Rate this topic

Recommended Posts

I totally agree, ronws.

Not only do vocalists change mics during the course of recording an album as you mentioned, I have also noticed that some swap mics during live venues as well. One in particular that comes to mind is Steve Hogarth (Marillion), who uses a Shure (wireless) 58 and the infamous Neumann KMS 105 condenser (wired of course). Others I've noticed are C.J. Snare, Lawrence Gowan, Joey Tempest, and other too numerous to mention. Note : Would LOVE to have the Neumann, but at $700.00, that's far beyond MY budget !!! :mad:

As for recording, personally I don't get involved in that myself. Rather, it's much easier and less time consuming for me to just go to the studio and let the professional sound engineers do all the "work". Over the years, I only recorded one song myself and it took so, so long !!! Compliments of a friend in the computer business, I was "hooked up" with several recording programs, but they were too time consuming for me....

On another note, I can't understand WHY MXL discontinued the 9090 ??? It's a Sweet mic ! With the simple flip of a switch, you can switch from a bright sound to a warm sound.

And that brings me to the Shure Beta 57A. What a difference between the "straight" 57 ! You have to "eat the mic", but it's still a nice addition to my "arsenal".

You keep mentioning the Sennheiser e835.... I think that's going to be my next investment as soon as I get a chance to check it out !!! :cool: And that brings me to a topic that is often discussed here. Beginners take note >> You must find a mic that suits "YOUR" voice. What works well for one singer may not work well for YOU !!!

While I don't coach a lot of singers, I often assist a lot of local (beginner, and some more advanced) singers. After they try several of my mics, I often hear : "Which one worked the best for me again ? I can't remember since we tried SO MANY" !!! :D

Edit : I simply have to add that if it were not for MY coach, my TMV World partner, and my FRIEND, Maestro Robert Lunte, I would not be able to help the local singers. Some of which I would like to add are members of The Modern Vocalist World !!! Thank you Robert for all you have done for me personally, and enabling me to assist others !!!

Share this post


Link to post
Share on other sites

Would it shock you guys if I told you I recorded my entire CD, instruments and voice all on my Shure SM57?

Share this post


Link to post
Share on other sites

Would it shock you guys if I told you I recorded my entire CD, instruments and voice all on my Shure SM57?

No, because you are a cheap old fart, just like me.

No, really, you are a genius.

And certainly, theoretically, if you know how to use one mic, you can do so much with it, as I have been learning. However, in the other books I have read from recording engineeers, they have lots of mics. Which doesn't make them better or you wrong. It's just a thing.

Like I have said before, with the condenser mic I have, I really like the round and warm tone it brings for my acoustic guitar. And, since I have learned to dial the interface and the input mic level in my DAW down from 11, I am getting a better sound out of it for my voice. So, there is that, it is not always the most expensive or prestigous mic.

And, as I have pointed out numerous times, Bruce Dickinson literally made a career with the shure 58, especially on the stage. As in a career spanning several decades. And even though he later talked about using headset mic's, I notice that he stuck the the 58 on the nostalgia tour "Flight 666."

Share this post


Link to post
Share on other sites
This is still and awesome thread, David, especially because it simple and direct and I hope you don't mind that I have bumped it up and that I want to add to it. Not because your posts were not complete but because I still think about this and some thoughts were more appropos here than in my thread about Audacity, though I mention some of it there because it is important, regardless of DAW.
 
I am not a recording or mixing engineer, certainly not a mastering engineer. I am a singer and guitar player and some might say, barely that.
 
But, in this modern digitial age where we all have some way of recording and getting it into digital format, even a simple as video from a phone and shared on FB or Youtube, it has thrust upon us, to some extent, the need to become better at some of these things.
 
So first thing, back to equipment. In another thread, a guy was asking if a USB mic would do what he needed and I felt the answer depends.
 
If you don't play instruments and only will sing over karaoke tracks, then a USB mic will be good. You will have to adjust latency in your DAW. Then, on playback and editing, put latency back to the factory default, otherwise, the shortened time will cause clicks and skips. A good USB mic is usually a condenser mic with a built-in analog to digital converter and it is really good for podcasts, narrations, audio books and interviews, because they are inexpensive but do really well with the spoken voice.
 
If you sing and play other instruments, then you really need some kind of interface. And Scott has mentioned the firewire and you can choose from a few different models, the more expensive having more inputs though, eventually, either from that outboard or an actual 2-bus put across it, your end product is going to be 2 outputs and mixing down to 2 tracks (stereo or mono, whichever you prefer.) And that is really helpful if you plan to record several instruments at once and certainly would be needed in a studio that hopes to operate commercially.
 
But, if you are like me and many others and play instruments and can play one part at a time, you don't have to go any fancier than a 2 in, 2 out interface. The Scarlett 2in2 and the m-track m-audio both do that. Live monitor with a fader dial between live input and usb (which would be playback from previous tracks.)

