whats the pros/cons of running analog vs. digital.

B

businessjeff

Junior Audioholic
If I were to feed an amp a signal, and what would the pros/cons be of sending a digital signal vs an analog signal?

Lets say as well, if I had to run it a long distance or short. Is one going to get more interference/noise vs the other?

What actually causes a signal line to pick up noise, higher voltage, current?

How does a digital signal differ from an analog signal in terms of voltage and current?

Thanks.
 
TLS Guy

TLS Guy

Seriously, I have no life.
If I were to feed an amp a signal, and what would the pros/cons be of sending a digital signal vs an analog signal?

Lets say as well, if I had to run it a long distance or short. Is one going to get more interference/noise vs the other?

What actually causes a signal line to pick up noise, higher voltage, current?

How does a digital signal differ from an analog signal in terms of voltage and current?

Thanks.
First of all, amps don't usually have DACs so you can't send them a digital signal.

Now until recently, domestic equipment was almost exclusively interconnected with high impedance unbalanced connections. Now there are units that have low impedance balanced connections.

High impedance balanced lines can only transmit a signal fairly short distances because: -

1). the longer the line the more likely it is to pick up extraneous noise such as SCR noise.

2). Because the impedance is high, as cable length increases the capacitance of the cable becomes significant, so there is progressive high frequency loss as the cable length increases.

This problem was solved back at the dawn of audio, well before WW II.

The solution is the 600 ohm balanced line.

This is a cable with two conductors and a screen. In an AC wave form the positive half is on pin 2, the negative deflection on pin 3. Pins 2 and three therefore have potential to each other but not ground. Pin 1 is ground and carries no signal.

Now in a good balanced low impedance cable, the characteristics of the conductors connected to pins 2 and 3 are kept to high tolerances.

Now these conductors pick up identical interference. If the input stage, is perfectly balanced, be it dual op amp chip or transformer, the interference cancels.

The system works beautifully and has stood the test of time, a long time. Very long cables runs are possible, and all professional systems use this system.

You seem to want to advance in the hobby, but we can not possibly give you an adequate education on a forum like this. I strongly suggest you have some formal education and start with some reading.

Honestly, you have asked a very basic question which has been answered before on these forums, and yet you have posted about, designing and building amplifiers, active crossovers and putting together high powered pro systems for large venues with your designs.

Let me be honest you have not started a million part of your journey.
 
B

businessjeff

Junior Audioholic
First off I was meaning, receiver, not amp. And I dont think it was to hard to pick up on what I was meaning so.... my bad? Receiver has an amp.. idk its like saying car rather than vehicle to me.

Second, I read everyday on this stuff all day. My mind works differently than most, I look for an understanding of whats actually going on, not just a word that sums it up. I.E. theres an important difference between knowing something and understanding something, specially in this area. So thats why I ask these basic questions, but seem to get on occasion a broad answer rather that an explanation. Thats fine though, its not your job or anyone elses to teach me, I understand that, but this is where I look to learn, its free :)

Rather than just say, the longer a cable is, the more noise it will pick up, a reason is what im looking for. I already knew that about cable length and interference but I dont know whats going on really that causes this.

The cable you described are now referred to as XLR cables, but I dont see them being used in sending analog signals to and fro, besides with Microphones in musical applications. Why is this?

Thats why I wonder that actual difference between an Analog and Digital signal, in fact someone just wrote a post wonder the gains he would receive from sending a digital signal vs. an analog. With optical is obvious as to whats going on and why, however electricity is the medium for an analog and digital signal, so it stands to reason that they will both be have the same pitfalls and benefits.

This is the difference between knowing and understanding that I look for. I always appreciate any help I get, but please dont let your answer be, "Go learn if you want to know" that kind of logic doesn't even make sense.
 
TLS Guy

TLS Guy

Seriously, I have no life.
First off I was meaning, receiver, not amp. And I dont think it was to hard to pick up on what I was meaning so.... my bad? Receiver has an amp.. idk its like saying car rather than vehicle to me.

Second, I read everyday on this stuff all day. My mind works differently than most, I look for an understanding of whats actually going on, not just a word that sums it up. I.E. theres an important difference between knowing something and understanding something, specially in this area. So thats why I ask these basic questions, but seem to get on occasion a broad answer rather that an explanation. Thats fine though, its not your job or anyone elses to teach me, I understand that, but this is where I look to learn, its free :)

Rather than just say, the longer a cable is, the more noise it will pick up, a reason is what im looking for. I already knew that about cable length and interference but I dont know whats going on really that causes this.

