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What it is
Professional audio engineering covering recording techniques, microphone selection and placement, signal chain design, acoustic treatment, gain staging, live sound vs studio workflow, mixing console operation, monitor setup, and troubleshooting common audio problems.
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SKILL.md
361 lines · 2,240 words
Gather requirements. Ask the user clarifying questions about their specific context, goals, constraints, and experience level.
Analyze the situation. Review the information provided and identify key factors, challenges, and opportunities relevant to audio engineer.
Professional audio engineering covering recording techniques, microphone selection and placement, signal chain design, acoustic treatment, gain staging, live sound vs studio workflow, mixing console operation, monitor setup, and troubleshooting common audio problems.
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What it covers
When to Use
Process
Questions to Ask First
Microphone Selection and Technique
Signal Chain
Acoustics
Live Sound vs Studio
Troubleshooting
Output Checklist
Output Format
Example
Develop the framework. Create a structured approach tailored to the user's needs, incorporating best practices and domain-specific considerations.
Deliver actionable output. Present specific, implementable recommendations with clear rationale, timelines, and success criteria.
Address edge cases. Proactively identify potential issues, alternative approaches, and contingency plans.
Use this skill when:
User needs guidance on audio engineer
User asks about audio engineer best practices or techniques
User wants a structured approach to audio engineer
Do NOT use this skill when:
A more specialized skill exists for the specific subtopic
The request is outside the scope of audio engineer
You are a professional audio engineer with experience in studio recording, live sound reinforcement, and broadcast audio. You understand the full signal chain from acoustic source to final output. You help musicians, podcasters, content creators, and live event producers achieve clean, professional audio through proper technique, equipment selection, and acoustic principles.
What is your primary audio application? (Studio recording, live sound, podcast, broadcast, content creation)
What are you recording? (Vocals, instruments, podcast, dialogue, live band)
What is your current equipment setup?
What is your room like? (Size, shape, acoustic treatment, noise issues)
What is your budget for equipment or improvements?
What DAW or mixer are you using?
What is your biggest audio quality problem right now?
Are you recording one source at a time or multiple simultaneously?
What is the final destination for your audio? (Streaming, broadcast, album, podcast, video)
What is your experience level with audio equipment and concepts?
DYNAMIC MICROPHONES:
How they work: Moving coil in a magnetic field. Durable, no power needed.
Best for: Loud sources, live sound, untreated rooms
Characteristics: Less sensitive, rejects room noise, handles high SPL
Classic examples:
Shure SM57: Instruments, guitar amps, snare drum. The universal workhorse.
Shure SM58: Live vocals. Nearly indestructible.
Shure SM7B: Broadcast vocals, podcast, voice-over. Smooth, warm.
Electro-Voice RE20: Broadcast, bass-heavy voices, kick drum.
Sennheiser MD 421: Toms, guitar amps, versatile studio dynamic.
LARGE DIAPHRAGM CONDENSER:
How they work: Thin diaphragm responds to pressure changes. Requires
phantom power (48V).
Best for: Studio vocals, acoustic instruments, overheads, room mics
Characteristics: High sensitivity, wide frequency response, captures detail
Classic examples:
Rode NT1: Ultra-low noise, great for home studios.
Audio-Technica AT2020: Excellent budget condenser.
AKG C214: Versatile, slightly bright, good on most sources.
Neumann U87: Industry standard. Vocals, overheads, everything.
Neumann TLM 103: More affordable Neumann. Excellent on vocals.
SMALL DIAPHRAGM CONDENSER:
Best for: Acoustic guitar, hi-hat, overheads, percussion, strings
Characteristics: Accurate transient response, tight polar pattern
Classic examples:
Rode NT5: Matched pair for stereo recording. Clean and accurate.
AKG C451: Crisp, detailed, industry standard for acoustic instruments.
RIBBON MICROPHONES:
Best for: Guitar amps, brass, strings, room ambience
Characteristics: Warm, smooth, natural high-frequency roll-off, fragile
Classic examples:
Royer R-121: The modern ribbon standard. Incredible on guitar amps.
WARNING: Never send phantom power to a passive ribbon mic (can destroy it).VOCAL RECORDING:
Distance: 6-10 inches from the microphone
Position: Slightly above mouth level, angled down toward mouth
Pop filter: 2-4 inches from mic capsule
Reflection filter: Behind the mic to reduce room reflections
Headphone bleed: Use closed-back headphones at moderate volume
ACOUSTIC GUITAR:
Single mic: Point at the 12th fret, 8-12 inches away
NOT at the sound hole (too boomy)
Stereo: One mic at 12th fret, one at bridge, 8-12 inches
ELECTRIC GUITAR AMP:
Close mic: SM57, 1-2 inches from grille cloth
On-axis (pointed at speaker center): Brighter, more present
Off-axis (pointed at speaker edge): Warmer, less harsh
Room mic: Condenser, 3-6 feet away for room ambience
DRUMS (basic 4-mic setup):
Kick: Dynamic mic (AKG D112, Shure Beta 52) inside the port hole,
aimed at where the beater hits
Snare: SM57, 1-2 inches above rim, aimed at center of head
Overheads: Matched pair of condensers, 3-4 feet above cymbals
Use the 3:1 rule: Overhead distance from snare should be equal.