Share this post


Link to post
Share on other sites
The other advantage with the USB or Firewire interface is that you can plug the best mics and instruments you have because the inputs are for instrument 1/4" TRS cables and xlr cables. And the modern interfaces also have midi in and out, if that is what you are doing. Gain knobs for each of two channels. +48 switch for phantom power for condenser mics. I usually leave it turned off until I plug in the condenser mic. Reason being, I am like to record guitar first. On my interface, +48 is on channel 1. I usually plug guitar and other instruments into channel 2, but I leave +48 off, unless I am actually using the condenser mic.
 
I liked Adolph's list of mics. And I recently got, for 88 dollars from B+H in NYC an MXL V67G large diaphragm condenser mic and I really like how it sounds. Funny thing is, it technically has less range than my Fame CM1. The CM1 brags 20 to 20k flat response. The 67 brags 30 to 20k. But it is not about having the largest range of response, it is about having the right response for where most of the sound is.
 
I don't think I am currently capable of nor will I be of singing something down near 30 Hz, let alone 20. But I notice that the 67 gives me a warmer and darker sound than the CM1. Though that could also be due to me getting a smidge better at mixing the recording. Anyway, the 67 is my preferred mic for vocals but I will keep the CM1 because I like how the acoustic guitar sounds with it.
 
I have 5 guitars but only play 3. The Hondo Flying V (looks like a Gibson with maple finish,) usually through the Roland GS-6 digital effects unit that I bought a few decades ago. It was designed for pro touring. A Spectrum acoustic with built-in pick-ups, volume, 3 band eq. Granted, a cheapie but I make it work. A Yamaha C45M classical guitar. Sometimes, you need the soft sound of nylon strings. Casio LK-165 keyboard. With that,  I can fake some keyboards and play bass. And recently learned I can play drums on that with keys, as well as using one of the pre-set drum patters. And the Kat KTMP1 percussion unit, that I play with actual drumsticks.
 
My recent cover for "Highway to Hell" was recorded with the Flying V and Roland, and the Casio. Though I was using the CM1. And instead of a pop filter, I literally pulled a (clean) sock over it to cut out air and plosives. And the most important thing for that is I dialed down the input gain until high notes were in yellow. Granted, still a chance of distortion but I was going for something as close to the original recording as I could manage, minus the mixing abilities of Mutt Lange, who recorded and mixed that album.
 
And that's another thing I have found. Adjust your mic for each song. Some sungs are sung more loudly and you may adjust mic proximity. Others are softer and quieter and closer and you even bring up input gain.
 
And Owen is right, a recording is as much a product of the mixing and mastering stage as it is the brilliance of the musicians.
 
That being said, and I know my brother can attest to this, the best recordings and albums you have ever heard started with great performances. Felipe has also said this. When you take a pile of trash and mix it, you get a shiny and echo-y pile of trash. When you perform stellar, and record it clean, mixing is essentially a judgement call of placement in the stereo field. Essentially, great performances mix themselves.

Share this post


Link to post
Share on other sites

I wanted to share this, since I am doing more and more recording these days... thought I would share what Im learning and using. 

It has taken me some time and trial and error, and private lessons from a producer to get a great home recording chain going. Here is what I use to track all my vocals and comp., prior to being mixed at a professional studio with a console.  Let me know if you have any questions.  

You can get most of this gear at Amazon.com if you run a search at top right of the web site, or click on The Vocal Gear Store

 

Microphones:

Pearlman TM2 (Tube Mic) - 1st Choice

RODE K2

RODE NT1

RODE NT1A

Electro Voice Cardinal

 

Headphones:

Extreme Isolation - X-29s

 

PreAmps:

Focusrite ISA One

Universal Audio 720

Share this post


Link to post
Share on other sites

Thanks for pinning this, Robert. I had thought about it but I did not want to be presumptuous. Though, of all the recording threads, this would be one that needed pinning, in my opinion.

Share this post


Link to post
Share on other sites

A few things that I find helpful to keep in mind when recording, to avoid getting in your own way:

 

1 - What you are doing is called "tracking":

The making of a song involves multiple tasks, from actually making the song (or at least a sketch), learning it properly on your instrument(s), getting the interpretation down in whatever way you want to do it, tracking it, pre-mixing, mixing and mastering it. So, its multi-disciplinar. Its knowledge about music, the instrument, maybe writing, mixing, audio and even electronics sometimes.

So, do ONE thing at a time. You may use the tools to study, to make your song, of course. But its important to separate those things in your mind. If you want to be badass and just sit down and record your magnus opus out of the blue using 30 minutes of a sunday afternoon, you will need to be even more badass on each of these individual tasks. Some people are, that is indeed true, and they will make it look easy. But if this was your case you probably would not be reading this.

After a certain point you will be able to deal with 2 of these going on at the same time, yes, and as you do, you will have more creative possibilities, but its in my opinion overwhelming to do so while you are still getting your way around it. And even if you can, it may be wiser to step back and break it down. Its not a circus show.

Because of this division, we know that to be ready to start tracking the vocals, you need to know what you are going to record and have at least an overall idea of how it should sound like. Its not the time to worry about getting your singing together.

 

2 - Setting up for tracking.

The thread already gives some very good advice in regards to mic gain. I will give some practical insights on how to set a session for tracking.

