Classic Ac/dc Tone From A Dsl-50?

[solarburn]

No problem. People get too wrapped up around JMPs and JCM800s. If Angus Young plugged into a JCM2000 50 DSL you would have a hard time knowing. Sure the base Marshall tone, found in most good valve Marshall amps is a head start, but it really is in your fingers. You have the biggest role. After that it’s the amp, guitar, cab, speakers, room, effects, etc.

My ears thank you. This in spades.

 

[JeffMcLeod]

The music tones on an album have been all "studio'd up". I'm sure that most, if not all, of your favorite artists have played out live at one time or another, and even they aren't going to match the record tone exactly. You can't. Live is as raw as it can get, and that's the tone you should be shooting for.

AC/DC? The first 15 seconds of this video is all mankind needs to know.

 

[JCarno]

The music tones on an album have been all "studio'd up". I'm sure that most, if not all, of your favorite artists have played out live at one time or another, and even they aren't going to match the record tone exactly. You can't. Live is as raw as it can get, and that's the tone you should be shooting for.

AC/DC? The first 15 seconds of this video is all mankind needs to know.

Note: No wireless being used. Just sayin.

 

[What?]

Here is a live recording demonstrating Malcolm's tone.



You can easily hear the clarity and treble in Malcolm's tone without any studio tricks. I find that live sound to have the same tonal characteristics as the studio sound, even though it wasn't well recorded and is lacking lower frequencies.

Here is the studio sound (jump to 16 minutes into it for this one).



I think the thing with his tone is that you can clearly hear what the guitar sounds like, not being burried under a bunch of muck.

 

[EADGBE]

I think it can.

 

[What?]

I think this is about as close as I am going to get to Malcolm's tone with the DSL-50. https://app.box.com/s/7ufz92o4gzxltmi2n488sg1dx49acyya

That's just a quick straight up recording. No eq, compression, or other foolery. Also, a room mic would help. I don't have time to set one up at the moment.

 

[Allterr]

I think this is about as close as I am going to get to Malcolm's tone with the DSL-50. https://app.box.com/s/7ufz92o4gzxltmi2n488sg1dx49acyya

That's just a quick straight up recording. No eq, compression, or other foolery. Also, a room mic would help. I don't have time to set one up at the moment.

That is very close . Well done!

 

[What?]

That is very close . Well done!

Thank you! It's a learning experience, for sure. By the way, here are the settings I ended up with.

Green channel volume 2
Gain 2
Mid and Bass just above 2
Presence and High 6
Deep switch on
Mid shift off

Speaker is a Weber 1230-55 pre-rola. It's a good speaker, but I would recommend it as a supplement to another speaker that has more mids rather than on it's own. For the type of sound on Live Wire, I was fighting against this speaker's nature. It's a very warm speaker, and I think it is just about perfect for Thin Lizzy stuff, such as Jailbreak.

An old Realistic dynamic just off center of the speaker and an inch off the grill. It's a fairly flat sounding mic with a little a bit of brightness. It's my alternative to an sm57 when I don't want an sm57's low end and upper mid boost.
An apex 205 ribbon mic around a foot away from the center of the speaker. It's my first ribbon mic, and I'm digging it so far. It has alot of weight in the low end with rolled off highs. It's really prone to proximity effect up close but when pulling it back a bit and mixing it under a dynamic, it works well. Sounds great on acoustic guitar too after shelving the low end off, where it sounds much more natural than typical cheap condensers which tend to sound overly bright and artificial to my ears.

The hardest part of this little learning experience was getting the mics right, meaning placement, phase alignment (delaying the dynamic mic track to be in phase with the further away ribbon mic track), and finding the right balance between the track levels. Keep pushing the mics around and the results are much better than trying to correct things using post eq.

To really get in the ballpark of Malcolm's sound on It's a Long Way to the Top, I would need a different speaker that has a vowely midrange. Maybe an alnico speaker? After lots of listening and comparing to tracks on High Voltage I realized that Malcolm and Angus had a few pretty different guitar sounds each across the album. I think that with a few right speakers the DSL could passably do those sounds. Speakers make a huge difference!

 

[mtm105]

Wampler Pinnacle on Green channel.

And Greenbacks.

And an SG.

 

[Neil S]

I post a track trying to show you I can get an almost acceptable AC/DC sound from a DSL then you go and blow me away!, I love it!! Great work.

I was told by a recording engineer how to get a good quick starting point for phase delay with multiple microphones but never have used it, I may be wrong so here goes....

