Use Cases

Real-world applications and workflows where TabDSP enhances audio quality or enables professional monitoring in browser-based environments.

Music Listening Enhancement

Transform streaming audio into a more engaging listening experience.

Why Use TabDSP

  • Compensate for compressed streaming codecs (MP3, AAC) by adding high-end air and warmth
  • Tame harsh or fatiguing mixes (over-compressed modern mastering)
  • Boost bass on laptop/small speakers without distortion (multiband compression prevents muddiness)
  • Personalize frequency response to match your hearing or preference

Recommended Settings

Preset: Start with "Music Enhancement" or "Gentle Warmth"

Custom approach:

  1. EQ: Gentle high shelf +2-3 dB at 8 kHz (adds air), low shelf +2 dB at 100 Hz (warmth)
  2. Multiband Compressor: Bypass (not needed for casual listening unless taming harsh mixes)
  3. Single-Band Compressor: Threshold -18 dB, Ratio 3:1, Analog mode (adds subtle glue)
  4. Limiter: Ceiling -1.0 dBTP (prevents clipping on loud tracks)
A/B Comparison

Use the A/B button frequently. What sounds "better" initially may be fatiguing over time. Trust your ears after 10+ minutes of listening, not the first 30 seconds.

▶ Gear Head Details

Streaming codec limitations: AAC/MP3/Opus codecs sacrifice high-frequency detail and stereo imaging for file size. Gentle EQ boost at 8-12 kHz can restore perceived detail lost to psychoacoustic masking during encoding.

Loudness wars compensation: Modern mastering pushes LUFS to -8 to -6 (Spotify normalizes to -14). This creates brick-walled, fatiguing mixes. Light multiband compression (2:1 ratio, high threshold) can restore dynamics by catching only the loudest peaks.

Podcast & Voice Clarity

Make spoken word content clearer, more intelligible, and easier to listen to—especially in noisy environments or on poor speakers.

Why Use TabDSP

  • Reduce muddiness in voice content you're listening to (common with USB mics in untreated rooms)
  • Boost speech intelligibility without making voices thin
  • Control dynamics (loud laughs, quiet mumbling) for consistent listening level
  • Remove low-frequency rumble (HVAC, traffic) that doesn't carry speech content

Recommended Settings

Preset: "Podcast Voice"

Custom approach:

  1. EQ:
    • High-pass filter at 80 Hz, 24 dB/oct (removes rumble, proximity effect bass)
    • Cut 200-400 Hz by -2 to -4 dB (reduces muddiness)
    • Boost 2-4 kHz by +3 dB (presence, intelligibility)
    • Gentle de-ess at 6-8 kHz: -2 to -3 dB, narrow Q (tames sibilance)
  2. Compressor: Threshold -20 dB, Ratio 4:1, Attack 10ms, Release 100ms, Analog mode, Sidechain HP at 150 Hz
  3. Limiter: Ceiling -2.0 dBTP (prevents ear-piercing peaks during laughter/yelling)
▶ Gear Head Details

Voice intelligibility frequency range: 2-4 kHz contains consonants critical for speech understanding. Boosting this range increases clarity without adding harshness. Avoid boosting 5-8 kHz excessively (sibilance, ear fatigue).

Sidechain HP in voice compression: Prevents low-frequency plosives (p, b, t) and room modes from triggering compression. Detection signal is high-passed at 150 Hz, but output audio retains bass warmth.

Attack/release for speech: 10ms attack catches transients (plosives, hard consonants), 100ms release follows natural speech cadence without pumping.

Video Conferencing Audio Improvement

Enhance audio quality for Zoom, Google Meet, or other web-based conferencing when monitoring other participants or recording meetings.

Why Use TabDSP

  • Compensate for poor microphones and room acoustics on the receiving end
  • Balance volume differences between quiet and loud speakers
  • Reduce fatigue during long meetings (de-essing, rumble removal)
  • Improve intelligibility when recording meetings for later review

Recommended Settings

  1. EQ:
    • High-pass filter at 100 Hz, 12 dB/oct (removes HVAC, mic handling noise)
    • Gentle cut at 300 Hz, -2 dB (reduces boxiness from laptop mics)
    • Boost 3 kHz, +2 dB (speech clarity)
  2. Compressor: Threshold -24 dB, Ratio 3:1, Analog mode (adapts to varying speaker levels)
  3. Limiter: Ceiling -3.0 dBTP (prevents sudden loud sounds like keyboard typing, coughing)
Monitoring Only

TabDSP processes audio you hear, not what you transmit. To process your outgoing audio, use system-level tools (Krisp, RTX Voice) or DAW routing (Reaper, VoiceMeeter).