Now, your DAW, be it whatever DAW it is, must have some functions that allows you to make a submix, so that you can with one fader knob control the resulting levels (without changing the mix itself). If you are recording with a finished backing track, its simple since its just one fader. However, if you have more instruments recorded on a pre mix, create a subgroup and route your mix throught it. In both cases, you will have a single fader to adjust the levels of the instruments.

Now, the trick here is this: make it a habit to work around -12dB PEAK inside your DAW. This means, that you will turn down the levels on this fader, be it the submix or the backing track, until you see the green light peaking at -12. Precision is not necessary, visualy gauge it, and set it around -12db. Why? There is a good deal of reason why this is a good idea, but, not doing so will make your life really hard when setting the microphone levels.

Once you got this down, its time to set your monitoring levels. The master fader on the DAW should be at 0dB. Adjust the volume of your headphones or loudspeakers on THEIR volume knob, NOT INSIDE THE DAW, keep the -12db peak on your submix. How loud? For tracking, its ok to be a bit loud, can be more exciting to record with louder sound, just don't overdo, if your hearing gets tired too soon you risk not finishing the job.

And now, with monitoring levels defined properly, and a good headroom margin, you will adjust the gain of your mic, bring the gain up slowly until you can hear yourself well against the instruments, and do a few tests. Using the -12dB rule you will probably be safe from cliping and while being able to hear yourself even with heavily compressed songs.

This is a good rule to remember, you will only want to bring things up near 0db when preparing for release. Its important to leave the headroom so that you don't clip your audio interface I/O (even if the DAW can handle internal floating point processing). On digital recording, 0dB is not the "normal" level, 0dB is absolute MAXIMUM volume your system can produce.

 

3 - Tracking

With the stage set, time to record the song.

Keep a copy of the lyrics near you, plenty of water, set the mic height to comfortable level, and, find a way to control your DAW properly. Its too tiresome if you have to walk though the room, move a chair out of the way, and jump over the bed everytime you need to go back and redo a take. If your setting is too cluttered and troublesome, there are APPs for Iphone and Android that allows you to create a control surface through MIDI, they are very handy (Touch DAW is one of them).

Also, focus. Lock the door, ask people to not disturb. Your concentration should be on what you are going to do, if you are distracted by non-sense all over the place, you will mess it up and its REALLY tiresome and frustrating. Specially when you are tracking your own stuff, you are your only reference, and is easy to be distracted and forget what you were aiming to do, or lose track of the mood, etc.

Normal overdub is quite straight forward, you will hear the track, and sing to it. The key here is to pay attention to the damn thing and sing with it, not all over it. If you have a problem with tempo, you should not be recording anything, you should be studying with the help of your friend the metronome. I mean it, its not a joke, rhythm is the most fundamental notion of music, working with some simple exercises like clapping your hands to specific time divisions and note durations can have a much more relevant impact on performance than technique itself.

You can also work with take lanes, and do punch in recordings. Take lanes means having multiple recordings of the same track you can choose from and combine, and punch in, means recording over a single section of the song that you want to remake.

 

First about takes.

I know that the first thing we may think is: "I will just record 80 takes and then find the best pieces and do this awesome master blaster final version." And then when you actually do it you will realize you recorded 80 meaningless versions of the same thing and they are all equaly not fit, becoming actually worse when you build your frankenstein.

You can use the take-lanes of course, but do NOT make a meaningless take lane. Something that is not sounding how you want should be DELETED. Use the take-lanes to record viable OPTIONS, not trash. Trash you discard, options you keep. Remember that! And learn right now to make choices, after you do a take and ask yourself "Do I want to keep this one?", if the answer is no, don't be afraid, delete it. Sometimes is not easy to decide, but decide anyways.

 

Now about punch ins:

Punch in recordings require a certain technical control when it comes to vocals. If you change the voice quality too much, it will sound weird. Even the phrasing may sound unnatural depending on what you do. I recomend not doing punch-ins on vocals recorded on different days if your technical control is not very defined, and, to do it and avoid problems with phrasing, you can do the following:

- set your daw to auto-punch in (it means that you will choose the area you will re-record, and the DAW will ONLY record that specific section)

- set the DAW to come playing from the last phrase before that section;

- sing along yourself on the previous phrase, continue singing on the section you want to record again, and sing at least the next word after it.

This will make sure you don't take unnatural breaths or make strange attacks during it. If it sounds unnatural, trash it and just record a larger section.

 

4 - Evaluating the quality of the recording

This should be going on as you track your vocals.

If your gain staging of the microphone was good, you should have a good level on your DAW already. If necessary, ajust the levels INSIDE THE DAW to bring it to a nice level against the instruments (don't change the gain or the instrument levels during the rest of the tracking session). Now, depending on your taste, you can set some compression to ballance levels (specially if you are using soft dynamics on loud passages of the song), but, DONT OVERDO the effects, keep it raw and simple.

Why? Well because you will be making decisions on quality, the less you tamper with your ability to hear what you recorded, the better your decisions will be. If you get a raw version of the vocals that sounds nice and interesting against the track, it will be a breeze to mix. If not, well, you are screwed. It COULD work if someone else was doing the mix, but if it will all be done by yourself, if you could not make the call now, you wont be able to make the call later.

