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https://youtu.be/Ex85UBvDrgQ

I watched a Sound on Sound brief history of Capitol Records in Los Angles, CA. Now owned by Universal Music as of 2012. First established in 1942 by Johnny Mercer, Buddy DeSylva, Glenn Wallichs. These 3 men owned and operated a small studio on Sunset and Vine in Hollywood.
By the 1950s Capitol Records was one of the largest Record labels In the country. It was around this time when  construction of the  main headquarters, in the shape of spherical tower with a spire on top, was started.
There are 2 main studios in the tower. Studio A is a large, fully featured with the industry standard and vintage stalwarts Like multi-track tape recorders and analog sound processing and preamps as well as 2 Neve 8068 used as 1 console. Studio B remains largely original from original to the 50s 60s 70s.
In my past exigence doing home recording with an old band I was in. I recall trying to mimic the live reverb chamber technique that was invented at Capitol by Guitarist extraordinaire Les Paul. We ended up having a small 1 10 bass amp play at one end of a long stairwell and using a pair of small ribbon microphones in a blumeline configuration at the other end.
While I know I might never get to work at Capitol, I'm still very much inspired and take and use ideas from them and their artists .

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SOS Visit the world-famous Capitol Studios in Los Angeles

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Carlos Padilla Assignment Mic ID Chart (MUSX 125)

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Recording Vocals in your Home Studio - Part 1: Getting Started

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Recording Vocals in your Home Studio -  Part 2: The Perfect Vocal Mic

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Recording Vocals in your Home Studio - Part 3: Comparing Different Mics

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How to Record Acoustic Guitar with only One Mic - Part 1: Getting Started

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How to Record Acoustic Guitar with only One Mic - Part 2: Microphone Positions

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How to Record Acoustic Guitar with only One Mic - Part 3: Hear the Sound of Different Microphones

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Before hitting "record" in your DAW, you'll want to make sure you have audio routing configured. Sometimes, this is automatically set - some DAWs and/or interfaces make for a plug-and-play experience. Regardless of whether things were automatically configured or not, it's good to check these settings to ensure they are configured correctly.

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Knowing where things are located in a DAW is important for an efficient workflow - you'll be able to produce faster if you can get around your DAW with ease. While each piece of software has its own design and organization elements, some tools and features are common across all. Listed below are a few common useful features found in a DAW.

Common Elements

Workspace/Tracks Area - This is where active audio content is located. This is where you'll find your recorded audio content.

Transport - This is where you control the playhead (Play, Stop, Forward/Back, Record, etc.). A display showing time or project details will be found here as well.

Track Info Area - This is where you'll find info related to a selected track or all tracks. This is usually where levels can be adjusted or signal processing can be applied.

Editor - This is where you can edit selected content. For example, a selected audio file/region/clip can be edited, manipulated, or fine-tuned easily within the editor window or area.

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a "sweet spot" between -18dB and -10dB. This is generally a good starting point, but it's ok if you deviate a little.  go somewhere between -9dB and -6dB peaks (instead of the suggested -10dB). 

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a "sweet spot" between -18dB and -10dB. This is generally a good starting point, but it's ok if you deviate a little.  go somewhere between -9dB and -6dB peaks (instead of the suggested -10dB). 

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Time Signature

Most popular music sticks to the typical 4/4 time signature (pronounced "four-four"). This means each bar/measure will have 4 beats/pulses. Most popular music heard today is in 4/4. There might be situations where another signature is required. Here are a few examples:

3/4 is common for a waltz-style beat (like "ballroom dancing" with a Boom-chick-chick, Boom-chick-chick beat)

5/4 is less common, but not unheard of (Dave Brubeck made this time signature famous with the jazz hit "Take Five." You can also hear this in the "Mission Impossible Theme" by Lalo Schifrin.)

