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Do You Have a Favorite Vocal Chain?


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I'm doing my Friday Tip for the PreSonus blog on the vocal chain I use in Studio One, and it got me curious about what y'all use as a vocal chain. For me, it's:

 

* Steep, very low-frequency cutoff

* Limiter to tame peaks

* Compressor, usually an LA-2A type, set for not much compression

* EQ, which is mostly about a high-frequency shelf boost, additional LF rolloff, and sometimes, slight midrange lift around 3-4 kHz to make the voice pop

* Delay, typically 1/8th notes, mixed in low for ambiance

* Convolution reverb using one of my custom impulses for voice

 

On rare occasions I'll add some sort of doubling, but that seldom goes all the way through a song. It's brought in for specific sections to differentiate them from other sections. Also, sometimes I use the Waves Maserati plug-in for vocals when I want its distinctive sound.

 

How about you?

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I'll be checking in, this is an interesting thread idea.

 

Putting up a "mic booth" with an isolation box has changed how I can use a microphone. I no longer have to contend with the sound of the room or most of the background noise (upstairs neighbor vacuuming her carpets excepted).

 

I can turn the gain up higher and put the mic back farther without noise or room problems, a total game changer. Proximity effect can be controlled by distance, just for one thing.

 

Am testing and documenting all of my mics so I know what works. I've got some combinations to try too, the CAD D-82 ribbon mic sounds great on vocals but it could use a brighter mic as a companion, just for one example.

 

Haven't tried compressing on the way in, I do like the sound of the Focusrite ISA One mic-pre running into a mic pre in the Presonus Quantum the best so far for vocals. I have an FMR RNC inserted but haven't messed with it.

Once I find a preferred mic setup I will experiment with Super Nice Mode on the compressor with a low ratio and see if it improves anything.

 

I hope others post, I want to learn cool stuffs!!!!!!

It took a chunk of my life to get here and I am still not sure where "here" is.
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I'm doing my Friday Tip for the PreSonus blog on the vocal chain I use in Studio One

 

The microphone that looks right it's right for the job (and most of the time it is) into the most convenient preamp, then record. I don't use chains or presets, I listen for what the song and singer needs, and then do it. If a singer is very dynamic and the song calls for it, I'm not afraid to ride a fader during tracking.

 

If the singer is unintentionally very dynamic, huffs, spits, or pops, I'll suggest how to correct that before it gets to the microphone. Though I conceded that this isn't really what you're looking for.

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I listen for what the song and singer needs, and then do it.

 

Well, that's my point. I'm the singer, my voice is my voice, I know what it needs, and I have a vocal chain for it. Of course, you need to alter a chain's parameters to accommodate different vocalists and situations (for example, I might want to make some changes to mesh better with background vocals). But hat's what an effects chain is all about: a point of departure to save setup time, and start recording faster. Altering a few parameters, or enabling/bypassing effects in the chain, saves a lot of time compared to doing it from scratch.

 

And since there's no accounting for taste :) , a lot of people who've listened to my music comment on how much they like the vocal sound. So a chain will at least get them started in that direction.

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I'm the singer, my voice is my voice, I know what it needs, and I have a vocal chain for it.

 

I was about to revise my snotty remark about "no regular chain, listen first" when It occurred to me that Craig may have been asking what you use for a voice that you already know pretty well - in this case, his own voice, as he said in his reply to me. Sure - no reason not to use what you already know works most of the time, with enough handles built into the chain so that minor tweaks are easy to in order to better fit the song.

 

If I'm working on an album project over an extended period of time I'll get to know the singer(s) and if I find, when mixing (even for a rough or monitor mix) that I'm always using a little compression or a little EQ, I might plug that in for starters when the next session comes along, if only to remind me how I wanted the vocal to sound the last time we worked.

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I'm the singer, my voice is my voice, I know what it needs, and I have a vocal chain for it.

 

I was about to revise my snotty remark about "no regular chain, listen first" when It occurred to me that Craig may have been asking what you use for a voice that you already know pretty well - in this case, his own voice, as he said in his reply to me. Sure - no reason not to use what you already know works most of the time, with enough handles built into the chain so that minor tweaks are easy to in order to better fit the song.

