Algorithmic Reverb v. Convolution Reverb

     Hello! My name is Marcos Cirino and I will be talking about the two basic reverbs in most DAWs. I am from San Juan, Puerto Rico, and am currently using what I have learned from this class to improve my band's performance and sound. Enjoy!
     There are two basic types of electronic reverbs in your DAW: convolution reverbs and algorithmic reverbs. Convolution reverbs use real sound samples, recorded from real rooms (or modified but based on real recordings in some cases) known as “Impulse Responses”.  They produce these in magical ways, and all you really need to know is that they use multiple microphones to capture the acoustics of a room and produce an IR file for you, so you don’t need to do any of that.
The reverb plug-in then filters your sound through this impulse response to generate a “believable organic tail” based on the characteristics of a real room.
     Since the effect process for these types of reverbs involves running filters over your signal and mixing that with another signal, they often have a larger impact on CPU (this is not always the case, but in general is true).  The CPU hit is similar to another audio track in your project.  So two tracks, each with a convolution reverb as an insert effect will make your DAW behave like it has roughly four tracks playing simultaneously. One common problem with reverb in general is the amount of bass build up they can cause.  Physics lesson: Lower frequencies tend to penetrate surfaces, and higher frequencies tend to bounce of surfaces (the effect we’re trying to produce).  Reverbs tend to just apply their tail to everything. Convolution reverbs, using their impulse response files, tend to exaggerate… you guessed it, the impulse response from your sound.  This is the initial collision between a sound and the surrounding material.  If this is not set up correctly, it can cause your bass to build up much faster than algorithmic reverbs. The good part about convolution reverbs is that they sound very realistic.
     We also have algorithmic reverbs. This is your standard reverb plug-in. They get the job done, but don’t sound amazing.  This is because a basic reverb is pretty easy to create, but very hard to master.
Algorithmic reverbs generate your reverb sound strictly based on parameters you set in the DAW, they try to generate the same thing as convolution reverbs but because they are simulating the impulse responses (as opposed to the convolution case) they can tend to sound fake especially when isolated in a solo instrument case.  Unless you have a very good reverb plug-in, I recommend not using algorithmic reverb on your solo instruments. The good thing about algorithmic reverbs is that they have less impact on your CPU.
Sources:

http://brian-doyle.com/2011/10/28/convolution-vs-algorithmic-reverbs/

Dynamic Processors

Hello! My name is Marcos Cirino, and I am going to be explaining the basic dynamic processors. I am from Puerto Rico and I am part of a band called 23rd street. I hope this post is helpful, and I look forward to your feedback!

A dynamic processor is something that outputs a signal, where the level of the outgoing signal is based on the level of the incoming signal. In other words, a loud signal coming in will come out differently than a quiet signal coming in. There are four basic types of dynamic processors: compressors, gates, expanders, and limiters. The most common of these three is the compressor.

First of all, we have compressors. What happens when a compressor is set? The louder a signal is coming in, the less level it provides going out. In a compressor, a target level is set — called the “threshold” — and any signal coming in that exceeds that level will be reduced. The higher the level above that threshold, the more reduction will occur.

Next, we have limiters. Limiters are like super compressors. The idea is to ensure that the level does not exceed the threshold. Because this amount of compression is extreme, a limiter relies on certain functions and design that regular compressors do not have.

Next, we have expanders. The quieter the signal is coming in, the less level an expander. provides going out. In other words — it makes quiet signals even quieter. Much like a compressor, the threshold is set at a certain level. Any signal that does NOT exceed that threshold is reduced, and the quieter the signal, the more reduction is done.

Lastly, we have gates. Gates are like super expanders. Anything that does not exceed the threshold is reduced to inaudible. Again, because gates are extreme, they often require a slightly different design than a regular expander.

Now, we have talked about a "threshold". The threshold control adjusts the level where the dynamic processor will start to work. In a compressor or limiter when the loud passages at the input exceed the threshold level set, the unit will turn down these loud passages. In an expander or gate, the unit will turn down any incoming signals that are below the threshold level.   The threshold is usually adjustable by a control marked "threshold."  Reducing the threshold level means that more peaks of the signal will trigger the compressor to turn down the gain; increasing the threshold level means that an expander or gate will turn down more low-level signals.  The attack time is how fast the dynamics processor will react to a signal crossing the threshold level, going up.  In a compressor it is the time it takes the compressor to reduce gain on a high-level passage.  On an expander, it is the time that the expander takes to restore full gain after the audio level comes up after a low level passage. The release time is how fast the dynamics processor will react to a signal crossing the threshold level, going down.  In a compressor, it is the time it takes the unit to restore gain after the high-level passage is over with.  In an expander, it is the time the expander takes to turn down a low level passage (below the threshold level). 

There also exists the attack and release times.

