Is an analogue audio signal DC or AC?

I always thought that a standard analogue audio signal, such as the one that travels along a speaker wire was an AC current. This seemed to make sense to me as the current travelling along the wire must continually change to induce movement in the speaker cone, but after looking a wikipedia I’m confused.

The article on ACsays that audio signals are AC

However the article on DC seems to suggest that they are DC.

The second quote mentions constant polarity, which I assume means that the current doesn’t change direction as with AC and I can see how this would work. The current doesn’t have to change polarity to make the speaker move, it just has to vary over time, so I can see how it could be either.

Is the AC article wrong? Are some audio signals AC and some DC? Or is my understanding of this whole thing too simplistic?

Pure AC is a time varying electrical signal which periodically reverses it’s polarity, crossing zero twice per cycle. Pure DC is a constant, unvarying signal. Fluctuating DC is more appropriately termed AC with a DC bias; if you remove the DC bias (by passing the signal through a capacitor, for example), you are left with a pure AC signal. Thus, an analog audio signal can be either pure AC or AC with a DC bias.

Sounds right to me: the way to send an electric signal (digital or analogue) is to vary the potential between the two wires. It doesn’t matter if one wire is always negative compared to the other (say, one wire stays at 0 and the other fluctuates between 0 and 1), or if the potential is reversed (one wire stays at 0 and the other between 0.5 and -0.5). At least not in theory, but there are probably practical considerations to prefer a relatively low average potential.

The speaker perspective is a good one. You are correct that the alternating current moves the cone in and out. An AC signal with a DC component would also move the cone in and out, but the DC portion would permanently move the cone a bit in one direction, making a new zero point, as it were. As you might imagine, this would affect the sound quality.

DC components can cause other problems. Some uses of scrambling in transmitting a digital signal over radio waves are there to remove DC components. There’s no such thing as a DC offset of a radio wave, but the DC component might appear at the receiver or transmitter prior to sending or receiving the radio waves, so the scrambling function attempts to remove significant DC offsets for the amplifiers before the digital signal is recovered (at which point the DC offset is no longer a concern).

I quote a patent to explain this:

I didn’t read the whole patent but this point is interesting to think about.

DC is generally understood to be constant, and AC in most uses is generally understood to be a constant sine wave. Audio signals really aren’t either. Audio signals are varying sine waves in the frequency range of 20 Hz to 20,000 Hz. A constant AC signal (meaning of constant frequency) within that range would be called an audio frequency AC signal though. There’s enough weaseling within the common use of the terms that you could easily get away with calling audio signals AC. You just have to be careful of the context. “AC” to many folks means 120 volts AC 60 Hz. It’s more common to just refer to audio signals as “audio” instead of trying to call them AC or DC.

There’s a very important fundamental concept in signals that says any complex waveform can be broken down into a series of sine waves. Your ears hear in the frequency domain, so what you are really hearing is the frequencies of these individual sine waves. The cells in your ears (called hair cells) act like itty bitty bandpass filters, each focusing on a small range of sine waves. So, “sound” is really sine waves. You can’t hear DC. You can only hear sine waves. If you ignore the common usage of the terms, sine waves are by definition alternating, so yes, sound is technically AC. Like I said above, though, calling audio signals “AC” might confuse some people in some contexts.

Phone signals are a bit different than the audio signals used in your stereo. If you understand how phones work then the example they give really makes a lot of sense. However, if you don’t know the difference between AC and DC it’s pretty doubtful that you’ll know how a telephone works, so I think they lost the point on their target audience.

A telephone is really ingenious. People take it for granted but they don’t realize the creativity that went into its creation. In a telephone, you have audio signals coming in, you have audio signals going out, you have a ring signal, you have a way of detecting whether the phone is on or off hook, and you have a way of dialing. This is simple if you have half a dozen wires. The amazing thing is that they figured out how to do it using only two wires.

The telephone has a constant DC voltage applied to it. When you pick up the phone, the current draw from the switch being turned on signals the phone system that the phone is off hook. When the phone is on the hook, they can put an AC ring signal across the wires and the phone will ring. When the phone is off the hook, the voice coil modulates the current from the constant DC source, which transmits the audio signal from the phone to the switching station. The DC isn’t truly DC, though, since it also is modulated with the incoming audio signal. A very creative coil arrangement allows you to hear the person on the other end louder than you hear your own voice.

A few coils of wire and some switches, no modern electronics at all, and only two wires. Amazing.

Since you can think of the audio signal as modulating the current from a DC source, their example of a telephone makes sense as a varying DC signal.

That’s a product of the Fourier theorem, if anyone’s interested in the details.

The root of the confusion seems to be a difference between a common understanding of the term “DC” (a constant value of voltage or current) and an engineering use (constant polarity). In engineering terms, a full-wave rectified AC signal is DC, even though it’s varying between zero and peak at twice the frequency of the original signal.

If audio has a DC bias it was added by the electronic circuit because the original audio never has any DC component. Audio is pure AC with no DC bias. (BTW, my computer audio card adds a strong DC bias which I later have to remove. It stinks.)

If by “original audio” you mean the sound traveling through the air, it isn’t either of those things. Sound isn’t AC any more than light is electricity.

On some older computers the DAC chips would turn a digital value into a voltage that varied from 0 to 15 volts. This gave a nice sound playback. The common ground was also the reference for the analog output. Look at the schematic for a DAC chip if you can understand it. You programed a wave form table to make the sound you wanted and then changed the frequency you cycled through the table for the pitch. All wave tables used 0 volts as the bottom line of the graph.

The Apple II’s were pitiful in that they used digital on and off the speaker for tones. No wave table or volume control was programmable.

