Guitar Waveform

[Jasmine Lemaitre]

Ive been writing a program that functions similar to a MIDI interpreter, but the pure tones I'm currently using very poorly resemble the actual tones produced by guitars, pianos, and other instruments.

I know that instruments exhibit an overtone series, but I can't find any information about the specific overtone profiles of guitars. How exactly would I mathematically simulate the tone of an electric guitar? How exactly does its tone differ from a pure tone?

Yes - but even talking about 'the overtone series' of an instrument is a big simplification. In reality, each of the overtones might be considered to change amplitude and pitch over the duration of the note, according to a large number of different variables.

I'd like to start with a note that if I knew a good answer to this question, I would be spending my time earning money on selling guitar emulation virtual instruments, rather than sharing the information for free.

The first part of the sound is attack. It's percussive, i.e. continuous spectrum and broad band, depending on whether the string is plucked by a finger, finger nail, or plectrum, its shape, texture, and also angle and time duration of the contact with the string. Guitar body resonance (at least with classical and acoustic instruments) may contribute significantly to the attack sound.

Plucking the string starts with displacing it from the equilibrium into a triangular shape, with the tip rounded (due to finite size of the finger, nail or plectrum, and the string rigidity). Fourier transform of this shapes gives the initial ratio of amplitudes of the harmonic components.

As the string vibrates, the amplitude of various frequencies decays in various ways, as the energy is transferred to the instrument body, the air, or dissipated internally in the string, which depending on the string parameters.

Various characteristics of guitar body and construction affect how quickly energy various harmonic components is dissipated, for all types of guitars, both classical/acoustic, and for solid body electric as well.

The string transfers energy to the guitar body via the bridge. For nylon strings this is primarily via transverse vibrations, while for steel strings both transverse and longitudinal vibrations contribute to a similar degree.

Finally, how the actual sound is produced? In classical/acoustic instruments the sound is primarily emitted by the top plate, with some additional contributions of the sound hole (resonance of the air cavity), and the back, altogether captured by a microphone, placed at some location. Then you have instruments with piezo pickups, and finally electric guitars with electromagnetic pickups. There are also some more exotic systems like optical transducers or pickups capturing vibrations of the body. Any of these systems, and its parameters affect the final sound significantly.

Last but not least are various additional sounds, characteristic for guitar. What comes to my mind is the noise when sliding the finger along the string (in particular the wound strings) and fret buzz. Also, guitarists use multiple articulation techniques which extend the spectrum of available sounds.

Normally when i record in a electric guitar (via line in from the amp and using speaker IR) the recorded audio waveform is symmetrical above and below the centre line. I have noticed that with one particular amp the lower half of the wave form is cut ("clipped" perhaps).

Anyway, I am not sure why this happens - e.g. whether it is a feature of the amp design (THD Univalve - single ended cathode biased with only one power tube ) or indicative of a fault. I have tried different pre-amp and power valves but this makes no difference - so I know its not the valves per se causing this.

Yes, "polarised compression" seems a good description. This also leads me to realise that I don't really know what the top and bottom half of the wave form represent - i.e. what generates the bottom and what generates the top, and why they would normally be symmetrical.

I should add that I don't hear anything odd. For instance, if I set the amp clean to match another amp as closely as possible, and then record both amps with the same overdrive pedal for example, I see the different waveforms but don't really hear any difference other than subtle colourings of the two amps. Have to think that if the top half of the waveform was "compressed" in the same way I should hear it, but as things stand I don't hear anything to concern me, I just see something that puzzles me - not helped by a very limited understanding of what actually goes on inside a tube amp - let alone the unusual single power tube configuration of the THD Univalve (and also complicated in that THD no longer seems to be in business so no chance of putting the question to them).

It occurred to me that I do of course have balanced jack leads - used to connect my monitors to my Apogee Duet. So I just tried one of those straight from the Univalve line out, and got the same result.