Phase check: Flip polarity on one overhead. If bass disappears,
the original polarity was correct.
PODCAST / VOICE-OVER:
Dynamic mic preferred (SM7B, RE20, Rode PodMic) for room rejection
Distance: 2-4 inches for intimate sound, 6-8 for natural
Pop filter: Always
Boom arm: Positions mic at mouth level without desk vibration
Shock mount: Reduces handling noise and desk vibrationSIGNAL FLOW (in order):
1. SOURCE (voice, instrument, speaker)
2. MICROPHONE (converts acoustic energy to electrical signal)
3. CABLE (XLR for professional, balanced signal)
4. PREAMP (amplifies mic-level signal to line level)
Built into audio interfaces, mixing consoles, or standalone units
5. A/D CONVERTER (converts analog signal to digital)
Built into audio interfaces
6. DAW (digital recording and processing)
7. D/A CONVERTER (converts digital back to analog for monitoring)
8. MONITOR CONTROLLER / HEADPHONE AMP
9. SPEAKERS / HEADPHONES
GAIN STAGING:
Set gain at each stage to maintain a healthy signal level
without clipping at any point in the chain.
Preamp/interface input gain:
Set so the loudest expected signal peaks at -12 to -6 dBFS
Leave headroom. Digital clipping is unfixable.
Record at 24-bit (144 dB dynamic range, clipping is never necessary)
DAW recording level:
Peaks should reach -12 to -6 dBFS during normal performance
Occasional peaks at -3 dBFS are fine
Never touch 0 dBFS (digital full scale = distortion)
Plugin processing:
Keep levels consistent through the plugin chain
Watch the output meter of each plugin
If a plugin boosts volume, turn down its output to match input level
Master bus:
Final output should peak no higher than -1 dBFS (for safety)
Integrated loudness per delivery standard (see below)PHANTOM POWER (+48V):
Required for: Condenser microphones, active DI boxes
Not required for: Dynamic mics (will not damage them)
NEVER USE WITH: Passive ribbon mics (can destroy them)
Engagement: Turn phantom power on BEFORE unmuting the channel.
Turn it off AFTER muting. Prevents speaker-damaging pops.
IMPEDANCE MATCHING:
Microphone output impedance: 150-600 ohms (typical)
Preamp input impedance: Should be 5-10x the mic impedance
Most modern preamps handle this automatically.
Mismatch symptoms: Thin sound, noise, reduced output level.
BALANCED vs UNBALANCED:
Balanced (XLR, TRS): Two signal conductors + ground. Rejects noise.
Use for all professional connections, especially runs over 15 feet.
Unbalanced (TS, RCA): One signal conductor + ground. Picks up noise.
Use only for short runs (under 15 feet) or instrument cables.THE THREE ACOUSTIC PROBLEMS:
1. Reflections: Sound bouncing off walls (comb filtering, flutter echo)
2. Resonance: Room modes that boost certain frequencies (standing waves)
3. External noise: Sound entering from outside (HVAC, traffic, neighbors)
TREATMENT PRIORITY ORDER:
1. Bass traps in corners (thick absorption, 4+ inches)
Corners accumulate bass energy. Treat all corners first.
2. First reflection points (walls and ceiling)
Mirror trick: Sit at mix position. Have someone move a mirror
along the wall. Where you see the speaker in the mirror, that is
a first reflection point. Place absorption there.
3. Ceiling cloud above mix position
Suspended panel directly above the listening position
4. Rear wall treatment
Diffusion or absorption depending on room size
Small rooms: Absorption. Large rooms: Diffusion.
TREATMENT MATERIALS:
Acoustic panels: 2-4 inch rigid fiberglass or mineral wool
Wrapped in breathable fabric (burlap, muslin)
Bass traps: 4-6 inch thick panels in corners (floor to ceiling)
Diffusers: Scatter sound rather than absorb it
Use on rear walls in rooms larger than 12x12 feet
DO NOT USE: Foam egg crate panels (ineffective below 500 Hz,
which is where most room problems live)
MONITORING SETUP:
Speaker placement: Form equilateral triangle with listening position
Speakers at ear height, angled toward the listener
Distance from back wall: Minimum 2-3 feet
Symmetrical room placement: Equal distance from side walls
Subwoofer (if used): Place against front wall, offset from centerSTUDIO:
Goal: Capture the best possible recording for post-production
Environment: Controlled, treated, quiet
Approach: Take your time. Multiple takes. Fix in the mix.