▶ Gear Head Details

Web conferencing codecs: Opus codec (used by Zoom, Meet) is optimized for speech at 24-48 kbps. High frequencies above 12 kHz are heavily attenuated. EQ boosts above 8 kHz add little value and may amplify codec artifacts.

Analog mode compression: Uses adaptive attack/release based on signal characteristics. Faster attack for transients, slower for sustained speech. Better than fixed time constants for handling multiple speakers with different dynamics.

Audio Analysis & Monitoring Reference

Use TabDSP as a monitoring and analysis tool when listening to reference tracks, working in browser-based DAWs (Soundtrap, BandLab), or checking the playback quality of browser audio. TabDSP processes what you hear—it doesn't affect your recorded or exported audio.

Why Use TabDSP

  • Real-time spectrum analysis with constant-Q FFT (matches musical perception)
  • LUFS metering for loudness reference (see how your browser audio measures against streaming targets)
  • True peak limiting to prevent clipping in your listening experience
  • M/S processing to study stereo balance and mono compatibility of browser playback

Recommended Workflow

  1. Analysis mode: Bypass all modules, use only meters and spectrum analyzer to study frequency balance of reference tracks
  2. Comparison mode: Load factory presets to A/B your listening experience against professional mastering targets
  3. LUFS reference: Monitor Integrated LUFS to understand how audio measures against streaming targets (-14 for Spotify, -16 for Apple Music)
  4. True peak check: Verify True Peak stays below -1.0 dBTP on content you're listening to
  5. M/S solo: Use Mid Solo to hear centered elements (vocal, kick, bass), Side Solo to hear stereo width of browser playback
▶ Gear Head Details

Constant-Q FFT advantages for music: AnalyserNode FFT (8192-point) with constant-Q power summation (Q=12, 512 log-spaced bins) gives resolution proportional to frequency. Better for identifying resonances, overtones, and harmonic content than raw linear FFT display. Runs in browser's native C++ audio engine with zero JavaScript CPU overhead.

LUFS targets by platform:

  • Spotify: -14 LUFS (normalized, gain reduction applied if louder)
  • Apple Music: -16 LUFS (Sound Check normalization)
  • YouTube: -13 to -15 LUFS
  • Broadcast (EBU R128): -23 LUFS
  • Podcast (Apple): -16 LUFS

True peak compliance: Streaming services apply lossy codecs (AAC, Opus) during upload. Codecs can create inter-sample peaks 0.5-3 dB higher than sample peaks. True peak limiting at -1.0 dBTP prevents post-encode clipping.

M/S monitoring: Mid Solo reveals mono compatibility (what listeners hear on phone speakers). Side Solo reveals stereo width. If Side is empty, mix is mono. If Side is louder than Mid, mix may phase-cancel in mono.

Loudness Compliance Reference

Use TabDSP's compliance presets and metering to understand how audio you're listening to measures against broadcast, streaming, or podcast standards. This is a monitoring and learning tool—TabDSP processes your listening experience, not audio you're distributing.

Why Use TabDSP

  • Built-in compliance presets for major standards (EBU R128, ATSC A/85, Spotify, Apple Podcasts)
  • Real-time compliance warnings (caution/warning indicators)
  • True peak limiting with ITU-R BS.1770-4 compliant measurement
  • LUFS metering (Momentary, Short-term, Integrated)

Compliance Presets

TabDSP includes 15 factory compliance presets covering common standards:

Spotify

-14 LUFS, -1.0 dBTP

Apple Music

-16 LUFS, -1.0 dBTP

EBU R128 (Broadcast)

-23 LUFS, -1.0 dBTP

ATSC A/85 (US TV)

-24 LUFS, -2.0 dBTP

Each preset configures the limiter ceiling and provides LUFS target reference. Compliance warnings appear when levels exceed thresholds.