7/8 is also less common, but found in action film scores and heavy metal. For a time signature like this, the rhythm is often broken into smaller pieces that equal 7 (try counting these proportions 2,2,3 or 3,2,2). A recent example that features this signature is the distorted electric guitar theme for "Wonder Woman" by Hans Zimmer and Junkie XL.

7/4 is used in Pink Floyd's "Money" and Peter Gabriel's "Solsbury Hill."

5/8 and 7/8 alternate in the theme for "Fantastic Beasts" by James Newton Howard

4/4 and 3/2 (or slow 3/4) are mixed in Hozier's "Take Me To Church."

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As the name or category suggests, these types of instruments use electricity to make sound and are made louder by an amplifier. There are a number of reasons for this, but one major reason is to amplify the instrument. Some instruments are unable to produce a loud enough sound, so they are boosted using a small microphone pickup and an amplifier. Acoustic guitars are able to amplify themselves due to their construction, but even they would need sound reinforcement to fill up a stadium with sound. Amplifiers don't just boost the volume of an instrument, they can have a significant impact on the sound characteristics. That's why there are so many options and configuration possibilities.

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Recording Electric Guitar

There are two common ways to record electric guitar: placing a mic in front of a guitar amp and direct injection (DI). Each have pros and cons. Here are a few points to consider:

Mic an Amp

Pros:

Control over tone/sound

More realistic

Some special effects are possible (for example: guitar feedback)

Cons:

Requires additional gear in addition to a guitar (you need an amp)

It can get loud

Sound can bleed into other mics when recording an ensemble

Potential for more noise from surroundings in recording

DI

Pros:

Potential for very clean signal

Not loud (good for bedroom recording or late night sessions)

Isolation from other instruments recording simultaneously

No amp required (you can use amp simulator software as an alternative)

Most interfaces already include a DI feature (Hi-Z input)

Cons:

Not always ideal sounding or less realistic (some associate a "flat" sound quality)

Not possible to create natural effects (like feedback)

If you can't decide what is best for you, you can do both at the same time. Using a dedicated DI box makes it possible to record the guitar DI while also capturing the sound of the amp. DI boxes are configured so you take a single input (guitar) and send out 2 outputs (1 to the amp and 1 to an interface or mixing console). This requires twice as much space on the computer because you are recording 2 inputs with this setup, but the potential benefits are worth it. If you want to change something later, or even send the recording back into an amp at a later time, you can do that with this arrangement.

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Recording DI

There aren't as many videos dealing with recording guitar DI. Part of that is due to the fact that most guitarists prefer recording with an amp (it really is more fun). Another reason is because after recording guitar DI, you often apply signal processing - and that's something we'll explore in later modules.

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Recording DI Bass

Recording electric bass DI is far more common and actually sounds great without a lot of effort. It's also very useful and practical in live performance settings. Some player and engineers do record a bass amp along with DI, but this is when they have a particular sound in mind. DI bass sounds good most of the time and often gives a cleaner result compared to a bass amp.

These videos from Reverb.com shows a variety of recording techniques including some that aren't DI. They also demonstrate combining DI and amp signals for recording (this was mentioned earlier). They both come off a little more like a sales pitches - go figure, Reverb.com is a website for purchasing music equipment.

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If you've never played a drumset or seen one up close, it can be a little intimidating to approach recording the instrument. There are a number of things that make drumsets complicated from different sizes of drums to various kit arrangements. Unlike a guitar or other "standard" instrument, there really isn't an official standard for drums - this is also true when it comes to drum notation when writing music. However, there are some common things that are agreed upon by most players. For example, the basic drum layout (or where a certain drum component will go). Once we deviate from a "normal" or common setup, that's when we run into challenges that need problem solving. Even if you're not a drummer, these are good things to keep in mind. And even if you don't plan on recording or playing physical, acoustic drums, your music can still benefit from understanding some general setups and mechanics.