First measure the distance accurately between the two diaphragms in feet then,
Ds is delay in milliseconds
X is seperation in feet
C is an average speed for sound (used by audio engineers) 1200 ft per second
if your two mic's Diaphragms were 4 feet apart you wold have
4 x 1200 / 1000 = 4.8 millisecond delay then fine tune by ear ( if you can )
I don't have two mic's so can't test it out but I Googled it and saw the same calculation, just don't know if I understand how to apply it correctly.
My room sounds like crap but now you have me wanting to fiddle with a second mic again $$$

 

[Fender]

yup pretty close one !
I had been able to nail it too with a 30th anniversary and channel 2 mode B with… a telecaster

 

[What?]

I post a track trying to show you I can get an almost acceptable AC/DC sound from a DSL then you go and blow me away!, I love it!! Great work.

I was told by a recording engineer how to get a good quick starting point for phase delay with multiple microphones but never have used it, I may be wrong so here goes....

First measure the distance accurately between the two diaphragms in feet then,
Ds is delay in milliseconds
X is seperation in feet
C is an average speed for sound (used by audio engineers) 1200 ft per second
if your two mic's Diaphragms were 4 feet apart you wold have
4 x 1200 / 1000 = 4.8 millisecond delay then fine tune by ear ( if you can )
I don't have two mic's so can't test it out but I Googled it and saw the same calculation, just don't know if I understand how to apply it correctly.
My room sounds like crap but now you have me wanting to fiddle with a second mic again $$$

I'm terrible with math. Funny, because I do math as a hobby, filling in my swiss cheese high school math education bit by bit. But yes, that is roughly correct. It will get you in the ballpark. But when dealing with phase alignment, the ballpark is huge and fine tuning by ear can be really tedious. But since we're working with digital these days, we can do very precise alignment after the fact without ever breaking out a tape measure. The dead simplest way is to drag the waveform of one track to be in alignment with the other track. But you want to zoom in at the sample level when doing so. Being 1 sample off makes a big difference. You will easily hear it. It takes a little practice (not much) to do this. One tip here is to adjust the gain of your mics to be roughly the same so that the waveforms are about the same height. That makes identifying the shapes of peaks easier. So after your gain is set, record a note or chord. Zoom in at the beginning of the recording so that you can clearly see the shapes of the peaks, and drag one waveform to be roughly aligned with the other. Now move your play cursor to a peak and zoom in at the sample level and align the waveforms at the sample level making sure that the samples are vertically aligned. That is the part that can take a little fiddling at first, but you will immediately hear it when you are dead on. You should be able to do this in any recording software. Once you find the phase alignment, I suggest that you drag one of the waveforms to be 1 sample off in either direction and listen to it for comparison. It's a big difference. Play around with it a bit. Try 2 samples off. 1 ms off. The best solidity of the mids is only there when the tracks are phase aligned.

Once you have that down, you'll probably want to set up a sample delay so that you don't have to drag around waveforms anymore. But you'll still need to initially drag a waveform to find phase alignment, and if you move the mics at all you will have to manually find the phase alignment again. So after you have manually aligned your waveforms, jump to the starting point of one of the waveforms and make a time selection spanning to the beginning of the other waveform, at the sample level. That is the time offset in samples. You'll want to set your recording software to display time in samples for this rather than seconds and milliseconds. Remember that number in samples. Then you can ditch your scratch waveforms that you used for finding the offset and add a sample delay to the track that has the closest mic, setting the time (in samples) to the offset that you found. No more dragging around waveforms.

I should also mention here that I use Reaper as my recording software, and it has some features builtin that are helpful for phase alignment. The first feature is that if you use a folder track to contain your 2 mic tracks, the folder track will display the 2 waveforms as an overlay, which makes manual alignment easier. The second feature is that Reaper has a builtin sample delay plugin called Time Adjustment Delay which works in samples.

But you should be able to do sample accurate phase alignment in any recording software either manually, or using a free sample delay plugin such as Voxengo Sound Delay.

I will also mention here that I have been playing around more with the dynamic mic / ribbon mic combination. It works really well for every type of tone that I have shot for so far, be it crunch, clean, or heavy high gain stuff. The ribbon brings a ton of low end weight and thickness in the mids to the sound. It's best to spend some time finding good placement of the two mics (and finding phase alignment for each placement) before ever touching any eq. Use the dynamic up close and pull the ribbon back around a foot to knock off the big proximity effect. If I get time on my next day off I will post some recordings of various sounds using the DSL-50.