1. Connect a different amp that I have and connect its speaker out to a dummy load (THD Hotplate) with the Palmer PD01 in the line. I can then take a balanced line out from the palmer at mic level using a balanced XLR lead to the Duet. I would hope to get a normal waveform.

2. I will then simply swap the second amp for the Univalve - keeping everything else the same, i.e. not using the THD Univalve direct line out, but rather taking the speaker out to the Hotplate via the Palmer (strictly speaking I don't need the Hotplate for this since the Univalve has its own dummy load, I will keep the two set ups identical apart from amp). This should at least eliminate the direct line out from the equation.

3. If this still gives a strange waveform I will then take the speaker out from the Univalve direct to a cab via the Palmer and again use the Palmer to generate the line out signal. This will replicate the second set-up but with a real speaker as load instead of the Hotplate.

4. I may also then do a final test of removing the Palmer and mic up the cab instead - I will use the same XLR cable as I use for the two tests above. Its a while since I did this but don't recollect ever noticing anything odd with the waveform back when I did.

Thanks Jordi, that is an interesting article and inspired me to look for some more general info on what the waveform, and particularly its top v bottom, actually represents - which led me to this thread:

This includes some seemingly well informed comments on what can be expected form a single ended cathode biased amp - which is exactly what my Univalve is! (though i couldn't tell you what this means exactly - I just know thats how it is described by THD). Now on the face of it that is my original question answered - the asymmetric compression seems to be an expected and even desirable characteristic of a single ended amp.

1. I found that when connecting a second amp (with a conventional push/pull arrangement) to the palmer Pan01 and then Hotplate dummy load I got the expected "normal" symmetrical waveform form the Palmer line out, So far so good.

2. I found that when swapping the THD Univalve in but keeping everything else the same I got the asymmetric waveform via the Palmer line out (though inverted as compared with the Univalve's own line out). Ok, according to the above thread I should actually expect this and its OK.

So I now seem to have a slightly different puzzle to the one I started out with - why am I only seeing the asymmetric wave when using a dummy load (either internal to the Univalve or the external Hotplate) and not when using a cab?

I did recognise that when I originally pugged into the cab I used the Univalve power attenuator to keep the volume bearable - whereas when plugging into the Hotplate I had been setting the Univalve to full power (i.e. no attenuation). But I then eliminated this variable by trying it into the Hotplate again but with Univalve attenuated (to same degree as when plugged into the cab) and still got the asymmetric wave via the Palmer + Hoplate (compared to the symmetric waveform via the Palmer + cab). So it doesn't seem that whether or not the Univalve attenuator is used is a factor affecting the waveform.

The only thing I haven't t tried is to run the Univalve into the cab at full power and see what that gives. I guess I should try this as the final test - perhaps a real speaker must be driven much harder than a dummy load before the "expected" asymmetric compression will be revealed, but I don't know why that should be the case. I will give this a shot - but because this is home studio situation I need to pick my moment for cranking the amp through the cab - and sadly it isn't right now.

The recording plays back fine bouncing around -18 dB but the waveform is barely visible. I have another nearly identical audio track using a Logic X amp plug in with the waveform completely visible as per normal. I have made sure recording delay is set to zero as suggested and tried refreshing the overview to no avail. What am I doing wrong here??

You may have recorded too low if the Wave's plugins were active. When you are recording the file is created PRIOR to the gain staging of the insert/fx/plugins. When recording never use the channel strip meter at the fader if you have active plugins, you need the levels of the input PRIOR to any plugins. To see the recording levels you add the Meter plugin in the first slot and (basic rule) want your RMS level bouncing around at -24 to -16 dBFS and your peaks well below -8dBFS. To compare add the meter plugin in the FIRST slot of your channel strips in each track and compare the two tracks. If the track is clean add some gain with a plugin, a Gain plugin or look into the "Normalize Gain Function" for the region.

Now this might not have anything to do with Audacity itself, but the thing is, this has been happening with another guitar as well. After my first guitar started giving me these ugly waveforms, I switched to my other one and that one gave me smooth waveforms again!! I was soooo happy!

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