If you watch from the 6:10 point in this video , there is a interesting explanation to this question of DC or AC currents. This experiment is dealing with Iron, but it probably applies to some types of audio signals as well.

I think the OP is mostly correct in his understanding that audio signals are AC. More particularly in the case of the telephone I think this is AC with a big DC bias when it is generated by the carbon granule microphone they used to use. Modern electronic phones may preserve this for system compatibility - or, I’m not sure, they may no longer have any bias. Alot of these things depend on how telephone systems are evolving.

But I think the terms AC and DC are more often used to describe AC that is a sine wave at one frequency, and DC to describe constant voltage or perhaps voltage varying on a time scale people could observe.

There is a very convenient tradition of passing audio signals along through a capacitor that removes DC bias for various reasons. If there wasn’t a special reason to have a DC bias there, somebody would probably have done this capacitor treatment somewhere along the way, or multiple places along the way, in a circuit somewhere.

>There’s a very important fundamental concept in signals that says any complex waveform can be broken down into a series of sine waves.
AFAIK Fourier proposed his treatment only for waveforms that are inherently periodic. He thought first of the temperature around the circumference of an iron ring, half of which was hot and half cold, as the temperature profile evolved while the heat conducted (and the heat conduction equation has since been called Fourier’s law). I don’t think Fourier, or LaPlace or Lagrange (two of the French Academy of Science members most interested in Fourier’s work with sines), would have considered Fourier’s treatment appropriate for something like sound or an electronic analog of sound.
>Sound isn’t AC any more than light is electricity.
In fact it is much less so than light is electricity. Light is a radiating periodic variation in electric field strength. Sound is to air pressure as light is to electric fields. Sound hasn’t any particular relation to electonic phenomena until there are microphones and speakers in play.

Either you are being silly or you are looking for a silly argument.

An analog electrical audio signal is a faithful analog representation of audio (or of pressure if you prefer) and that original audio signal does not contain any constant but only alternating values. When faithfully converted to an analog electrical signal this signal does not contain any DC component but only AC. If the system adds a DC component then it can safely be discarded because it was not part of the original audio. Is that better?

One thing which puts me off participating in technical threads is people who are not even nitpicking but who are just playing silly games with language when the meaning is clear to all. That does not help clarify anything; it just muddies the water. Like those people who think they gotcha by saying “but America is not a democracy, it is a republic, so there!” :rolleyes:

No, I was trying to be gentle about it, but since you insist on pressing the issue, it was not a nitpick, it was a correction because your original statement was dead wrong.

sailor, I’m curious about this. Why do you remove it, and how?

Hijack …

Continuing on Napier’s pithy comment that “Sound is to air pressure as light is to electric fields”.

Sound is a variation of air pressure over time. That variation is applied to what we call ambient air pressure, e.g. the standard 14.7 psi / 1013 mB.

The ambient pressure is the analog of a DC bias voltage. For engineering convenience we don’t include that in typical audio appications, and use zero voltage as representing ambient pressure.

As engineer_comp_geek so eloquently explained, traditional (“POTS”) telephony operates as an audio signal variation on top of a DC bias voltage. So in that sense, telephony is a more pure analog of sound physics than ordinary audio electronics is.

An interesting digression … Considering some of the way-out musical composers & compositions … I read about a guy who’s written a piece that will take 100 years to play. A single note is sounded monthly or some such.

We can apply this idea to recordings made which faithfully record ambient pressure as well as the traditional variation. Pure Platonic silence recorded on perfect equipment over a span of months would include a very low tone (measured in microHz) representing the passage of high & low pressure weather systems through our studio. An Artiste might call it the music of Gaia or some such New Age fluff.

Conventional music recorded including the ambient pressure would replay differently depending on the ambient pressure difference between the recording and playback environments.

I’m not quite sure how it would “sound” different, and I’m sure the difference would be inaudible to humans, but it’s fun to think about.

Actually, that was my pithy comment and was put forth in just the spirit you expanded nicely upon.

No, it was not dead wrong, it was dead right. Except that you were twisting it to make it look like I was saying or implying that sound waves are electricity or something like that but those are your words, not mine. Because anybody with a modicum of intelligence will see is not what I said or meant. Since I never said or meant or implied that and since it cannot be understtod by anybody with half a brain that I meant or said or implied that, you are just being obnoxious. We all know full well that sound waves are not electricity and it is just disingenous to imply otherwise.

No, those are your words. More specifically, these are your words:

Whatever it was you meant to say (and note that I have no doubt you didn’t actually mean to imply sound is electrical), what you did say was wrong; AC and DC are strictly electrical terms and no amount of weaseling is going to change that fact. You can either accept that and move on or continue to be bullheaded about it. I really don’t care.

My sound card has this defect, I don’t know why (probably because it is crap and because it does not filter out the microphone bias voltage, which it should do).

A DC bias in digital audio files introduces a problem. Suppose the range of values goes from +127 to -127 with 0 being the center point. So your effective range is from +127 to -127. But now suppose your center, resting, point is +50. Now your range upwards is only from 50 to 127 (77) and though you have more range downwards you cannot use it because the original signal is (usually) going to be simmetrical in range so your new signal is going to be from +127 to 50 to -27 and you lose the rest of the range. So you really want to avoid any DC bias being superimposed on your signal and you want to make the signal go as close as posibble to 100% in range without going over. That way you get the greatest resolution and fidelity.

The audio program I use has a function to “remove DC bias” (or somesuch label) and another one to “normalize” (which amplifies so the highest peak goes to 100%). I suppose this is common in audio processing programs but I am no expert in this. Mine has an oscilloscope function where you can see the DC bias very clearly visible.