 

If I'm working on an album project over an extended period of time I'll get to know the singer(s) and if I find, when mixing (even for a rough or monitor mix) that I'm always using a little compression or a little EQ, I might plug that in for starters when the next session comes along, if only to remind me how I wanted the vocal to sound the last time we worked.

 

I am an "Army of One" here too. It's pretty common. So I will find what it best for me, document it and make life simple. I've been singing close up into dynamic mics for 40 years performing live music so this is a bit different.

Evolutionary process for me at this point but I've narrowed some things down. No room sound, much less noise, those are permanent additions. Soon I'll have vocal samples, labelled, in set circumstances and I can compare them.

It took a chunk of my life to get here and I am still not sure where "here" is.
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Various mics > Neve preamp > RNLA compressor > DAW. I never use EQ during tracking.

 

In the DAW: I don't actually have a specific chain, but it's usually something like - EQ with steep roll-off for bottom end > limiter (relatively mild settings) > compressor (very mild settings) > Tapehead > EQ (totally depends on the singer and type of song, but I often like a lot of airiness in my vocals if the song calls for it).

 

I send it through an aux send. This will usually have a delay and reverb. Most of the music I do is very spacious and ambient, so they're typically rather long.

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I don't go through anything on the way in to the DAW, other than the interface preamp.

 

Ken, your chain is pretty similar to mine. I also use a send to delay and/or reverb if there are other vocals that need to have commonality.

 

Background vocals are a different animal, I usually send them to a bus (pre-fader), and they get their own processing.

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I don't go through anything on the way in to the DAW, other than the interface preamp.

 

First, I am certainly not saying that your way is wrong, there are good reasons for it.

 

That is what I used to do as well. Then I tried a Cloudlifter, I wanted one because of the claim to improve the signal to noise ratio. It does that, it also can provide more options for working the proximity effect of a microphone since you can easily get more gain without adding noise - allowing you to back off the mic more. That does work better on some mics than others so it is part of the experiment. In a sense, it reminds me of the improvement I get lowering my pickups on my guitar - although it is an entirely different set of parameters. Distance seems to smooth out sibilance without compromising the high frequencies, just for one thing. It also allows finding the sweet spot for proximity effect.

 

Then, it occured to me that using a mic pre as a "mic pre pre" could also provide the same benefits and more. When I had an opportunity to do a partial trade for the Focusrite, I jumped on it.

That gave me more options - variable gain, 48v so I can use condenser mics, a 4 way impedence (Mike Rivers says it's probably a resistance) switch that provides different responses from the same mic, a HPF and a switchable insert loop.

 

Plus an excellent sounding DI, which is a completely different signal path from input to output.. Since both outputs are XLR, I just run them straight into mic inputs on the Presonus Quantum.

 

By gain staging from back to front - setting up a recording channel with headphones on and turning up the gain on the Quantum preamp until I hear noise, then backing down a bit and then adjusting the coarse and fine gain controls on the Focusrite, I can get more gain with less noise. Using a Neat King Bee, I can now back off from the mic - the difference between singing at 8 inches or at 16 inches without increased internal noise is a big benefit to me, I really prefer the overall sound of being back farther, something I can now do easily.

 

I do like the "flavor" of the input transformer too, a value added feature.

 

It's made a positive improvement on the sound going in, I will keep doing it. I've got two channel strips to play with as well but thought I would pick my favorite vocal mic first and then compare the my 4 (including the Cloudlifter), mic pre pre options.

After that, if a small tweak on the EQ or compressor options provides a better result then I will use it. Results have been encouraging so far.

It took a chunk of my life to get here and I am still not sure where "here" is.
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On my recently-retired setup, I'd go through an RNC compressor on "really nice" with just a tad of reduction. Then in the DAW, the Waves LA-2A, the Waves Hybrid EQ for low end rollof and just a touch of EQ bringing up the 4k-5k region more or less. Nothing particularly unique in this.

 

I also got used to having a bit of reverb on the vocal while tracking - not just for singer's vibe, but also cause I tend to think of vocals as a single thing with two parts - the voice and the "room" together always.

 

Now that I've got my new Apollo Twin X, it's back to the drawing board. UA has packed the thing with zero-latency plugins geared for tracking purposes. One of which is an emulation of the UA 610 preamp - which I already own in the hardware incarnation - so it's going to be really interesting comparing the hard and soft versions. And all the other UA toys including the Manley Voxbox emulation...much fun in store.