Also notice, when you are tracking your own vocals, DECIDE FOR QUALITY. Don't decide for "high notes", dont decide for "I studied long so I should be able to sing this good", dont go for "I will keep this distortion here because I wanted it to be more agressive". Quality first. IF there is good quality, then it is a viable option, if not, trash it away and do it in another manner where you can have good quality.

What is good quality? I have no idea, its your taste that you are going to use, so use it. If you need a reference, pick a favorite recording of yours and play it on another system (if you play a mastered rec on a system set for -12db peak monitoring it will be painfully loud, so dont), and listen. Does it sound good? Cool, what you want is to play yours back and it should "sound good" too.

 

5 - Set a deadline

You can pay a producer to press you and keep you moving forward, or, you can do it yourself and aim to get the job done. So set timelines. 2 hours to track it? Cool! Then you will keep track of time during your recording session.

But it doesnt mean that after 2 hours you will just stop it midway and leave it to another 2 hours session, and another, and another. No. It means that you will start tracking and will be making definitive decisions about it, so that at the end of these 2 hours, you have a track done. Really 2 hours, is PLENTY of time to record vocals for a song, if you know what you want and if you don't waste time.

How so? Well, thinking of a vocalist recording him/herself. Lets say that you reach a point on the song where there is a high note that is not working, or an effect you cant get to sound how you would like. What do you do? Decide! Change the line, or do it on another manner that also sounds good and move on. "Oh but I am studying to be able to do it". No, you are not studying, you are tracking vocals (its another job!). And keep going, solve the problem with what you are able to do, and move forward.

Understand, when it comes to the control of the instrument, if you keep trying to use things that you still cant, you will never have a reliable performance. You can grow to be better than all singers that ever lived, but if still try to use something that is outside what you can do, you will be always setting yourself for failure. Its impressive how often people do it, and I understand perfectly how easy it is to build this trap, its natural because of the focus on self-improvement. But again, its a different job, you are NOT studying. Its singing.

AND DO SOLVE THE PROBLEM! Yeah, if you hear something that you don't find acceptable, fix it. Dont overlook or skip it.

 

I hope this helps someone. The structure I mentioned up there is the usuall workflow you have when producing a song: pre-production, tracking, pre-mix, mix and mastering. Its important to understand that the result you are aiming to get is usually done by more than one professional on these areas, sometimes 2 or 3 of different areas of the process colaborating together, and their job is much more critical than technical to be honest (critical as in, making calls about stuff, from mic position and gain, to take quality and mix levels) .

Dont make the mistake of thinking that you will just sit down in front of the computer, turn a few virtual knobs here and there, load the "soundgoodlizer VST plugin" and the magic will happen. Organize your own thoughts, and understand all these situations are specific and have different needs and a specific knowledge involved, you will probably get a much better result.

In fact you will probably learn more about your own needs as a singer: what you really need, instead of the things you think it would be cool to do.

Share this post


Link to post
Share on other sites

Looking at the past to understand the present:

DAWs are pieces of software designed for music production. And being so they were modeled after the existing solutions both from a practical standpoint, as well, as to allow the professionals involved with the craft to transition more easily. I find that without looking at audio gear, the notion of a "channel strip", "tape saturation", and even effects like Reverb may not make much sense.

So lets start a trip.

Mechanical audio recording.

Around the end of the 19th century and the begining of the 20th, some very clever guys devised that by attachig a stylus to horn in a mechanical manner, they could engrave soundwaves captured by the horn on a form of media. At the begining, they used a Wax cylinder, later the recording was done on vynil discs. Both the process of recording the audio, and reproducing it was mechanical.

The Wax type was the Graphophone, take a look at the monster:

bt2.jpg

 

And this is the Wax cylinder it used:

cy-b.jpg

 

The vynil system was known as the, more popular, Gramophone:

gramophone.jpg

And here is a vinyl, which some of the youngest folks may never have seen :)

Vinyl_record_LP_10inch-1024x768.jpg

 

Now, for those trying to understand this, mechanical means that this pre-electronics, it was done with clockwork engines and clever machinery. There was no microphone or loudspeakers. And producing copies was a very complicated process. In fact, on the early stages, the master recording needed to be destroyed on the process of duplication, and there was significant degradation each time a ! So here is how people would do it: they would line as many recording devices they could in front of the artist, a full band generally, and they would ask the guys to sing the song over and over, to produce as many masters they could.

So, its unnecessary to say that the quality would vary greatly even between the masters from a single section, since each device would be at a varying distance and angle.

With vynil the capability to duplicate improved considerably, and so more care on the recording process itself was possible, but for a long time, the recording process remained mechanical.

I've heard an interview of a singer, done in 1965, about a record done on1915 recently, and its fun to hear the singer say how the "modern" process (modern here being 1965!) made life easy for singers, and how on 1915 people need to be *true* singers to produce a record, because, from his words, anything other than full out loud would not be properly engraved by the mechanical process.