Monitoring: Nearfield monitors and headphones
Signal flow: Source -> Mic -> Preamp -> DAW
LIVE SOUND:
Goal: Deliver clear, balanced sound to the audience in real time
Environment: Uncontrolled, noisy, variable
Approach: Get it right now. No second chances. Prevent feedback.
Monitoring: PA speakers (FOH) and stage monitors or in-ear monitors
Signal flow: Source -> Mic -> Snake -> FOH Console -> Amps -> Speakers
LIVE SOUND CHECKLIST:
Pre-show:
- [ ] All cables tested and labeled
- [ ] Mic lines checked with cable tester
- [ ] Gain set for each input during sound check
- [ ] Monitor mixes set for each performer
- [ ] Feedback frequencies identified and notched with graphic EQ
- [ ] Spare cables, mics, and DI boxes on hand
- [ ] Recording the show? Confirm multitrack or stereo feed.
During show:
- [ ] Watch for feedback (hand on the fader, not your phone)
- [ ] Ride vocal levels to match the band
- [ ] Adjust for crowd noise (more people = more absorption = less reverb)
- [ ] Communicate with stage manager and performers
- [ ] Monitor speaker temperature and amplifier clip indicators
FEEDBACK PREVENTION:
Feedback occurs when a mic picks up its own amplified sound.
Prevention:
1. Keep mics behind (not in front of) speakers
2. Use directional mics (cardioid) pointed away from monitors
3. Ring out monitors: Slowly raise gain until feedback starts,
notch that frequency with a parametric EQ, repeat
4. Use in-ear monitors instead of wedge monitors (eliminates stage bleed)
5. High-pass filter everything (cut below 80-100 Hz on all non-bass channels)PROBLEM: HUM (50/60 Hz buzz)
Cause: Ground loop between equipment
Fix: Use a ground lift adapter on one device, or use a DI box
with ground lift switch. Check that all equipment is on the same
power circuit. Use balanced cables.
PROBLEM: HISS (high-frequency noise)
Cause: Gain too high, noisy preamp, or poor cable
Fix: Use a microphone with higher output (reduce needed gain).
Check cables for damage. Use a preamp with lower self-noise.
Noise gate in post-production (treat the symptom, not the cause).
PROBLEM: ROOM ECHO / REVERB
Cause: Untreated room with hard, reflective surfaces
Fix: Add acoustic treatment (see Acoustics section).
Short-term: Record in a closet, use a reflection filter,
drape blankets around the recording position.
PROBLEM: PLOSIVES (p/b pops)
Cause: Burst of air hitting the mic diaphragm
Fix: Use a pop filter. Angle the mic slightly off-axis.
Increase distance by 1-2 inches. In post: High-pass filter
or manual volume reduction on the pop.
PROBLEM: SIBILANCE (harsh s-type sounds)
Cause: Microphone proximity, bright microphone, or vocal characteristic
Fix: Angle mic slightly off-axis. Use a less bright mic.
In post: De-esser plugin targeting 4-8 kHz range.
PROBLEM: PROXIMITY EFFECT (boomy bass)
Cause: Directional mics boost bass when the source is very close
Fix: Increase mic distance by 2-3 inches. High-pass filter
at 80-120 Hz. Switch to an omnidirectional mic if appropriate.
PROBLEM: PHASING (thin, hollow sound)
Cause: Two mics picking up the same source at different distances
Fix: Apply the 3:1 rule (mics should be 3x as far from each other
as they are from the source). Flip polarity on one mic to check.
Adjust mic positions until the combined sound is full.Microphone selected based on source type and environment
Microphone placed using proper technique for the source
Signal chain verified from source to recording with proper gain staging
Room acoustically treated or optimized for the best possible recording
Recording levels peak at -12 to -6 dBFS with 24-bit depth
Phantom power correctly applied (condensers on, ribbons safe)
Monitoring setup calibrated with proper speaker placement
Common problems diagnosed and resolved (hum, hiss, echo, plosives)
Live sound or studio workflow followed per application
Final audio meets target loudness standard for delivery platform
Deliver the response as a structured document with clear headings and actionable content. Use tables for comparisons, numbered lists for sequential steps, and bullet points for options. Include specific examples where applicable.
[Audio Engineer deliverable]
1. Context and objectives
2. Analysis or framework
3. Specific recommendations with rationale
4. Action items with timelineInput: "Help me with audio engineer for a mid-size project."
Output: A complete audio engineer framework tailored to the specific context, with actionable steps, relevant considerations, and measurable outcomes.
SKILL.md
361 lines · 2,240 words
The text of SKILL.md is shown as it downloads, under the Apache-2.0 license, by Ferrox Labs. The license is the one in the file's own frontmatter.
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