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As for mechanics, what you need to know is that drummers use all 4 limbs - 2 arms and 2 legs (this is not a requirement, and there are drummers with fewer than this average). The point is that playing the drums is incredibly physical to make sound. If you're thinking about this from a writing/composing standpoint, keep in mind that with this physical limitation you can't expect a drummer to be able to perform more than 4 hits at any one time (that would imply all limbs are hitting something) (this is a dead giveaway when you hear a drum part that has too many hits that would be physically impossible to play).

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When recording, it's important to consider the physical aspect of playing because the performer could potentially hit microphones that are placed too close. This is easy to forget when you setup when the drummer is away from the set -  the moment they start playing, be sure to observe and adjust as needed. It can be helpful to have the drummer play a little first so you can get a feel for the amount of space they will need to play naturally.

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Pianos cover a wide range of frequencies as well as physical space. Those characteristics along with numerous playing styles make the instrument potentially challenging to capture effectively (there's also the issue of different companies making different design decisions, so there's no perfect universal layout).

There are two common styles of pianos: Grand Pianos and Upright Pianos. Grand pianos are more expensive and typically found in studios and performance halls, while upright pianos are typically more affordable and common in smaller venues and in the average home.  When recording, you'll want to try a compare a few placements to see what works best.

In the following videos, a variety of strategies will be displayed (similar to what you saw for the acoustic guitar demos). Take note of the setups you find sound best. I also recommend that you add notes about what genre or style was performed along with these setups - some genres sound better with a specific setup.

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Now we are starting to enter into the world of instrument families. This means that we aren't only talking about one instrument or one general type of instrument (like guitar or piano). Within the string family, from high to low there is the violin, viola, cello, and double bass. While they all sound like strings, each has its own frequency range and resonant characteristics. In general, they do all operate the same - using a bow to vibrate strings (or sometimes other articulation methods) and a body that amplifies and resonates the sound. It may not be obvious where to place a mic, so this series of videos will give you the basic idea.

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Brass is one of the more difficult instrument families to record. It's easy for them to sound flat in sound quality or just plain fake (for those of you familiar with software sample libraries, brass and saxophones tend to be the most fake sounding out of all the available instruments recorded). These videos will give you an idea of where to start, but just like what was mentioned with strings, feel free to experiment with mic placements.

A word of caution: brass instruments are some of the loudest acoustic instruments and they primarily direct sound and air in a single direction. When placing a microphone directly in front of the sound hole of a brass instrument, make sure the mic can handle the air pressure intensity. If it can't or you're not sure, try backing away a little or move the mic off-axis, but pointed at the same spot.

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Here's another set of videos showing techniques for recording woodwinds. You'll notice that this also includes saxophone - while it looks like a brass instrument (and does have some of those qualities), it is in the woodwind family. One thing to be aware of with woodwind instruments is that sound doesn't only resonate from the bell (usually the end of the instrument), but also has a lot of resonance throughout the body of the instrument. You may need to try a number of locations to get the best sound.

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When recording more than one musical part at a time, there are a few options available: recording everything at once (live), recording each part individually, or a combination of simultaneously recorded parts and individual parts. Each approach has pros and cons. The following articles linked in this module will go into some of the se elements.

Keep in mind that these techniques will work for both ensembles/bands as well as individuals playing multiple parts.

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A few more thoughts about Overdubbing

A came across an older article that shared some tips and hints going into the overdubbing process for better results:

1. Know exactly what you are working on that day

2. Have all the resources you need available.

3. Make sure all parts have been rehearsed and that all performance issues have been sorted out.

4. The overdubbing musician should play their part at the tempo of the song when getting sounds.

5. Set up a comfortable space for the musician to work in.

6. Always have a plan B, incase everything you planned goes wrong.

7. Never rush through performances in an attempt to complete your goal for the day.

8. Never settle for average performances that will need to be fixed with editing.

9. When you capture the essence of a part, record it everywhere it needs to be without delay.

10. Take regular breaks, especially if there is frustration and confusion in the studio.

11. Always communicate with the musician immediately after a take, even if it's to tell them you are not sure or need to listen to it again.

