.
and if your loud sine wave was electrically generated by a theoretical perfect function generator which contains no distortion or other sonic content before being sent to the adc, would it be more damaged if it's amplitude were divided by 64 before being decoded or if it were decoded and sent through a resistor whose value was chosen specifically to dissipate 63/64ths of it as heat in order to make it as quiet as the quiet sine wave?
of course it wouldn't.
to your last question, let me rephrase it into something I can agree with: you cannot possibly attenuate and then amplify in any way and hope to get the same signal back. It’s a lossy process which destroys the signal by design.
Ok, the analog attenuation part makes sense now I think. I assumed that an amp increases the amplitude of a signal, and that a pot achieved the inverse (i guess dividing the signal?) but it's not, it's effectively subtraction?
Back to my DAC/amp, realistically am I ever intentionally attenuating the analog signal in order to get it to a listening volume? Or am I only ever amplifying it? I think that's the main difference in my head. If I output my digital signal to the DAC at 100%, and then only ever amplify it to a listening volume, then there's no way for the signal to be attenuated at all, right?
No, yeah, that makes sense. I was thinking that, it didn't matter how strong the signal was, as long as the full sine wave was still present, then quality is preserved. So dividing it down to be a very small voltage, or amplifying it up to be super large, as long as the relative voltages of the signal are retained, we wouldn't lose any quality (is my, likely flawed, impression).
I think I just don't know how to think of analog signals. I understand frequency response in theory, but I can't talk about a signal intuitively in terms of frequency space like you do. Does perfectly amplifying a signal change its frequency response? You don't have to keep answering, at this point I'm just poking your brain lol.
if you use the volume control on your amp, you're attenuating the signal. that's assuming that the amplifier uses a pot as a voltage divider on the input rather than a gain control for the amplifier IC it uses. both are common, but if it's used as a gain control you're relying on the linearity of the pot to inform the gain of the amplifier IC and generally frequency linearity across wide gain ranges aren't fantastic which is why old stuff that used a bunch of discrete transistors instead of chips had set gain and varied the input signal level with an attenuating pot acting as a voltage divider. there are counterexamples. this is complicated.
if you don't use the volume control on the amp then there's no attenuation. the downside is that it's really loud. too loud for safety. too loud for comfort. sometimes too loud for the sustained operation of the equipment.
perfectly amplifying a signal does change its frequency response by adding noise picked up in the process of conveying that signal to and from the perfect amplifier. the noise is added because it's amplified. there's some ways to lessen the effect of this, some are effective enough to allow absolutely miniscule signals like those produced by a moving coil phono cartridge to be amplified to tremendous volumes. but there is always noise.
all this talk about amplifiers and volume control is wonderful, but have you considered the stuff that comes after the amplifier? your headphones are a great example: if you pan over to the right channel so that only the right headphone is making sound and slowly pull the headphone away from your ear you'll hear the frequency response change. the lowest notes will go away, then higher and higher pitched notes will become inaudible until all you hear are tinny high frequency sounds. that's the effect of attenuation of the sound wave propagating through the air. naturally in air, that attenuation acts as a high pass filter, reducing the volume of the lower pitched notes more than higher pitched notes.
if you pull it away in time with the beat of the music it acts like a bpm synced filter effect or a wah pedal (depending on your age).
the distance from the driver to your ear has an effect on the sound you hear. is that distance correct? everyones heads are a little different...
how about the ears themselves, are they clean? it's gross to think of, but wax acts just like a pair of earplugs.
and what about your own brain. i'm not even close to versed in psychoacoustics but just imagine agent smith from the matrix saying "you think that's music you're listening to?"
there's nothing wrong with wanting good quality equipment and for it to perform as close to perfect as possible but at some point we're picking only the most perfectly ripe, unblemished, peak season tomatoes to make salsa.
Ahhh, I think this is the part I was missing. So I should think of an amp as "injecting" a fixed amount of energy to the signal, way more than I need, and then the volume pot attenuates it back to a comfortable volume. That makes sense since we've established that pots attenuate, which necessarily destroy the signal. I still had it in my head that the amount of energy used to amplify the signal was proportional to the volume knob position.
I didn't know how air affected frequency response, but that makes sense.
As for how clean my ears are, I'm completely deaf, so who cares?
jk 😁. Thanks for the talk, I learned some things!
No problem. It’s nice to talk about this stuff. If you want the skinny on amplifiers:
There’s two parts: the volume control and the amplifier. The volume control sits in between the input and the amplifier circuit itself. The amplifier circuit can be thought of as a fixed multiplier of whatever signal goes in. If there’s x50 gain then it’ll make an input of 20 into 1000 arbitrary units(tm). An input of 2 would be made into an output if 100au and you choose between those two inputs or any number you like with the volume control, acting as an attenuator before the amplifier starts multiplying the signal.
To look a little closer, the amplifier takes a big dc voltage and modulates it based on the small input voltage. If it was a tube amplifier, all the dc voltage gets put on the plate of the tube, the input is sent to the grid (a literal grid of wire in between the plate and cathode) and electrons jump through the grid to the cathode in proportion to the grid current. If the grid current is a song, then the massive amounts of electrons gathered at the high voltage plate will jump across to the cathode in proportion to it and if an enterprising person were to put a transformer and speaker in between the cathode and their path to ground those electrons could be used to move it back and forth!
Of course, they only do that in a vacuum and when heated up, so all that takes place inside an evacuated glass tube with those parts crammed into it next to a light bulb filament to make them toasty.
Point is: you’re not even getting the same electrons!
Your solid state amplifiers are doing the same thing but with transistors instead of vacuum tubes. Feed the input into the base of the right kind of transistor and it’ll let more or less voltage move in the direction of its arrow.
You get a few of the input electrons with a transistor, but it’s mostly electrons from the big dc voltage.
Ahh ok, that makes more sense. I think I never saw the connection between a transistor and an amp until now. Using a small signal to modulate a different, larger signal. Or like a relay.