 

So the first stages:

Mechanical process all the way, from recording to the end listener;

Poor duplication capability;

No means to monitor or evaluate quality during the process, the bands/singer would play, and that was it.

Poor quality control of the end results;

Poor fidelity to the original audio.

No control over the audio already recorded in any manner, except maybe by manipulating acoustics on the playback.

 

In here we have the concept of "master" defined. A "master" was the original recording that would be used for duplication. Although "mastering" makes no sense in this realm, since no control over the audio recorded was possible.

 

Think about these machines, which were revolutionary at the time, and think about what you can do with your cellphone in regards to audio recording... :)

This is a very interesting site to visit and have a listen to wax in all its explendor:

http://cylinders.library.ucsb.edu/

Share this post


Link to post
Share on other sites

Electronics and Audio

With the advent of the electron tube, valves, new possibilities in regards to audio started to appear.

With the use of microphones to convert the sound into electrical signal, and then electronics to make the final cut in the disk, a lot of the problems of the mechanical process were solved.

On mechanical recordings, the coupling between the open air and the stylus was done by a horn, which added undesired resonances to the sound. There was also the problem of the necessary sound pressure to actually move the stylus, and the frequency response of the mechanism that was also limited due to the physical construction of the device.

A microphone avoided most part of the problems of a horn, and the coupling between stylus and sound was now done with amplifier circuitry, something nearly impossible to do in a mechanical manner at the frequency that is necessary for audio.

With the use of electronics to record, it was also possible to have more control over the process. Ajusting gain, levels, using more microphones and mixing their signal.  But there was still no real editing features. Electronics + vinyl duplication improved the audio quality a lot, so much that for a while the released media was "held back", limiting the frequency responce of the recordings, to not overwhelm the mechanical devices that the consumers still had.

With electronics, the following was possible:

- Use of microphones to transform sound into electric signal;

- Control of levels and gain;

- Control of tonal quality;

- A more precise cut of the vinyl disks.

 

The core of this revolution, the electron tube, or valve, looked like this:

electronic-valves-19143155.jpg

 

These little guys deserve some explanation since a lot of the effects we use derive from the way they function.

An electron tube, or valve, works somewhat like a water valve.

Lets say that you are playing around with a water faucet. To make it easier to model it, its one of these quarter turn faucets. If you turn the knob, it opens and the water flows, if you close it, water stops.

Now, if you have huge amount of water flowing on a really large pipe, with a lot of pressure, this small movement you do on the knob, will control a much larger amount of energy on the water flow. You need much less effort to turn the knob, than to block the flow of water on the pipe directly with your hands. If you could turn the knob fast enough, and the water also moved fast enough, you could actually modulate sound by opening and closing the faucet, but we would need to be The Flash for that.

An electron tube does just that. A small, weaker electrical signal controls the "knob", and the knob controls the larger power that is flowing through another piece of circuitry.

Ideally, if our faucet was magical (note, magic, not perfect), if you wanted more flow of water, you could just keep opening it endlessly, and the more you open the faucet, the more water would flow. The reverse would also be true, you could keep closing it and after a certain point water would begin flowing back into the pipes.

That however does not happen. obviously. A valve, when totally open, is said to be saturaded. Now this is something you probably seen on effects correct? This is why I am talking about this boring subject.

But a valve besides not being magical (they cant open more than totally open), are also not perfect, valves are just human :P When you get NEAR saturation, the way they behave becomes non-linear, it means that the flow on the pipe does not change in a linear manner as you turn the knob anymore. And this produces a certain kind of distortion.

In this specific kind of distortion, the peaks of the signal becomes less pronounced, which introduces harmonics and, changes the peak levels, compressing the dynamic range.

So now you know where these basic terms come from, and what inspired a number of effects that are based on it. It does not mean that a compressor works with valves, not really (some do), compressors simply mimic this behaviour that, someone on a given time noticed that, besides being a technical problem, could very well be a creative tool and produce desirable results on a recording, improving, or even, creating content.

Share this post


Link to post
Share on other sites

Magnetic Tape Recording

The audio recording on magnetic tape was developed by the Germans on the 30's and became avaiable to the rest of the world after the WWII.

Basically, the process consisted on using a tape to record magnetic signal that could be later read and reproduced.

Over the years, the quality of the recordings improved gradually, by modfications on the process and the technology used to do it, but the base process was the same until the digital era. In fact, even after the digital era many continued to use tape recorders on the studio.

As the recording techniques evolved and technology improved (including the solid state devices technology), new options based on tape recording started to appear. One of them being the 4 track recorder:

ampex_440_vintage_tape_machine_f.jpg

Now, it may not look like much when you can open an app like Audio Evolution on your phone and record 16 tracks or more at high quality, but, when all you had before was a single go at the vinyl master, being able to record 4 separate microphones and then mix their individual levels as you saw fit, was revolutionary.

With time this improved, and eventually 16 and even 32 track recorders became avaiable, with very high audio quality.

 

But on the middle of this technological development, a guy, called Les Paul (yes, the same dude of the guitars) devised a very particular technique that came to be known as Overdubbing.