In conclusion, I would like to explain the ratio. The ratio control determines how much the signals that are being compressed or expanded will be turned down. If a compressor has a 2:1 ratio, the compressor will turn down the gain so that if the input signal is 2dB above the threshold level, the output increases only one dB. If the input signal is 4 dB above the threshold, the gain will be turned down so that the output only rises 2 dB above the threshold (a 2 to 1 ratio).   At a 4:1 ratio the input signal has to be 4 dB above the threshold for the output to increase 1 dB. When the ratio control is set to 10:1 or more, the compressor is called a limiter because the unit is effectively preventing the peak levels from increasing any significant degree above the threshold level.

References:

http://www.recordinginstitute.com/da154/ARP/chap3Sig/asp2.html
http://theproaudiofiles.com/dynamics-processing/

Submix Concept

Hello! My name is Marcos Cirino and I am taking this class in order to further my understanding of music. I have an alternate rock band called 23rd street and we are current,y recording original songs, so I am also taking the course to help in the production process. I will be talking here about the submix concept.

One function of a Bus is to provide a way in which you can combine multiple audio tracks and send to a single location. This is known as creating a submix.

Let's say for example, that you have recorded drums using multiple microphones (Kick, Snare, Hi Tom, Floor Tom, Overheads , etc.) and you’d like to combine them all into one stereo track. 

The idea of using submixes, in general, is to group things that belong together. So, you can submix background vocals, doubled acoustic guitar parts, or in an orchestral setting you might want to create a different submix for brass, woodwinds, strings, percussion, etc.

The benefits of using the Bus system in your DAW for submixing as described above are many. It gives you the ability to simplify extensive track counts into fewer faders for a more manageable and easier mixing process. 

Naming each bus (or submix) is encouraged in order to avoid issues caused by unintentionally routing different tracks to the same Bus.

The basic steps to creating a submix are as follows:

1. Create an Aux Input Channel and name it.

2. Choose the input to the Aux input.

3. Make the outputs of the channels match the input of the Aux input channel. 

Sources:

http://mixcoach.com/mixing-tools-bus-part-1-submix/

The Analog to Digital conversion process

Hello! My name is Marcos Cirino, and I am from Puerto Rico. I currently play in an alternate rock-band called 23rd Street. I am taking this class to further my knowledge of the music creation process. Here I will talk about the analog to digital conversion process.

The device responsible for changing an analog signal into a series of numbers is the analog-to-digital converter (or A/D converter). It samples the strength of the changing voltage at regular intervals, generating a steady stream of numbers. Two parameters directly affect the quality of the resulting audio: sample rate and bit depth.

The converter's sample rate dictates how often it measures the signal to generate a new value. The more frequently the converter measures the signal, the more accurate the resulting data. Sample rate corresponds directly to frequency response; the highest frequency a digital system will capture is exactly one-half the sample rate. To capture the full audio spectrum up to around 20,000 cycles (or 20kHz), a sample rate of 44.1kHz is common. Higher sample rates make for increased treble response and a more "hi-fi" sound. Low sample rates sound duller and darker.

Bit depth affects how many bits the converter uses for each numerical measurement of the signal. More bits equal a more accurate measurement, which explains why 16-bit CD audio sounds so much better than an 8-bit multimedia sound file. A low bit depth is like forcing the converter to measure the sound with a yardstick marked only in inches. A higher bit depth allows the converter much greater accuracy (a yardstick marked in 1/8th-inch increments, for example).

Thank you for reading, and I look forward to your feedback!

Sources: 

http://www.videomaker.com/article/3258-audio-advice-analog-to-digital-conversion

Microphone Polar Patterns

When setting up a home studio, it is important to choose the right type of microphone or microphones. You see, each microphone has its own polar pattern. A polar pattern is a graphic representation of a microphone's sensitivity to sound relative to the direction or angle from which the sound arrives. There are four basic types of polar patterns for microphones: Cardiod, Super Cardiod, omnidirectional and bidirectional; each serves a different purpose.

A Cardiod microphone is a microphone that is most sensitive to sound from the front and least sensitive to sound from the back. Their unidirectional pickup makes for affective isolation of unwanted ambient sound and high resistance to feedback when compared to omnidirectional alternatives. A cardiod microphone are probably the best option for a live performance.

Supercardioid microphones offer a narrower pickup than cardioids and a greater rejection of ambient sound. However, they also pick up a small amount of sound from directly behind. For this reason, it is particularly important to place monitor speakers to the side facing the 'dead spots'. Supercardioids are highly suited to very loud stage environments as they are very directional with high gain before feedback. 

Omnidirectional microphones are equally sensitive to sound arriving from all angles. Therefore, the microphone does not need to be aimed in any particular direction. This can be particularly useful when using a lapel mic to capture a speaker’s voice, as the individual can move their head without affecting the sound. The disadvantage is that an omni mic cannot be aimed away from undesired sources, such as PA speakers, which may cause feedback.

A bidirectional microphone (also called a figure-8 microphone) picks up sound from the front and rear of the microphone, but not from the sides. Microphones with this patterns are usually ribbon or large diaphragm condenser microphones. These microphones are very sensitive and therefore should be kept away from monitors.

Sources:
http://www.shure.co.uk/support_download/educational_content/microphones-basics/microphone_polar_patterns