 

nat

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Now that I've got my new Apollo Twin X, it's back to the drawing board. UA has packed the thing with zero-latency plugins geared for tracking purposes. One of which is an emulation of the UA 610 preamp - which I already own in the hardware incarnation - so it's going to be really interesting comparing the hard and soft versions. And all the other UA toys including the Manley Voxbox emulation...much fun in store.

 

One of the really cool Apollo features is that you can hear zero-latency effects while tracking, without committing to recording the effects - although you can, or print the effects later.

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I don't go through anything on the way in to the DAW, other than the interface preamp.

 

First, I am certainly not saying that your way is wrong, there are good reasons for it.

 

Well, there are certainly processes you might want to add while tracking, because that's the only way you're going to get them - like going through an audio transformer, which is a difficult effect to emulate in software. The reason I don't add any effects on the way is simply because vocals go on early in the process, so that the rest of the instruments can key off the vocals. Working with the most natural vocals possible, while listening to the other instruments, gives me a good idea of how I want to EQ things when mixing.

 

But this might also be because I postpone EQ until pretty late in the process, which probably isn't the way most people do it. The reason why, is that I prefer to have the tracks pretty much as unprocessed as possible, so I can get a perspective of where the issues are. If I compensate for issues before all or most of the tracks are down, then I frequently have to re-compensate later to take subsequent changes into account.

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I don't go through anything on the way in to the DAW, other than the interface preamp.

 

First, I am certainly not saying that your way is wrong, there are good reasons for it.

 

Well, there are certainly processes you might want to add while tracking, because that's the only way you're going to get them - like going through an audio transformer, which is a difficult effect to emulate in software. The reason I don't add any effects on the way is simply because vocals go on early in the process, so that the rest of the instruments can key off the vocals. Working with the most natural vocals possible, while listening to the other instruments, gives me a good idea of how I want to EQ things when mixing.

 

But this might also be because I postpone EQ until pretty late in the process, which probably isn't the way most people do it. The reason why, is that I prefer to have the tracks pretty much as unprocessed as possible, so I can get a perspective of where the issues are. If I compensate for issues before all or most of the tracks are down, then I frequently have to re-compensate later to take subsequent changes into account.

 

I get it, your way of going about it makes sense to me.

 

As you said, that transformer sound is tough to get with a plugin. It was a bit of a "wow" thing for me the first time I fired up the Focusrite and I've continued to like it - it's held up well.

So far, no EQ, I probably won't use much although I would think that if you consistently use a steep HPF on your vocals there may be a way to do that going in. My thought is that if I think I need to change the sound of the mids or highs much I am probably not using the best microphone for the job. I haven't been tempted to insert the FMR RNC in the vocal chain although I did find with a bit of tweaking it could subdue some high transients on bass guitar going into the DI.

A pop filter is a better way to do that with a microphone, or more distance provided it doesn't gut the low end.

 

I've tested 5 mics so far, the King Bee twice since the first test indicated an improvement to the isolation box - which was successful. I've got a 6th mic up and will test it later. Will also retest a mic that I noticed I'd had the HPF on, I want to hear the mics without anything switched in and go from there. Eventually I'll update my mic thread in Dr. Mike's forum.

It took a chunk of my life to get here and I am still not sure where "here" is.
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One of the really cool Apollo features is that you can hear zero-latency effects while tracking, without committing to recording the effects - although you can, or print the effects later.

 

I'm thinking maybe I can get it set up to print both in separate tracks in Reaper while recording - so I can A/B the dry and effected versions easily.

 

I'm just starting to get some traction using Reaper - I so did not want another learning curve after being a Sonar/Cakewalk user for decades. But now I'm starting to see how Reaper is so much more customizable - and I don't mean just custom colors and such, but you can speed up work flow tremendously in Reaper way beyond the older DAW.

 

nat

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I haven't been tempted to insert the FMR RNC in the vocal chain although I did find with a bit of tweaking it could subdue some high transients on bass guitar going into the DI.

 

I found the RNC to be very transparent in the vocal chain, if used with a light hand. I know I'll always compress vocals, so it doesn't bother me to start the process at the onset. It's like a couple of swipes with super fine grain sandpaper - just smooths it all down a bit, makes it a bit prettier for the "real" compression later.