Mr Les Paul was a very smart guy, and he developed tape machines that could read and write individual tracks on a same tape, allowing him to record in an assyncronous manner. It means that Les Paul would record a bass line on track, rewind the tape, set the bass track to play, ARM track 2 to record, and record a guitar line while he heard the bass recording he did previously.

 

So the tape had several advantages over the direct vinyl recording:

The quality of the recorded audio was very good;

The tapes could be easily edited (by cutting and gluing it back);

Tapes could be reused;

As the technology improved, it was possible to gradually add more tracks to a single tape, this was the advent of multi track recording.

With multi-track it was possible to make a mix AFTER the tracking process.

With overdub it was possible to construct a record one layer at a time, a technique that is basicly what we all use nowdays.

 

Tape still had some issues:

The tape degrades with multiple uses;

Because of the way the media works, there was a high frequency noise associated with it, known as tape hiss;

Because of the noise associated with it, making copies progressively degraded the quality of the signal;

Because of the different media, in order to produce a vinyl from a studio tape recording, special processing was necessary in order to allow its recording to the vinyl with high quality. This process of preparing the audio to produce a master copy, was called mastering. It involved, amongst other aspects, dynamic range compression and special equalization to fit the song to the technical limiations of the vinyl.

Although editing was possible, there was limitations to what could be done with the tape.

 

I will follow with the description of some of the gear that was used during this era of analog recording on tape, which lasted until the 80s and even 90s. Much of the structure we use on a DAW comes from this.

Share this post


Link to post
Share on other sites

Pre Amplifier

Microphones are devices that translate sound into electrical signal. And they do a good job at it. However, the signal they provide has no power to perform a more demanding task. If you connect a microphone on a loudspeaker, nothing will come out. Even if you plug it directly into a mixer "line" input, it will lack the power to adequately "move" the mixer circuitry.

So before you can do anything about it, a special kind of device is required, an amplifier to give this signal adequate power, so that you can work with it. The name of this device is pre amplifier.

A pre amplifier has several jobs:

- Integrating ballanced line signals;

- Adequating the signal level (gain);

- Matching impedance from the source (microphone) with the load (mixing console);

- Providing a power source for the mics that require it (phantom power).

 

Today, with around $10, you can pick a high performance IC, a couple of resistors and build a pre-amplifier that outperforms anything that existed until the late 90s. With near flawless technical specs.

burrbrown2134.jpg

 

 

Back on the day, this was not the case. The input section of the pre-amplifier had wire transformers to deal with the ballanced signal and impedance (lots of dynamic mics like the SM58 still use it), the circuitry was mostly valve based with all the problems valves had (high voltage, hum noise, capacitor coupling, etc). So at many times, the circuitry was not fast enough to deal with the full range of the audio spectra, there were compromises that were made, distortion levels were significant, high noise floor, high cross-talk, non-linear frequency response, etc.

Depending on how a pre-amplifier was designed, it would have a given sound. The way the transformer was winded was enough to change it! So pre-amplifiers, besides their technical role, imprinted a characteristic sound color on the recording.

 

384614d1392271672-have-questions-vintage

 

And it continued to do so until much later, when the technology allowed using solid-state components and clever circuit design that applied a lot of negative feedback and straping techniques, making the response very linear, very low distortion and very low noise floor. Suddenly, pre ampliers had no mojo to them, no color, no distortion, just, functional (doing their jobs as well as it can be done).

Understand, this is boring, for a while it worked, but very quickly people started to look for the old gear to get that sound they could with it. Because sounding good is more important than perfect technical performance! And new gear was developed trying to get the best of both worlds, selecting just the characters that they wanted. Some of the most awarded pre-amplifier designs combined solid state with transformer coupling, etc. And these award-wining pre-amplifiers set the standard to what the audience expected to hear on a recording.

M5F.jpg

 

With this we can understand why we have so many VST plugins around that attempt to simulate how a given mixing console from the 70s sounded.

 

waves_ssl_g_channel.jpg

 

I advise a simplistic view however. Before you can deal with the functional side of the tools, using them to just do their job, its not wise to start looking for different sounding tools. Think of a painter that is learning. You have different kinds of pincels, of course, different sizes, thinkness, and this matters. Also you have a multidude of colors to choose from. Ok.

Now, if before the painter can apply the painting techniques properly, he starts to lose his mind on paint BRANDS, and start looking for the ultimate pincels that have the perfect weight, made with the golden hair from fresh captured mermaids, in order to solve a problem where his sun looks funky, this will become a quest for magic really fast.

 

Its the same with pre-amplifiers simulators and coloring. If you must use a simulation, choose one you like and just stick with it. If possible dont even leave the choices avaiable for you, clean up everything you wont use.

 

Pre amplifiers also may need to handle sources with much higher impedance like an eletric guitar. They may be specialized like guitar pre amplifiers, DI-boxes and guitar effects pedals. Or they may be general and have a hi-z switch. Hi impedance allows the pre amplifier to be sensitive to very, very low power signals, just touching a guitar cable will be enough to produce a response on the circuitry. However, this is also a downside, since it also picks up noise much better.