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Once you have audio recorded in your DAW, it's time to edit. Depending on the project and performance, you may need to edit a lot of things or just a few. At minimum, you'll likely want to apply fades to the start and end points to eliminate the unnatural click or chop effect (this is what you hear when you go immediately from not sound to recorded sound and visa versa). The list below describes some common editing tools and features you'll want to be familiar with.

Trim

This is for trimming off ends of an audio recording. Sometimes you end up with a long amount of silence at the end (this is common for home producers because you may need time to reach for the stop button if you were performing). This tool works on the start/end of a file only. The image below shows trimming in Logic Pro X.

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This is the opposite, or maybe companion, to the Trim tool. Cut or splice is done anywhere other than at the start/end of a file. You may want to remove a part that was wrong or remove silences between sound. This can also be useful when you want to duplicate a part or move a section somewhere else. Cutting/Splicing can make the selection easier to manage. The image below shows a spliced audio region in Logic Pro X.

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This one should be familiar since it functions the same as other copy/paste tools outside of audio. If you ever copy/pasted text, this works the same way. The only word of caution is to make sure you align the audio properly when pasting. If done wrong, it may sound out of time or off rhythm.

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This is another straight forward tool. If you want to move audio, just grab it and move to another location. The image below shows an audio region being moved to the right in Logic Pro X.

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If you ever turned the volume knob up or down on a device, you know the sound of a fade. This is essentially an edit tool that applies automation of the volume control for a selected portion of audio. When applied to an audio region, the playback volume will follow the fade (fade in or fade out). In cases where 2 different pieces of audio overlap, you can apply a crossfade to smooth out the transition (which usually sounds rough). The first region will fade out as the second region fades in to create a seamless transition/connection. The image below shows audio region with fades in Logic Pro X (in order: Fade in, Cross fade, Fade out).

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When it comes to signal process in audio production, we can divide it into two categories: Gain-based processing and Time-based processing. Gain-based has to do with levels, or "gain," while time-based refers to effects like reverb and delay. During the mixing process, you usually utilize gain-based signal processing to help balance levels better. Time-based effects are used to provide more depth to a mix, but they tend to be applied after. This video by Waves is a brief introduction to audio effect signal process.

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What are Audio Effects? Sound Basics with Stella – Episode 1

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EQ or equalization allows you to adjust levels of specific frequencies rather than the entire volume level. This is great for reducing unwanted frequencies or boosting weak frequencies that you want to hear better. You can also use EQ for editing for"surgical EQ" needs where you focus on a very small range of frequencies for precise editing. This video by Waves looks at the use of EQ.

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EQ Explained – Sound Basics with Stella Episode 2

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As the name suggests, mixing is the process of mixing all the tracks in a production so they blend with each other. This process usually happens after everything has been recorded for a production. Sometimes, the mixing process can occur before the recording process has finished. One of the reasons why mixing can happen at different stages in the production process is because the digital production process is much more flexible than analog production. Digital production is very common for producers, so flexibility is something to be expected.

The mixing process mainly consists of balancing audio levels, choosing panning settings, and signal processing with EQ and compression. The following video has a nice introduction to the mixing process.

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This is the process of setting volume levels so that each track or instrument can be heard evenly. If you set good levels during the recording process, you may not need to do much for this element. But volume level may not be the only thing that needs adjusting to get a balanced level. For example, if a track has too many loud and soft moments or certain frequencies are more overpowering than others within the same track, signal processing with compression or EQ might be necessary. We'll look at this on the next page.

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Panning is the panoramic position of a sound. Think about what a typical band looks like on stage and where they are positioned. The singer is usually in the middle and typically the drums as well. Guitarist, keyboardists, and other instruments are typically off to a side. This  arrangement can be produced using panning. You can move a sound from left to right. The result is a more spaced out and balanced combination of sounds. Another benefit of panning is that by moving some sounds to one side and other to the opposing side, you allow a listener to hear each element with more focus. Think of these two scenarios: (1) 6 audio signals equally coming out of a set of stereo speakers; (2) 6 audio signals split across 2 stereo speakers so 3 signals are on one side and 3 on the other. This is an over-simplification, but the effect would be that the lister can distinguish each element better when they are split up. It's harder to distinguish many signals coming from the same source.