The cable you described are now referred to as XLR cables, but I dont see them being used in sending analog signals to and fro, besides with Microphones in musical applications. Why is this?

Thats why I wonder that actual difference between an Analog and Digital signal, in fact someone just wrote a post wonder the gains he would receive from sending a digital signal vs. an analog. With optical is obvious as to whats going on and why, however electricity is the medium for an analog and digital signal, so it stands to reason that they will both be have the same pitfalls and benefits.

This is the difference between knowing and understanding that I look for. I always appreciate any help I get, but please dont let your answer be, "Go learn if you want to know" that kind of logic doesn't even make sense.
Analog and digital signals are totally different. An analog signal has a voltage that varies in proportion to the analog wave form, such as the pressure against a microphone diaphragm.

A digital signal is a coded signal of ones and zeros. It is a binary signal that has a bit rate and sampling frequency. It requires a coder to code it, the analog digital converter, and a decoder to play it, the digital audio converter.

Now a digital signal can, and usually does have errors, however there are error correction algorithms, that reconstruct and even make a guess (interpolation) of missing data. Digital signals are immune from interference in the usual accepted sense, although they do still have a noise floor dependent on the bit rate, and the dither required to prevent distortion with low level signals. Digital signals will drop out when the errors exceed the buffering capacity of the error correction.

When designing digital systems it is best to minimize the number of digital analog and analog digital conversions. That it why it is usually best to make a digital connection between devices were possible.

The reason long high impedance analog cables are a problem is that they become an increasingly efficient antennae for radio frequency interference and AC circuit induced hum. Also they become increasingly larger capacitors as length increases. A single conductor with insulator and shield has significant capacitance, that increases with length, and therefore causes high frequency signal loss.

Balanced 600 ohm connections are standard practice in analog connections in professional equipment.

Trying to get an education on these forums, is not free. It is real work.

If you want an education in this area, then you have to pay your dues, like everyone else. You have to understand the mathematics and physics associated with audio equipment usage and design.

This does not pretend to be a professional audio engineering forum. Unfortunately you have asked about performing tasks that require you to have a professional body of knowledge.

If you are serious about accomplishing the goals you have pondered, then you will have to acquire that professional body of knowledge. That is fundamental to any discipline. So yes, you do have to go and get an education, and right now is a very good time to do it. My answer and advice is perfectly sound and logical.

As we recover a long way down the road from this economic morass, those with a good body of knowledge in the technical and scientific disciplines will be set to get ahead.

You need to understand, that this crisis has for ever shattered the barriers between wealthy and poor nations. Whether you like it or not, you are now in competition with every citizen of the world. This country can not, and should not isolate you from that reality.

So please take a hard look at your recent posts and see what they tell you about yourself, and your ability to survive the new realities.

This world will come out of this crisis radically changed and barely recognizable. Education is your best weapon in your armamenterium coming out the other side of this crisis.
 
B

businessjeff

Junior Audioholic
I understand about a digital signal, buffers, and the whole lot. But its still being sent via electricity, which is why I wonder how its so greatly unaffected by interference side from the use of buffers. Does it not follow the same laws and principles an analog does? If so, thats quite a feat.

An analog signal is sent in an AC waveform, does a digital signal use AC as well?

Interferences in an analog AC waveform result in pulling, stretching, squeezing, ect. of the AC waveform, which obscures the original waveform along its path down the wire from A to B, thats what really going on is it not?

With that said, the reason why a digital signal is not affected as greatly would be because the wave form having to only represent 1's and 0's. If you squeeze a 1 it still looks like a 1, as well if you stretch a 0 it still looks like a 0. Rather than a waveform thats supposed to be directly proportionate to a sound wave, if that gets obscured in any little way its audible as distortion. Is that a crude idea of whats going on?

With audio, does the data being sent have headers in them for the buffers to trace back missing pieces?

Why not run DAC's and mini amps in every speaker? That makes sense to me, and I claim to be far from genius.



Anytime is a great time to get an education, its really one of the few things ever worth doing. However if you are aware of the current economic status, how would one go about paying for an education? Its still possible, but my point is its not easy these days. Perhaps through grants, scholarships, a job that offers credit matching and then loans I could pull it off, but it would not be easy. And until im 24 I cant get any grants, because I fall under the wings of my parents, who can so blatantly afford, let alone are so willing, to pay for 100% of my schooling....
 
TLS Guy

TLS Guy

Seriously, I have no life.
I understand about a digital signal, buffers, and the whole lot. But its still being sent via electricity, which is why I wonder how its so greatly unaffected by interference side from the use of buffers. Does it not follow the same laws and principles an analog does? If so, thats quite a feat.