▶ Gear Head Details

Compliance workflow:

  1. Load appropriate compliance preset (e.g., "Spotify" for streaming music)
  2. Monitor Integrated LUFS meter (should converge to target after 10+ seconds)
  3. Check True Peak meter stays below ceiling (orange bar should not exceed preset limit)
  4. Watch for warning indicators: Caution (yellow) = approaching limit, Warning (red) = exceeded

EBU R128 vs ATSC A/85: Both use ITU-R BS.1770 algorithm but differ in targets. EBU (Europe) targets -23 LUFS, ATSC (US) targets -24 LUFS. ATSC allows -2.0 dBTP ceiling (more headroom), EBU uses -1.0 dBTP.

Streaming normalization: Platforms apply gain adjustment to match target loudness. If your mix is -10 LUFS and target is -14 LUFS, platform applies -4 dB gain reduction. If below target, gain is boosted (but limited by true peak ceiling).

Audio Engineering Education

TabDSP serves as an interactive learning platform for understanding audio processing concepts, DSP fundamentals, and professional workflows.

Why Use TabDSP

  • Real-time visual feedback (spectrum, waveforms, meters) shows immediate effect of parameter changes
  • Compare processing modes (Linear vs Minimum Phase EQ, RMS vs Peak detection)
  • Experiment with signal flow order (drag-and-drop module reordering)
  • Learn professional terminology and standards in context

Educational Exercises

  1. EQ Fundamentals:
    • Load "Default" preset, play pink noise or music
    • Add high-pass filter, watch low-end energy disappear on spectrum
    • Compare narrow Q (surgical cut) vs wide Q (musical shaping)
    • Toggle Linear Phase, hear latency difference and phase coherence
  2. Compression Behavior:
    • Load "Podcast Voice" preset, watch GR meter respond to speech dynamics
    • Adjust attack: Fast (catches transients) vs Slow (lets transients through)
    • Adjust ratio: 2:1 (gentle) vs 10:1 (aggressive limiting)
    • Compare RMS (smooth, musical) vs Peak (aggressive, fast)
  3. Loudness Perception:
    • Play consistent music track, note starting LUFS
    • Apply compression + makeup gain, watch LUFS increase without clipping
    • Understand difference between peak level (meters) and perceived loudness (LUFS)
▶ Gear Head Details

Visual learning advantages: Real-time spectrum and waveform displays create immediate cause-and-effect understanding. Seeing frequency content change as you adjust EQ reinforces mental model of filter behavior faster than reading textbooks.

Signal flow pedagogy: Drag-and-drop module reordering demonstrates non-commutative nature of DSP (EQ → Compressor ≠ Compressor → EQ). Visual signal flow with bypass arcs clarifies routing topology.

Professional context: Factory presets and compliance modes expose students to industry standards (EBU R128, ITU-R BS.1770-4) in working context rather than abstract specification documents.

Live Streaming Audio Enhancement

Improve audio quality when monitoring or recording live streams (Twitch, YouTube Live, webinars).

Why Use TabDSP

  • Compensate for poor streamer microphone quality or untreated rooms
  • Balance game audio vs voice levels when recording VODs
  • Reduce fatigue during long streams (de-essing, compression)
  • Enhance music streams with better dynamics and frequency balance

Recommended Settings

For voice/gaming streams: Use "Podcast Voice" preset as starting point

For music streams: Use "Music Enhancement" or "Gentle Warmth"

Custom approach for voice streams:

  • EQ: High-pass 80 Hz, boost 3 kHz (+2 dB), gentle de-ess 6-8 kHz
  • Compressor: Threshold -18 dB, Ratio 4:1, Analog mode
  • Limiter: Ceiling -2.0 dBTP (prevents sudden loud events)
Monitoring Only

TabDSP processes audio you hear, not what streamers transmit or what you broadcast. For outgoing stream processing, use OBS filters or system-level DSP.

▶ Gear Head Details

Streaming codecs: Twitch uses Opus at ~160 kbps, YouTube uses AAC/Opus. Both sacrifice high-frequency detail above 16 kHz. Boosting above 12 kHz has minimal effect and may amplify codec artifacts (pre-echo, aliasing).

Game audio vs voice balance: Use multiband compression to independently control voice (mid frequencies) and game audio (wide spectrum). Set crossover at 500 Hz and 4 kHz to isolate voice band, compress separately from game explosions/music.