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Another element of mixing that is important but occasionally overlooked is organization. After recording, you may have a project filled with tracks and it can be a mess to look at. Organizing tracks so they are grouped by instrument type, applying color codes, and labeling each track are just a few ways to organize a project to optimize workflow. Grouping elements together as well as using color can help you to locate elements faster, and labeling helps identify tracks without the need for listening. There are many ways to approach organization, so find a way that works for you and stick with it.

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The introduction of this article from Presonus (Links to an external site.) does a nice job summarizing the concepts of dynamic processing:

Audio signals have very wide peak-to-average signal-level ratios, sometimes referred to as dynamic range, which is the difference between the loudest level and the softest level. This can create issues in both live and recording environments when gain staging a signal because when enough gain is applied to capture the softest level adequately, a peak signal can cause overload in the audio chain, resulting in signal distortion. Dynamics processing is the process of altering the dynamic range of an audio source to make it easier to place in the overall mix. Common types of dynamics processors include:

Compressor. A compressor is a type of amplifier in which gain is dependent on the signal level passing through it. You can set the maximum level a compressor allows to pass through, thereby causing automatic gain reduction above some predetermined signal level, also known as the "threshold." Punch, apparent loudness, and presence are just three of the many terms used to describe the effects of compression.

Limiters. Similar to a compressor, a limiter is an amplifier that limits the upper dynamic range of a signal to a specific threshold. Unlike a compressor, which works gradually to reduce the signal, the limiter prevents virtually any increase in gain at the upper end of the dynamic range.

Expanders. Expanders increase the dynamic range of a signal after the signal crosses a threshold. There are two basic types of expansion: dynamic and downward.

Noise Gates. A noise gate helps to reduce unwanted sounds by only allowing the signal to be heard once it has exceeded a certain amplitude.

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Mastering is the final process of music production. After you have recorded or created music, edited, and mixing everything together, the last step is to make sure everything is polished. Imagine you baked a cake and put a layer of frosting. That would taste great, I'm sure, but additional decorations would give it the final touch making it ready to present

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Analogies are always perfect, but what I like about the cake analogy is that most of us can relate to it. A basic cake is fine. It gets the job done and still tastes delicious. But when added decorations and embellishments are added, the cake experience is transcended to a new level. And if you asked an average person and a desert chef/baker to comment on a cake, they would each have their own opinions and points to make. The same is true when it comes to mastering. Mastering is the final polish and if you asked an average person and a mastering engineer to comment on a piece of music, each would have their own thoughts based on their training.

In mastering, the bulk of the music work is done in the earlier phases (recording, editing, mixing, etc.), so what's left is the small details. Mastering engineers would make huge changes to the music. Instead, they will make a series of smaller adjustments to enhance the music. This usually results in more clarity and an overall professional sound.

Mastering take lot's of experience and practice, but there are still a number of typical techniques and procedures that happen every time. The biggest elements are: EQ, Compression, and Limiting. That's roughly the same sequence of effects you would see on the main output channel (aka: master bus, stereo out, main out, or other final stage reference.) We've already talked about EQ and Compression during the mixing module, and those would be used in a similar way. Limiting is the only one we haven't formally covered.

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A Limiter is very similar to a compressor in that it is a dynamic signal processor - it listens to the incoming signal and applies volume adjustments when the threshold is exceeded. As the name suggests, a limiter limits the audio signal. While compression attenuates a signal, a limiter will not allow a signal to pass a ceiling; thus it "limits" the output of a signal.

Sounds simple enough and for the most part, it is. However, if you're not careful, you can compress the audio output to the point where it's noticeably terrible. Here's a good video with animations showing how a limiter works