An analog signal is sent in an AC waveform, does a digital signal use AC as well?

Interferences in an analog AC waveform result in pulling, stretching, squeezing, ect. of the AC waveform, which obscures the original waveform along its path down the wire from A to B, thats what really going on is it not?

With that said, the reason why a digital signal is not affected as greatly would be because the wave form having to only represent 1's and 0's. If you squeeze a 1 it still looks like a 1, as well if you stretch a 0 it still looks like a 0. Rather than a waveform thats supposed to be directly proportionate to a sound wave, if that gets obscured in any little way its audible as distortion. Is that a crude idea of whats going on?

With audio, does the data being sent have headers in them for the buffers to trace back missing pieces?

Why not run DAC's and mini amps in every speaker? That makes sense to me, and I claim to be far from genius.



Anytime is a great time to get an education, its really one of the few things ever worth doing. However if you are aware of the current economic status, how would one go about paying for an education? Its still possible, but my point is its not easy these days. Perhaps through grants, scholarships, a job that offers credit matching and then loans I could pull it off, but it would not be easy. And until im 24 I cant get any grants, because I fall under the wings of my parents, who can so blatantly afford, let alone are so willing, to pay for 100% of my schooling....
Actually analog waveforms are complex waves that are not sinusoidal, however mathematically any complex analog wave can be reduced to its parent sinusoidal waves by Fournier analysis. Your concept of pulling and stretching signals is way off the mark.

Digital signals are packets of information that are represented by one and zero. This is the binary code, and the basis of all computing a microprocessor systems. The basis of processor and and memory chips is millions of tiny electrical switches. My second son designs and develops them.

Interference does not come in neat binary code packages, so interference and degradation as we understand it in analog signals does not occur.

The most common basis for error correction are the Reed Solomon codes developed in 1960.

I think you will find that as part of this huge stimulus package, and educational component will be there and large, at least it had better be or we will be well and truly sunk.

That is why I said now is a good time to look to getting an education. Good jobs will be hard to come by for a while and I think and hope there will be funds available.

At your age obtaining an education needs to be your top priority or you will be sunk.
 
OttoMatic

OttoMatic

Senior Audioholic
I understand about a digital signal, buffers, and the whole lot. But its still being sent via electricity, which is why I wonder how its so greatly unaffected by interference side from the use of buffers. Does it not follow the same laws and principles an analog does? If so, thats quite a feat.

An analog signal is sent in an AC waveform, does a digital signal use AC as well?
Yes, a digital signal, in the end, is an AC waveform of some sort. It's susceptible to interference as much as an analog signal, but it can handle such interference at the "other end". A very simple case of handling it at the "other end" is the use of balanced signaling. Since the interference has affected both the positive and negative signals in the balanced cable, it can be subtracted out when it reaches its destination. Using a very, very different mechanism, many digital signaling schemes can also remove errors during demodulation.

I noted in an earlier response that you said that the cables are "called XLR cables", in a correction of TLS Guy's post. I would agree that you may call them XLR cables, but I would also suggest that they are also balanced cables. I would consider XLR cables to be a subset of balanced cables. A balanced cable doesn't have to have XLR connectors on its ends; it could also have 1/4" TRS connectors, or bare wire for that matter. Anyway, just wanted to note that.

Interferences in an analog AC waveform result in pulling, stretching, squeezing, ect. of the AC waveform, which obscures the original waveform along its path down the wire from A to B, thats what really going on is it not?
I have to agree with TLS guy that the AC waveform won't be pulled, stretched or squeezed. It's not elastic in the sense that if its affected by some outside interference it will somehow maintain its original form in a general sense. A noise spike would look just like a spike if you were to view it on an oscilloscope. It wouldn't have nice, rounded edges. Although the terminology is arguable, I think you have the right idea. An analog signal can be directly modified by other electrical sources.

You initially asked what causes such interference, and I'm not sure you were directly answered. I'll try to give one example here. In general, electrical signals general electro-magnetic radiation (or electro-magnetic interference (EMI)). Look outside at the wires carrying electricity to and from the homes in your area, and imagine there's a slight glow that surrounds the cable itself. It's invisible, but it's there, and it can be measured with various tools.

Now, when a electo-magnetic force exists, it can generate a current in a wire. When a current is generated, a new AC signal exists in that "wire"! Some motors work this way, I believe my toothbrush charger works this way, etc.

