Scoring Central

Software:

• A digital audio workstation (DAW) is the main computer program used to record and edit audio and MIDI tracks and mix them into finished music or other audio. Some of the most common DAWs are Logic Pro X, Pro Tools, Cubase and FL Studio.
  
• Plugins are software components loaded into the DAW to perform different tasks. Common plugin formats are VST, AU (Mac) and RTAS or AAX (Pro Tools).
  • Instrument plugins (or virtual instruments) are plugins that generate sound from MIDI data.
    • Samplers generate sound by playing back samples of individually recorded notes and phrases of a particular instrument. A collection of such recordings is called a sample library.
      
    • Synthesizers generate sound with mathematical models of different waveforms. The timbre is often highly adjustable by the user.
      
  • Effect plugins are plugins that modify sound. Examples of effect plugins are compressors, equalizers and reverbs.
    

Audio settings:

• Buffer size determines how many samples the DAW processes at a time before sending it to the speakers.
  

Recording & mixing:

• Audio tracks are tracks where audio signals can be recorded digitally.
  
• MIDI tracks (short for Musical Instrument Digital Interface) are tracks where individual note positions, pitches, lengths, velocities, and various other properties can be recorded numerically. MIDI tracks can be recorded by playing a MIDI keyboard or by entering notes manually with a computer keyboard and mouse. MIDI tracks get routed to instrument plugins where the numerical data gets turned into an audio signal.
  
• Volume is typically adjusted in decibel (dB) values. +6 dB is perceived as roughly twice as loud, and -6 dB roughly half as loud.
  
• Panning is the left-to-right positioning of each respective track in the mix.
  • Balancing, when mixing a stereo track, is adjusting the left-to-right positioning of the track by adjusting the volume levels of the left and right channels in relation to each other. The pan pots in most DAWs function like this.
    
  • Stereo panning brings some of the left channel over to the right side when panning right, and vice versa when panning left. This enhances the stereo separation of the mix, but also affects the stereo width (and sometimes the timbre) of the signal. Some DAWs, like Ardour, have this built into them, but it is generally accomplished with effect plugins.
    
• Sends are means of sending a copy of an audio signal to an auxiliary channel (often called bus or aux). They are often used for CPU-heavy effects like reverb. Sending multiple channels to a single aux allows you to use the same effects for all of the channels.
  

Effect plugins:

• Wet and dry controls allow you to blend the modified (wet) signal with the unmodified (dry) signal. They can be found in many plugins.
  
• An equalizer is used to adjust the balance of different frequencies in the signal.
  
• A compressor is an effect that dips the volume when the signal reaches a certain threshold.
  
• A delay is exactly what it sounds like. It delays the signal by a specified time. Wet and dry controls can be used to create an echo effect.
  
• A reverb simulates reverberation.
  • An algorithmic reverb approximates reverberation using an algorithm that is often highly adjustable by the user.
    
  • A convolution reverb simulates reverberation by applying a recorded snapshot of a physical space (or reverb device) to the signal. This snapshot is called an impulse response.
    
• A saturation plugin is an effect which creates a special type of distortion that can benefit the sound. The effects are often based on properties of hardware such as tube amplifiers and mixing consoles.
  

Sample library terms:

• Velocity layers are samples of each note recorded at different velocities. As the timbre of a real instrument is often affected by the dynamics, a library with multiple velocity layers will often result in a more natural sounding performance.
  
• Round robin (RR) samples are multiple recordings of the same note at the same velocity. If the same note is played multiple times in succession, the sampler alternates between the round robin samples as simply repeating the same sample sometimes ends up sounding fake.
  
• Legato samples or true legato are samples of different two-note phrases. A real instrumentalist anticipates and expresses note transitions in a certain way, and legato samples are a way of simulating this.
  

NOTE: This glossary is by no means complete. I’m a bit busy this summer, so I’m going to revise and build upon it whenever I have a spare moment. Naturally, I’m open to suggestions. Most importantly, don’t hold yourself back if you think I got something wrong.

Maybe you could make a link to a Wikipedia article (for example delay, reverb, etc...) Let me know if you need help.

I'm glad if you can help. This was a bit unwieldy to edit with the forum software, so I have the entire thing as a Google doc. I'm sending you the link in a PM.

For some reason, I never stuck this thread. Now it's stuck.

Sample Rate: The number of samples, or slices, of audio in a signal per second. Commonly 44.1 kHz or 48 kHz (kHz = thousand cycles per second). Should be twice of the highest desired reproducible frequency; as human hearing only goes up to about 18 kHz for most people and 20 kHz for the truly gifted, then 44.1 kHz is plenty for most cases. This is the value most commonly related to sound quality, although really it only deals with the representable frequency range.
Bit Depth: The number of bits in each sample. This is related to how large the dynamics can be. Typically we use 16-bit audio, which can express a total theoretical dynamic range of 96 dB, the volume difference from gentle whispering to a Jet taking off a few hundred meters away and is totally sufficient for all consumer uses- in particular high-end or recording cases, 24-bit may be used instead. This has absolutely zero effect on sound quality, only on the noise floor (which is typically below human hearing anyway).
Bitrate: The number of bits per second in a lossy compression format such as mp3 or ogg Vorbis. The smaller the value, the more compression and greater loss of fidelity will be experienced. In lossless STEREO audio, this is 1,411 kbps, and half of that for mono audio (because stereo is two channels).

I find these three terms confuse people the most.
For more reference, there is always Monty's commentary on the matter-
http://xiph.org/~xiphmont/demo/neil-young.html

Edit, also, I'm not entirely sure, but I do not think Stereo panning is correct- I believe most Stereo Positioning plugins function to generate stereo panning by instead adjusting the delay of one channel slightly so that the channel which arrives earlier appears to be closer (as in real world acoustics).

The dual-panpot system you see in DAWs like Pro Tools is exactly what you're describing- bringing one channel into the other. I think this is to some extent how stereo width functions, but again, I don't know a whole lot about the inner mechanics of stereo signal manipulation, so I'd have to do some research.

Footnote to Samulis' post: an easy way of understanding and correlating sampling frequency and bit depth for beginners is thinking of them as X and Y axes in a graph. X being sampling frequency, as it is time-based, and Y being the dynamic resolution of each snapshot over a period of time.

This could even be made into a neat little graphical illustration of the concept I think.

(10-09-2016, 04:50 PM)Samulis Wrote: [ -> ]Edit, also, I'm not entirely sure, but I do not think Stereo panning is correct- I believe most Stereo Positioning plugins function to generate stereo panning by instead adjusting the delay of one channel slightly so that the channel which arrives earlier appears to be closer (as in real world acoustics).

The dual-panpot system you see in DAWs like Pro Tools is exactly what you're describing- bringing one channel into the other. I think this is to some extent how stereo width functions, but again, I don't know a whole lot about the inner mechanics of stereo signal manipulation, so I'd have to do some research.

Sorry for missing this. Anyway, what you describe is using a stereo delay to create a precedence effect. This is analogous to pivoting musicians around an AB stereo microphone setup.

Now, I don't think people refer to the precedence effect specifically when they say "(true) stereo panning". I could be wrong, though. My conclusion was mostly based on usage on audio forums, and the fact that the term "stereo panning" is semantically closer to dual pan pots. The exact terminology doesn't seem to be set in stone from a mixing engineer's perspective, anyway.

Then there's the matter of relevance as well. Personally, I think stereo delay panning is slightly beyond the scope of audio basics. Also, it's not exactly the most wieldy type of panning for creating mixes that need to translate well across different listening environments.