 

nat

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I haven't been tempted to insert the FMR RNC in the vocal chain although I did find with a bit of tweaking it could subdue some high transients on bass guitar going into the DI.

 

I found the RNC to be very transparent in the vocal chain, if used with a light hand. I know I'll always compress vocals, so it doesn't bother me to start the process at the onset. It's like a couple of swipes with super fine grain sandpaper - just smooths it all down a bit, makes it a bit prettier for the "real" compression later.

 

nat

 

Thanks Nat, I am pretty familiar with the RNC, I've owned one for a few years and tried all sorts of things with it. Yes, a light touch goes a long way. The Super Nice compression is very hard to detect if the ratio is low and the threshold is just keeping the peaks from blowing out.

 

I'm finding that my homemade mic booth and isolation box allow me to get back from the mic (up to 16") and that changes the situation considerably. I don't tend to sing with a huge dynamic range and moving back a bit reduces plosives and sibilance to a great degree. Right now I am testing and selecting mics, when I chose my favorites round 2 will begin with a pop filter. Once I find my happy set up I can leave it in the iso box and it will be ready. At that point I'll experiement with compreesion going in.

 

I also plan on testing both of my channel strips and seeing what they have to offer - HHB Radius 40 with much better tubes and a Presonus Eureka are both in my rack. Both have compression, HPF and EQ, lots to experiment with.

I am not opposed to finding my best sound for my voice before it hits the converters. If I was recording somebody else I'd probably just use the Cloudlifter with the same mic setup unless it bugged them or they brought something to try.

Cheers, Kuru

It took a chunk of my life to get here and I am still not sure where "here" is.
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I don't go through anything on the way in to the DAW, other than the interface preamp.

 

Ken, your chain is pretty similar to mine. I also use a send to delay and/or reverb if there are other vocals that need to have commonality.

 

Background vocals are a different animal, I usually send them to a bus (pre-fader), and they get their own processing.

 

Oh cool. I guess it seems to work. I might switch around the second EQ, tapehead or the compressor sometimes, but that's probably more due to lack of caring/faulty memory than anything else. :D

 

Background vocals: I always send them to a bus for their own processing.

 

For the main vocal's reverb, I tend to use the same main reverb/delay as everything else BUT I might slap on another delay or reverb on the vocal track itself if it needs something else, and if I don't like the way it sounds with the general buss reverb, I'll sometimes send it to another aux send with a delay/reverb to that. Why? Because there's a chance I might want to affect something else in that same way, and if I do, it's already set up.

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Well, there are certainly processes you might want to add while tracking, because that's the only way you're going to get them - like going through an audio transformer, which is a difficult effect to emulate in software.

 

It's true that a software-modeled transformer is a pretty rare bird. Most of the time when people want a "transformer sound" they use a compressor plug-in that gives them the sound that they're after. And, I'll bet, that as long as it's there, they use a little compression as well, rather than bypass the gain reduction section of the plug-in or set the threshold and ratio so that it's not doing anything.

 

But passing a recorded track through a transformer isn't any more difficult than setting up an outboard effect send/return path - assuming you have an interface with at least one spare in put and output - and you have a transformer that you like (or think you do). Scott Dorsey's favorite sound mangling transformer in a 500-series rack module is just waiting for you. Or you could salvage any old transformer and experiment on the bench to find something you like. This is a really good DIY project, and an exercise in using a capability of your DAW that you may not have tried yet. I haven't given it a lot of thought (there might be an article here) but it might even be possible to cobble up a way to do this with even a simple 2-in-2-out interface if you monitor in mono.

 

I postpone EQ until pretty late in the process, which probably isn't the way most people do it. The reason why, is that I prefer to have the tracks pretty much as unprocessed as possible, so I can get a perspective of where the issues are. If I compensate for issues before all or most of the tracks are down, then I frequently have to re-compensate later to take subsequent changes into account.

 

I believe that this is the way that it ought to be done. I suspect that the "most" people who add EQ or other processing to a vocal (or any other "natural" track) on the way in is because it's just so doggone convenient to do what you think it needs and be done with it. Setting up EQ in the monitor path (only) might be helpful if there's something distracting about the vocal that you can't fix another way, but indeed it's best to not get rid something that's better worked with during mixdown.

 

We all have our workflow preferences, and the more options we understand and can use, after a while, things will usually go along smoother than when using a canned setup - unless, as we've both said before - if you already know the characteristics of the voice well and can take care of things up front you know will be a problem..