Share this post


Link to post
Share on other sites

Mixing Consoles

Like everything else on audio, these evolved considerably, lets have a look at this fine piece of gear:

TechNotes_01.jpg

Now, remember, we are looking at this to understand evolution, I am not saying in ANY way that this is better than current technology.

Looking at each channel from bottom to top, there is an echo knob, to add the echo effect (which I have no idea of how it sounded), the larger knob apparently is the channel volume control, then some lights that seemed to indicate wheter the pre amp section was set to deal with mic or line levels, two knobs apparently for equalization of Lows and Mids, and a panning switch for Left, Center or Right, which indicates stereo capability.

On top of that another knob+switch that I would assume are the controls of gain and Line/Mic switch of the pre-amplifier.

Now, if you observe it well, you will see that each channel is an individual pannel attached to the system, its modular. This kind of construction was the default and still is the default on analog and even digital mixing desks. A channel of a mixer is called a Channel Strip. These of course does not quite look like a strip just yet, but, lets see a more modern one:

http://www.solidstatelogic.com/support/images/b-stock/large/629911XS.jpg

This is one channel of this larger console:

xl9000ksmall1.jpg

 

 

Of course, the channels of a modern SSL mixer have WAY more functions than the that old console, parametric EQ, compressors, auxiliar sends, automation, integration with other systems and devices (you can control a computer DAW from the central part of the console, thats a control surface), programmable effects, etc. But, its still the same concept, and more importantly, you will notice that DAWs are usually inspired on the same modular design idea.

The main job of a mixer is basicly summing multiple electrical into one or more resulting signals. The level of each channel is controled by a knob or fader, and on modern mixers we have an opportunity to add a considerable amount of effects to the signal, changing its frequency response, dynamic range, adding effects, etc.

Some common functions present on most mixers:

Pre-Amplifier Gain.

Assuming that you are using the internal pre-amplifier of the mixer, you will have a gain knob which you can use to ajust how strong the signal is being supplied to your mixer. On analog consoles, it was a very good practice to keep the signal loud to make use of the pre-amplifier circuitry to add enough volume and keep the noise levels as low as possible. A louder signal would increase the signal-to-noite ratio on all sections of the mixer until the end of the signal chain.

Too hot would of course lead to distortion, which on the analog console could not be that bad depending on what you wanted to do. The mixers would give some visual cue about the signal levels so that you knew when you where too hot or too quiet, a VU meter could indicate it, or a led could lit when the signal was too strong, etc.

Insert.

Next, if present, you probably would have a insert point. Inserts are "detours" on a channel, to which you can add an external effect by means of pluging a cable (P10 TRS - P10 TRS usually). The insert sends the signal coming from the pre amplifier through the cable and, when there is a cable pluged in, disconects the pre amplifier from the rest of the channel. The rest of the channel is then fed by the return signal from the cable that is pluged in the insert. Useful to plug an external compressor for example.

Dynamics Processing,

Not common on all mixers, but if present, dynamics processing is usually the next section in the chain. Its where you will find compressors, gates and expanders. The amount of flexibility and options vary with the mixer of cours.

EQ Section

Following the Dynamics, Equalizers are usually present. They come in various flavours, the most common is a semi-parametric configuration, with two knobs adjusting low and high frequency gain, and a pair of knobs that allows you to choose a mid frequency and adjust its gain. Some are totally parametric, some offer 2 middle frequency options, etc...

AUX Section

This is part of the internal routing of the mixer. Mixers can have Auxiliary channels where you can have a submix going on, and each channel allows you to send the signal of that channel to a given Auxiliar channel.

Auxiliar channels can be used for a number of functions, from creating an individual mix for your reverb, to dynamics processing and grouping instruments together. The options and number of auxiliar channels vary greatly depending on the equipment.

Pan

Next you will have the control that allows you to vary how much of your signal will be mixed to the master left or right channels. Pan controls are straight forward in their use, but, not so much when it comes to the way they work. As long as you are not trying to compare different equipment and programs, you can ignore the technical side, but, if you plan on doing so, I recommend studying about panning laws, because they affect the signal being mixed in a non-linear way.

On surround mixers, this section can be quite complicated, unfortunately I have no experience on it.

Fader

The main fader of the channel controls its level on the mix. There are many kinds of faders, some with longer excursions which allows you to have more precision, and some with automation. Automation means that during your mix, you can set the mix to record the fader movements and the mixer itself would recall it during playback, allowing you to program changes of levels dynamically through a mixing session. Automation is a concept that became part of most DAWs on the market and not only they mimic their analog counterparts, they are also made in such way to integrate with physical devices.

Besides the power this gives the engineer, its really cool to see the faders moving on their own ;) .

 

 

These are of course common controls present on most mixers, there are many, many more options that allows you to make small choices, like, are you sending the signal to the Aux channel before or after being processed by the Equalizer? Before or after the fader? You can also have a button to reverse phase, turn on Hi-Impedance on the pre-amp, Arm a track for recording on the Recording device whatever it is, Solo a channel, change the routing of the main fader, etc...