And by the way, my room is crap too. If you clap your hands in your room and hear that pingy short reverb sound, you'll know what I'm talking about, and that isn't saying anything about the much worse peaks and nulls of lower frequency standing waves in a small room. It gets much more noticeable at higher volumes, making notes sound a bit icepicky and smeared and chords sounding boxy. I really could use some room treatment for recording and mixing. I initially worried about how the figure-8 pattern of a ribbon would sound in my room, but it is not a big problem at a close distance.

 

[saxon68]

Hey all. First post here.

Is it possible to get that classic AC/DC tone from a DSL-50? I'm talking about the sounds on the High Voltage album.



More specifically, Malcolm's sound on It's a Long Way to the Top. It sounds almost clean, but has dirt in the upper frequencies, is punchy, and cutting. And it has some sort of resonance to it that I can't describe. It's a beautiful sound.

Malcolm also used a Gretsch and those pickups have a sound of their own. Also he used very little gain, lots of volume, thick strings and hit them like a truck.

 

[Neil S]

I'm terrible with math. Funny, because I do math as a hobby, filling in my swiss cheese high school math education bit by bit. But yes, that is roughly correct. It will get you in the ballpark. But when dealing with phase alignment, the ballpark is huge and fine tuning by ear can be really tedious. But since we're working with digital these days, we can do very precise alignment after the fact without ever breaking out a tape measure. The dead simplest way is to drag the waveform of one track to be in alignment with the other track. But you want to zoom in at the sample level when doing so. Being 1 sample off makes a big difference. You will easily hear it. It takes a little practice (not much) to do this. One tip here is to adjust the gain of your mics to be roughly the same so that the waveforms are about the same height. That makes identifying the shapes of peaks easier. So after your gain is set, record a note or chord. Zoom in at the beginning of the recording so that you can clearly see the shapes of the peaks, and drag one waveform to be roughly aligned with the other. Now move your play cursor to a peak and zoom in at the sample level and align the waveforms at the sample level making sure that the samples are vertically aligned. That is the part that can take a little fiddling at first, but you will immediately hear it when you are dead on. You should be able to do this in any recording software. Once you find the phase alignment, I suggest that you drag one of the waveforms to be 1 sample off in either direction and listen to it for comparison. It's a big difference. Play around with it a bit. Try 2 samples off. 1 ms off. The best solidity of the mids is only there when the tracks are phase aligned.

Once you have that down, you'll probably want to set up a sample delay so that you don't have to drag around waveforms anymore. But you'll still need to initially drag a waveform to find phase alignment, and if you move the mics at all you will have to manually find the phase alignment again. So after you have manually aligned your waveforms, jump to the starting point of one of the waveforms and make a time selection spanning to the beginning of the other waveform, at the sample level. That is the time offset in samples. You'll want to set your recording software to display time in samples for this rather than seconds and milliseconds. Remember that number in samples. Then you can ditch your scratch waveforms that you used for finding the offset and add a sample delay to the track that has the closest mic, setting the time (in samples) to the offset that you found. No more dragging around waveforms.

I should also mention here that I use Reaper as my recording software, and it has some features builtin that are helpful for phase alignment. The first feature is that if you use a folder track to contain your 2 mic tracks, the folder track will display the 2 waveforms as an overlay, which makes manual alignment easier. The second feature is that Reaper has a builtin sample delay plugin called Time Adjustment Delay which works in samples.

But you should be able to do sample accurate phase alignment in any recording software either manually, or using a free sample delay plugin such as Voxengo Sound Delay.

I will also mention here that I have been playing around more with the dynamic mic / ribbon mic combination. It works really well for every type of tone that I have shot for so far, be it crunch, clean, or heavy high gain stuff. The ribbon brings a ton of low end weight and thickness in the mids to the sound. It's best to spend some time finding good placement of the two mics (and finding phase alignment for each placement) before ever touching any eq. Use the dynamic up close and pull the ribbon back around a foot to knock off the big proximity effect. If I get time on my next day off I will post some recordings of various sounds using the DSL-50.

And by the way, my room is crap too. If you clap your hands in your room and hear that pingy short reverb sound, you'll know what I'm talking about, and that isn't saying anything about the much worse peaks and nulls of lower frequency standing waves in a small room. It gets much more noticeable at higher volumes, making notes sound a bit icepicky and smeared and chords sounding boxy. I really could use some room treatment for recording and mixing. I initially worried about how the figure-8 pattern of a ribbon would sound in my room, but it is not a big problem at a close distance.

