The Audio Stack

DAWs have multiple views/windows:

Main Views:

Arrange/Timeline — Linear view where tracks run horizontally over time (Logic: Tracks, Ableton: Arrangement, FL: Playlist)

Session/Clips — Grid-based clip launching (Ableton: Session, Logic: Live Loops, FL: Performance Mode)

Mixer — Channel strips with faders, pan, sends (like a hardware console)

Editor Windows:

Piano Roll — Grid for editing MIDI notes (pitch vs time)

Audio Editor — Waveform editing, warping, slicing

Score Editor — Traditional notation view

Automation — Parameter changes over time

Browser/Library:

Loop Browser — Apple Loops, Splice, built-in content

Sound Library — Instrument presets, samples

Plugin Browser — Available instruments and effects

File Browser — Your files and project assets

Other Windows:

Instrument/Plugin — Opens synth or effect interface

Smart Controls — Simplified parameter view

Step Sequencer — Pattern-based programming

List Editors — Event lists, markers, tempo

Audio is actual recorded sound — waveforms captured from the real world. MIDI is performance data that controls instruments:

MIDI Message Types:

Note On/Off — Which notes play and when

Velocity — How hard you press (affects volume/timbre)

Aftertouch (Channel) — Pressure after initial press (adds vibrato, filter sweeps)

Polyphonic Aftertouch — Per-note pressure (MPE controllers, digital wind instruments)

Pitch Bend — Smooth pitch slides (like bending guitar strings)

Mod Wheel (CC1) — Expression control (often controls vibrato)

Sustain Pedal (CC64) — Hold notes (like piano pedal)

Expression (CC11) — Volume/intensity control

Other CCs — Hundreds of assignable controllers

MIDI makes no sound until it triggers a synth, sampler, or external instrument. The same MIDI can play any sound.

Plugins extend DAW capabilities. Stock plugins are excellent for learning and often professional-quality. Third-party plugins exist for specific reasons:

Why Third-Party Plugins:

Unique Character — Different saturation, compression character (FabFilter vs stock sounds different)

Signature Sounds — Brand-specific sounds (Waves SSL, UAD Neve = modeled hardware)

Specialized Features — Specific tools stock doesn't have (Melodyne, Soundtoys)

Better Visualizations — More intuitive interfaces (FabFilter Pro-Q, iZotope)

Workflow Preferences — Some prefer certain layouts or controls

Stock Plugin Advantages:

No extra cost

Optimized for DAW (lower CPU, tighter integration)

Often underrated quality

No compatibility issues

Plugin Formats:

VST3/VST (Windows & Mac) — Most universal

AU (Mac only) — Apple's format for Logic, GarageBand

AAX (Pro Tools only) — Avid's proprietary format

Format must match your DAW and OS.

Different sources produce different signal levels and have different electrical properties. Using the wrong input causes noise, distortion, or weak signal:

XLR (3-Pin Circular):

For: Microphones (dynamic, condenser, ribbon)

Signal Level: Mic level (~-60dBu) — very quiet

Why: Balanced connection (3 wires: hot, cold, ground) rejects interference over long cables

Needs: Preamp gain to boost to usable level

48V Phantom Power: Required for condenser mics (powered from interface)

1/4" TRS (Tip-Ring-Sleeve):

For: Line level devices (synths, mixers, outboard gear)

Signal Level: Line level (-10dBV consumer, +4dBu pro)

Why: Already amplified, just needs level matching

Also: Can be balanced (TRS) or unbalanced (TS)

1/4" TS with Hi-Z:

For: Electric guitars, bass guitars (passive pickups)

Signal Level: Instrument level (between mic and line)

Why Hi-Z: Guitar pickups are high impedance — low impedance input loads them down, causing dull tone

Switch: Enable 'Hi-Z' or 'Instrument' on interface for proper impedance matching

Combo Jacks: Modern interfaces use combo jacks that accept both XLR (insert center) and 1/4" (insert as normal). Convenient but check which signal type is selected.

Sample Rate = how many measurements (samples) per second. Like frames in video — more frames = smoother motion. More samples = higher frequencies captured. Nyquist theorem: need 2x the highest frequency (20kHz hearing = 40kHz minimum, hence 44.1kHz). Bit Depth = precision of each measurement. 16-bit = 65,536 possible values. 24-bit = 16.7 million values. More precision = lower noise floor, more headroom. Latency = delay between input and output. Caused by buffer (computer needs chunks of data to process). Smaller buffer = less latency but more CPU strain. Under 10ms feels instant. I/O Count = simultaneous inputs/outputs. 2x2 = 2 in, 2 out (stereo). Recording drums? Need 8+ inputs.

Software Monitoring: MIC -> ADC -> DAW (processing) -> DAC -> Headphones. The round-trip through your computer adds latency (delay) based on buffer size. At 256 samples/48kHz, that's about 10-15ms — noticeable and distracting when singing or playing. Direct Monitoring: The interface routes your input signal directly to the headphone output BEFORE it goes to the computer. You hear yourself instantly (zero latency) while the signal simultaneously goes to the DAW for recording. The trade-off: you can't hear DAW effects (reverb, compression) on your voice while recording — you hear the dry signal. When to Use Each:

Direct Monitoring ON: Recording vocals, acoustic instruments, anything where hearing yourself in real-time matters

Direct Monitoring OFF: Editing, mixing, or when you need to hear plugin effects while recording (and can tolerate some latency)

Nearfield monitors minimize room problems by sitting close to you. Position them to form an equilateral triangle with your head, tweeters at ear height. Angle them inward so they cross just behind your head.

Open-back headphones sound more natural and spacious — better for critical mixing. Closed-back isolate you from outside sound and prevent bleed into mics — essential for tracking vocals/instruments.

Your room affects what you hear more than your monitors do. Untreated rooms lie to you — bass builds up in corners, reflections confuse stereo imaging. Treat reflection points, add bass traps, or rely heavily on headphones.

Why 'Flat' is a Myth: No monitor measures perfectly flat. No room is acoustically perfect. Your ears have their own frequency response that changes with fatigue. The goal isn't perfection — it's consistency and calibration. Practical Strategies:

Reference Tracks: Import 2-3 professional mixes you know intimately into your session. A/B constantly. If your mix lacks bass compared to the reference, add bass — even if it sounds 'right' on your monitors.

Multiple Playback Systems: Check your mix on headphones, earbuds, car stereo, phone speaker, laptop. Where do all systems agree? That's closer to truth.

Know Your Room: Spend time listening to music you know well. Notice: 'My room has a 100Hz bump' or 'Highs sound duller here.' This mental calibration is invaluable.

Room Correction Software: Tools like Sonarworks SoundID, IK Multimedia ARC, or Genelec GLM measure your room and apply corrective EQ. Not perfect, but helps.

Spectrum Analyzers: Use visual feedback to cross-check what you hear. If you think there's enough bass but the analyzer shows none, investigate.

Fresh Ears: Take breaks. Ear fatigue makes everything sound duller over time. Come back the next day before finalizing.

DAW USB/Thunderbolt DAC (converts to analog) output volume control cables monitors/headphones. The volume knob on your interface controls the DAC output level — use it as your main volume.

Most problems have simple causes. No sound? Check the input routing. Crackling? Buffer size or CPU. Latency? Buffer size or use direct monitoring. Systematic troubleshooting along the signal chain finds the issue.