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Further thoughts...

 

This thread starts at a time when I am doing pre-production for a long overdue recording project. The songs have had plenty of time for consideration and I've learned much gigging them.

Currently I am selecting mic options, a HUGE part of any vocal chain - possibly the most important equipment consideration for the entire project since my songs are about the lyrics and telling a story with MY voice.

I've got a bit of jumping through hoops ahead but i am excited to be doing it. The major problems of uncontrollable background noise and an unattractive room sound have been addressed to a degree that I feel confident moving forward.

 

If I were recording somebody else's project and they just wanted me to engineer - YES, I would record everything as close to how it actually sounds when played as possible. Since I am not the composer, arranger or producer, getting things to fit together means I may need to use EQ. I may question if EQ really addresses the problem but sharing that question is not what I've been hired to accomplish. So I use the tools available to me and leave my options open going in.

 

Which is why I am testing ALL of my mics, even though I know already that quite a few of them are not really the ones I favor for my own voice. At the same time, my voice can and does inform to a certain degree what a particular mic might be suited for, a couple of the mics I tested last night struck me as being good choices for female vocals and one seemed a good choice for backup vocals. Knowing that may be useful later so I document it now.

 

Where am I going with all of this? Having heard MANY bands and having run sound for more than a few - I would say that the "problems" are not so much EQ as Arrangement and Composition. A big part of that is creating interesting parts that stay out of each other's way and serve the song.

 

I am not saying that unison parts should not be used, there are too many examples of that technique being done well to argue against it. At the same time, bands like The Police, Yes or even the Rolling Stones (who have certainly piled things in but at the same time, the beds that Keith, Mick Taylor, Charlie and Bill create on some tunes are iconic), have arrangements that work well together and leave space for the other parts.

 

I have a "vision" (strange word to use when describing audio!) for how I want my bass and kick drum tracks to interact, both musically and sonically. If I can achieve that they should be mix ready going in. If I document it, it is repeatable.

 

Vocal track sounds first, then Kick, then Bass. If I get those working well together the rest is hopefully more a matter of de-cluttering in advance than of trying to fix it with EQ later.

 

Very open to thoughts and feedback! Cheers, Kuru

It took a chunk of my life to get here and I am still not sure where "here" is.
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But passing a recorded track through a transformer isn't any more difficult than setting up an outboard effect send/return path - [/i]assuming you have an interface with at least one spare in put and output[/i] - and you have a transformer that you like (or think you do). Scott Dorsey's favorite sound mangling transformer in a 500-series rack module is just waiting for you. Or you could salvage any old transformer and experiment on the bench to find something you like. This is a really good DIY project, and an exercise in using a capability of your DAW that you may not have tried yet. I haven't given it a lot of thought (there might be an article here) but it might even be possible to cobble up a way to do this with even a simple 2-in-2-out interface if you monitor in mono.

 

That's an interesting idea...sounds like a product waiting to happen. I do integrate hardware with my DAW, but rarely. Even though "external hardware" plug-ins can ping the audio path to compensate for latency, it always seems like I need to do some manual timing tweaking to get things to line up. Besides, I need to reach around into strange places to connect patch cords, and make sure I don't set up a feedback loop in the virtual mixer by mistake. I'm really trying to figure out how to minimize any kind of interruptions to the recording process.

 

I guess it comes down to workflow. If you have a favorite transformer and always use it with your voice, you can speed things up by just recording with it. But deferring to a mixdown decision might be the best option if you're not 100% sure you want that sound.

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That's an interesting idea...sounds like a product waiting to happen.

 

As I pointed out, there already is a commercial product, though not a very well publicized one. Scott's looking for someone to get Sweetwater or Full Compass to start handling his line. Seems that small software companies do better than small hardware companies in this regard - probably because it's easier to return software products that don't sell. ;-(

 

I do integrate hardware with my DAW, but rarely. Even though "external hardware" plug-ins can ping the audio path to compensate for latency, it always seems like I need to do some manual timing tweaking to get things to line up. Besides, I need to reach around into strange places to connect patch cords, and make sure I don't set up a feedback loop in the virtual mixer by mistake. I'm really trying to figure out how to minimize any kind of interruptions to the recording process.