Some good tips there thanks.
I use Protools so there should be a way to do what you mentioned.
Lately I have been fiddling with some old tracks I recorded and do a fake double track by duplicating, panning left and right then delaying one track a few milliseconds with a little chorus like wow on one track. My room has that bad pingy reverb and bad cancelation effects. I can move only a couple of feet and go from feeling the amp rattle me to feeling nothing even at high volume.
I had a look at some ribbon mic's while my wife wasn't watching last night

 

[ibmorjamn]

The older amps relied on power tube drive and volume that is why it is clean sounding . Very hard to duplicate with a DSL because they just don'T have any punch . It is missing it's ball's in my opinion . I know guys get good tones but it is gutless to me.

 

[Flyingv4me]

I use a DSL 50 head as my main amp. You can get AC/DC out of it. Use the green channel, bring up the volume first....then bring up the gain. A lot has to do with the OLD Celestions Angus used...old green or black back 25 watt speakers. Best speakers ever for hard rock guitar

 

[What?]

Lately I have been fiddling with some old tracks I recorded and do a fake double track by duplicating, panning left and right then delaying one track a few milliseconds with a little chorus like wow on one track. My room has that bad pingy reverb and bad cancelation effects. I can move only a couple of feet and go from feeling the amp rattle me to feeling nothing even at high volume.

Try comparing a fake double track with a real double track. The thing to me with a fake double track is that it lacks complexity in the sound because the only difference between the tracks is a small static time difference. With a real double track there is small moving time differences (in both directions), small differences in performance, and so on. To my ears a real double track makes for an end result that is much more interesting to listen to.

Room modes are a strange thing at first. There are peaks and nulls in the low frequencies all over the room. Any speaker that can reproduce lower frequencies can be used to demonstrate this (not laptop speakers). Try playing a low frequency sine wave out of your speakers. It doesn't need to be loud, and I would suggest keeping it low volume to prevent possibly damaging your speakers. I would start with the volume all the way down, start playback, and then bring up the volume to speaking level. With that playing, slowly walk around your room and you will hear the huge holes and bumps in the bass. Stand in a corner and you will hear a huge bump in bass, as if the bass is coming from the corner. In other spots in the room, it will almost sound like there is no bass at all. These holes and bumps are where standing waves are summing out of phase and in phase. And it's why recording engineers go on about finding the best spot in the room when recording amps and drums and when setting up studio monitors for mixing. If you are setup to record or listen in a null, bass will be very weak. If you are setup to record or listen in a peak, bass will be heavily exaggerated. For recording, that can be used to our advantage for adding to or taking away from a cab's low frequency response.

Here is a fairly clean 100 hz sine wave to use:



If you play that out of laptop speakers, it will sound buzzy because those tiny speakers aren't capable of reproducing 100 hz cleanly.

As a side note, I ran onto this one:



That is what aliasing sounds like. Hear that higher frequency buzz mixed in with the lower frequency sine? It's what happens when a frequency outside of the audio range is being calculated by the dac (or resampling software) as a frequency within the audio range, commonly called foldback distortion. Someone should have checked that tone before uploading it to youtube.

If you want to explore further, you can use a tone generator plugin in your recording software to produce audio tones of any frequency. But remember to turn down the volume before firing up a tone generator so that you don't damage your speakers. A steady level tone is much more taxing on your speakers than music that varies in level and frequency. You can also try placing your amp/cab in a corner vs. a null spot in the room and listening to it from a peak vs. null spot.

We have veered off topic here. Maybe all this should be going in a new thread.

 

[What?]

I found that Malcolm tone! It took a plexi to get there. Absolutely nothing else is special.

https://app.box.com/file/721846433535?s=m6pva4iyybhhtr6qs2bcsvomcv4wavro

Guitar is a cheapo Charvel with low output alnico II pickups. Cab is an old Peavey VTM MS412 with G12K-85's. Mic is an sm57, into a handheld recorder. Tubes are all JJ. Nothing special here, except the amp which is a plexi clone.

That was my first go at it, and it's just a straight up recording with no processing. Maybe it could be dialed in closer, but I'm stoked with finding that. It is recorded in crappy conditions so lots of noise.



Also, at post #66 in this thread I gave a clip of much trial and effort using a DSL50 2 years ago. That amp doesn't breakup the same and can't get bright enough.