A similar radiation exists in the 120V AC cabling in your home. Now consider the signal that you are sending down a single-ended (i.e., unbalanced and analog) cable in your audio system. It's tiny compared to the 120V signals that power everything in your home. The low-level audio signal is probably on the order of 200 mV. If you place a 120 VAC cable adjacent and parallel to your audio signal, it's very possible that the EMI generated by the 120 V line will induce a current in the audio line. When that happens, you might get a 60Hz hum, since that's the frequency at which your home AC operates. That's just one basic example.

With that said, the reason why a digital signal is not affected as greatly would be because the wave form having to only represent 1's and 0's. If you squeeze a 1 it still looks like a 1, as well if you stretch a 0 it still looks like a 0. Rather than a waveform thats supposed to be directly proportionate to a sound wave, if that gets obscured in any little way its audible as distortion. Is that a crude idea of whats going on?
Digital modulation schemes don't normally just send a 1 or a 0 in the final waveform. If it existed, I would imagine that it would be the very first modulation scheme, and it's prone to problems with sampling, error recovery and EMI radiation due to potential repeating characters. I don't know exactly what modulation scheme is used for audio signals such as S/PDIF, but I would suspect that it's somewhat advanced and uses some type of error detection and correction.

Interesting, the concept of squeezing and stretching a signal applies more aptly in the digital world. Bits encoded for transmission can have effects on the signal level of adjacent bits. If you look at this on an oscilloscope, it definitely can look squished in the picture! But that's a different story... Unless there's something wrong with the setup or the system, the demodulator at the other end will be able to recover the digital data, regardless of how it looks (unless it's utterly destroyed by noise, which is indeed a possibility).

With audio, does the data being sent have headers in them for the buffers to trace back missing pieces?
I don't know specifically about audio schemes, but that type of thing is definitely a possibility in digital signaling (though I usually think of that data coming after the audio data, so I wouldn't think of it as a header).

Why not run DAC's and mini amps in every speaker? That makes sense to me, and I claim to be far from genius.
Some do; see Meridian. More frequently, speakers have amplifiers in them, but not DACs (more in the pro audio field). The reasoning not to do that is that the end user is then limited in flexibility in choosing their own amplifier or DAC. It also forces the user to digitize the signal at some point, which may be undesirable for some folks (e.g., someone playing a vinyl LP may not want anything digital in the audio path). Also, most consumer grade components are not set up to output a digital signal that could be sent to the speakers. Finally, there's just never been much demand from the consumer for this. There's no doubt that it's technically possible, but it's just never taken off.

Now, on to our other topic: getting an education is a great idea, and it sounds like you have some plains laid out. That's great and stick with it. At the same time, I think this type of forum is a GREAT place to ask questions! These are in-depth technical questions that are difficult to understand, and very difficult to explain through a forum like this. At the same time, it's a starting point and will encourage one to start to understand the issues, to ask more questions, and to check out other avenues of learning. I encourage you to continue to ask the questions, and hopefully you'll get some meaningful information in return.

Good luck!
 
TLS Guy

TLS Guy

Seriously, I have no life.
Digital modulation schemes don't normally just send a 1 or a 0 in the final waveform. If it existed, I would imagine that it would be the very first modulation scheme, and it's prone to problems with sampling, error recovery and EMI radiation due to potential repeating characters. I don't know exactly what modulation scheme is used for audio signals such as S/PDIF, but I would suspect that it's somewhat advanced and uses some type of error detection and correction.
Actually PCM does use just ones and zeros. Here is how an 8 bit PCM stream is constructed.
 
OttoMatic

OttoMatic

Senior Audioholic
Actually PCM does use just ones and zeros. Here is how an 8 bit PCM stream is constructed.
Well, kinda. It's still using an NRZ scheme to maintain clocking and remove DC (at least). So it doesn't directly and always send a 1 for a 1 and a 0 for a 0. See "Table 1" in the app note.

Also, I previously said this:

If it existed, I would imagine that it would be the very first modulation scheme, and it's prone to problems with sampling, error recovery and EMI radiation due to potential repeating characters.
I think it would actually be RF interference generated in the sidebands of the frequency spectrum when transmitting digitally or in analog. Just a nit, but I think it's RFI and not EMI at that point.
 
B

businessjeff

Junior Audioholic
Amazing, I will write a response later, but for now I have to say thank you very much for the info. You both have provided me with a basic understanding so I know where to point my education inquiry compass.

real quick though, I used pinching and squeezing terms, as I pictured it in an atomic level, where the electrons traveling through the AC wave in a signal line would be physically moved around by the effects of EMI.

Anyways, I g2g right now, ill right a little bit later on today, thanks again.
 

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