 

Share this post


Link to post
Share on other sites

Reverb

When we are on a given ambient, all sounds we produce suffer reflections and decays, they echo back at you, their frequency response changes, standing waves are created. And our brain uses and expect all this information, being able to do very impressive things, like telling where a sound source is approximately, infering the room size, even distance based on it.

We do not perceive the natural reverb of a room in normal conditions though, not if you are not trying to pay attention at it, or if its the case of a very *live* room (a room where there is a lot of reflection and long decay). But its there. Even when you are on an open environment, you get some reflections from floor and from obstacles.

Consider then what happens when you pic a microphone and place at 2 inches or less from a source of sound you want to record.

That source will be very, very loud in comparision to the rest, and almost no reverb of the ambient will be present. Which is not in itself a bad thing, after all, the reverb of the room may not be pleasant.

However, this creates a problem. When you playback something recorded in this manner, the result you get is very unnatural. Specially on mono on headphones, a singer for example, it would sound like the person managed to, somehow, sing the same line directly into your ear, very close, on both ears.

So the solution is to add reverb artificially to the recordings, emulating a space where the sound is being played back.

Lets check some solutions and how they evolved along time:

Echo Chamber or Reverb Rooms:

Hallraum_TU_Dresden_2009-06-21.jpg

This is a very straight forward method, perhaps it may look like an overkill, but its still one of the most natural ways to deal with it. In theory its simple, you build a room with the reverb response you want, place one or two loudspeakers inside it, place on or two microphones inside it, and send the signal you want to be reverberated to the loudspeaker. The microphone on the room will capture the reverb, then you mix it as you see fit.

It also allows precision on modeling, since you can build a room thinking of the ambient you want, instead of trying to calculate the parameters of the reverb itself.

 

Plate Reverb

emt_140ts_pic-534x670.jpe

 

The plate reverb is also simple, but its more of a device than a natural effect.

Straightforward idea. A thin piece of metal is suspended with tension applied, then this plate is excited with the signal you want to reverb, and, two contact microphones on opposite sides of the metal place capture the vibrations. Because of the irregular vibration and the high frequency resonance of the metal, a stereo effect is created, and a very specific timbre is obtained.

Plate reverbs are still used, since they are quite straight forward devices.

 

Spring Reverb

tank.jpg

 

Probably the most cheap and simple of all the devices, the spring reverb dates back to the 40s.

A set of springs is held with some tension in a space, usually called "tank".

At one end of the springs, there is a transducer that excites the springs with acoustic energy, at the other end, another transducer that pick it up, pretty much like the plate reverb.

The spring reverb is still used widely on guitar amplifiers and other instruments that depend on "vintage" sound, a hammond organ for example.

 

Analog Reverb

rsrvb-front.jpg

Unlike the other examples mentioned, these units aimed to simulate the reverb characteristics by means of electronics. Adding delay, controling decay of sound and equalization. Some units had specific settings for a given situation, something like a "preset" that aimed to replicate one or another ambient, like a chamber, or even a plate or spring reverb.

 

Digital Reverb

PHOTO-49-AMS-RMX-16.jpg

 

As digital processing became more powerful and capable of handling real time audio processing, the possibility of using it to simulate ambients appeared. A digital unit could control much more aspects of the sound in regards to time and frequency response. This flexibility and the ability of easily comparing the response from the unit with a real situation to fine tune it, allowed a much more realistic processing and, the units could use memory to store and recall user defined parameters with ease, as well as the built in presents.

Some of the older Digital units, in special the Lexicon Reverbs are still used.

 

Convolution Reverb

ir-live.jpg

 

With the increase of processing power on modern computers, it became possible to use algorythms to, instead of trying to model "by hand" the response of a given type of reverb, use a sample done in a specific manner to integrate back how the response of a given room would be.

The full process consists of:

A setting is made, very similar to the first 3 devices shown, where sound is played on the setting and capture it through microphones.

Then, a signal sweep is fed into the setting, and its result is recorded. The recorded result is called "Impulse Response", or IR file.

This IR file is then loaded on a device or plugin capable of processing it and reconstructing the Reverb of the room in question.

 

IR response files are being used currently for high quality reverb simulations, and even gear simulations, since you can use the same technique to capture the response of a pre-amplifier or a guitar cabinet.

Share this post


Link to post
Share on other sites

I wanted to add to Felipe's posts that some plate reverbs don't use mics but instead use transducers, pick-ups like you would have in a guitar and these pick-ups register the fluctuation of the metal plates the same way they register fluctuation of a metal string on an electric guitar. The delay (reverb is a short delay) is the amount of time for the signal to transport through the plates.

Spring reverb, which I have had on an amp, essentially uses a spring as a "choke" which retards the path of the signal. I could give you the formula for inductive reactance (apparent ac resistance in a coil) but it is better to just think of the sound having to go through that long and winding street in San Francisco while the dry signal took a straight route.

Share this post


Link to post
Share on other sites

Create an account or sign in to comment

You need to be a member in order to leave a comment

Create an account

Sign up for a new account in our community. It's easy!

Register a new account

Sign in

Already have an account? Sign in here.

Sign In Now