 

I've found that Reaper's delay compensation works remarkably well, once you calibrate it for the interface you're using. But, sure, there may be a little tweaking involved. Or maybe not. If you like the transformer sound, you don't necessarily have to add it in with the direct sound, you can just use it "barefoot," muting the original track and keeping it hand if you want to change your mind (or your transformer).. While not all delay compensation is phase/cycle accurate, other than perhaps strict-gridded drum tracks, it's sufficiantly musically accurate not to worry about it. And remember that hardware DSP monitoring and throughput isn't truly zero latency. With systems like UA's (which includes their drivers) it's a whole lot less than the garden variety USB 2 interface with its built-in DSP monitor mixer.

 

My Mackie 1200F is the first piece of hardware with a built-in DSP mixer, and I was tickled pink with it. At 15+ years old, it's still about the best user interface of any that I've seen - because it looks and works just like a mixer. Still, the throughput delay - analog in to monitor out - is about 3 ms, much greater than the sub-1ms delay that I measured when reviewing a Focusrite Scarlett. And while 3 ms isn't going to through your guitar timing off when doing an overdub, it is enough to cause comb filtering when monitoring a vocal with headphones. I have the working advantage of a real analog console with a patchbay, so when using the 1200F and recording a vocal, I just send the console output to the headphone mix and I have zero (for values close to the speed of light) latency.

 

I guess it comes down to workflow. If you have a favorite transformer and always use it with your voice, you can speed things up by just recording with it. But deferring to a mixdown decision might be the best option if you're not 100% sure you want that sound.

 

There are so many mixing decisions these days - too many options, or too many options used - that one more never hurts. But if you find that a particular mic always sounds better with a Cloudlifter (or some transformer you dug out of an old telphone exchange) between it and the mic preamp input, there's no reason not to consider that a component of the "mic system" for that mic.

 

A well thought out patchbay is a wonderful thing, too, when you have outboard hardware. Unfortunately, for those on a tight budget, it can get to be an expensive proposition with all the jacks and cables, and it takes up a couple of rack spaces in an era where people don't have racks.

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That's an interesting idea...sounds like a product waiting to happen.

 

As I pointed out, there already is a commercial product, though not a very well publicized one.

 

I still think there's an opening for the Rivers Cube. Not everyone wants to buy a 500-series rack just for a transformer, or has the know-how to solder up connectors to the board's edge connectors. Here's your chance to make big bucks!

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Soyuz has a toy called The Launcher, it is a similar form factor to a Cloudlifter and also uses 48v power to operate the electronics.

Up to 26db of gain.

 

There is a transformer inside and a circuit, these components are designed to add color for dynamic and ribbon mics. A trusted friend has one and likes it for some things. I'd like to try one but don't want to spend $200 right now.

It doesn't pass phantom power through so you can't use condensor mics. That is what I like about the Focusrite, it can use condensor mics.

And the transformer sound isn't extreme, it just sounds richer and "better" to my ear.

 

The Presonus Eureka channel strip has a transformer in the input stage and a saturation knob to add even order distortion. I haven't played with that yet on vocals, I liked the saturation on bass guitar. Both the compressor and the EQ have full bypass switching so it could just be a mic pre.

The HHB Radius 40 channel strip does not have a transformer in the input stage. It does have 3 vacuum tubes in it but you'll never get "good" tube distortion out of it, the noise level gets too high. Can bypass compressor and EQ on that too.

 

I still have 2 channels on the front of the Quantum that can be used straight in for mic, line or DI. Plenty of options, will make yer brain hurt!!!!

It took a chunk of my life to get here and I am still not sure where "here" is.
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I still think there's an opening for the Rivers Cube. Not everyone wants to buy a 500-series rack just for a transformer, or has the know-how to solder up connectors to the board's edge connectors. Here's your chance to make big bucks!

 

What kind of connectors should I put on it? For a mic, or for a line level signal? Switched taps?

 

You bring up a good point, though, about not knowing how to build something so simple. I think that soldering skills and a knowledge of basic electricity are some things that people should learn along with learning how to record good. There are always things that need fixing, and understanding signal flow, Ohm's Law, and the use of a multimeter can help diagnose a lot of problems. It's more logical than deciding which EQ plug-in to use and how to set it.

 

I'm trying to review a terribly edited book now called Practical Audio Electronics which I had hoped was another book that I didn't have to write. I had hoped, from the summary and table of contents that it would be a good guide for someone who wanted to get into DIY (do it yourself) audio projects or troubleshooting, but, nope. it's not really going there. Electronic Projects for Musicians is so much more useful, informative, and better written.

 

I should start a new topic about this, or someone else should. How do you learn this practical stuff today? Or do you simply thing it's not important? My father was a tinkerer, so that's where I learned how to solder and build simple circuits. We had a neighbor who had a ham radio station - this was in the days when nearly all ham gear was either completely home built or modified from military surplus gear - so I've had a ham license since I was 12 years old, and I built a lot of Heathkits during my school years. But YouTube, with a few exceptions, is a good substitute for my hands-on experience.

 

There's a groupt out of an AES chapter in Boston that has developed a DIY program. Last few years at the AES conventions, they've had a setup where they teach basic skills like soldering and circuit tracing and there's always a full house. They do road shows, too.

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I've had a pair of Beyer mic to tube transformers for many years. I've long since realized that I'll never use them to build a stereo tube mic preamp. I wouldn't even know where to begin with a project like that.

I read up on them and a mic can easily saturate them, they are small. Supposedly very good frequency response. I have some paperwork with them including schematics and test traces.

Supossedly they were used by Ampeg and Studer.

 

I thought of just hooking the input wires on one tranny to an XLR mic jack, then hooking like colored secondary wires together on both trannies and the primary wires on the second tranny to an XLR output jack.

Feel free to laugh, is that an insane idea? They are just sitting around now.

 

I've also got a slightly larger transformer that says Packard Bell on it, not sure which wires do what to be honest. And a couple of transformers that came out of boxes that were supposed to go inbetween a guitar amp and the speaker and tap off a line out for direct in to the PA. But, boxes sounded like crap, very harsh and thin - so I gutted them and tossed most of it because they were not made to be taken apart, which has never stopoed me!!!!

 

There are all kinds of vintage transformers around but that means a custom product, maybe it's a prototype?

 

As to what a transformer insert goodie box should do and be, maybe a 1/3 rack space box with a Mic in/out and a Line in/out. If possible, one of those could be switched to accept an instrument input. That would cover all bases. If it sells enough to bother, then maybe a dual unit in a full rack space so you could set up dual mono (but matched for stereo). The 500 series lunchboxes are getting lots of advertising, I don't know anybody who has one but I do know lots of people with racks still.

But, I live in a small town so not much of a demographic. Just thinking out loud! Cheers, Kuru

It took a chunk of my life to get here and I am still not sure where "here" is.
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I still think there's an opening for the Rivers Cube. Not everyone wants to buy a 500-series rack just for a transformer, or has the know-how to solder up connectors to the board's edge connectors. Here's your chance to make big bucks!

 

What kind of connectors should I put on it? For a mic, or for a line level signal? Switched taps?

 

The Rivers Cube has an XLR female input and an XLR male output. No switches, buttons, power supplies, nothin.' People just put the magical box between their mic and their usual mic input.

 

The box itself can either be the Apple approach - white plastic with rounded edges, but I'd make it black instead of white - or the funky metal studio box.

 

And of course, the Rivers Cube name opens up various cheesy marketing options, like "Rivers helps your signal flow."

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I mixed my last single with a singer I've worked with for many years. On our 2014 album, the engineer used a Shure SM7B into a Neve 1073 into a Purple Audio MC77 as the vocal chain. I used that sound as a basis (though I think the singer's mic at home is a TLM103). Here was the chain I used and was relatively happy with.

 

Arturia 1973-Pre

Waves SSL GChannel (I go back and forth between this and the Softube Summit Audio Grand Channel)

Waves De-Esser

Waves C4 Multiband Compressor

Waves API 550 EQ (I love how this sounds)

Waves CLA-76

 

I don't remember what reverb I used.

My Site

Nord Electro 5D, Novation Launchkey 61, Logic Pro X, Mainstage 3, lots of plugins, fingers, pencil, paper.

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Waves API 550 EQ (I love how this sounds)

 

It does indeed use a different topology compared to many other EQs. I looked into this and wrote an article for the Waves blog comparing the API-550A, 550B, and 560. It explains what the "proportional Q" circuitry does compared to conventional EQs, you might find it interesting.

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