Voice cleanup and background-noise removal

Read this when the request names noise, hum, room tone, echo, or asks to "clean up the audio."

Treat noise removal as restoration, not mastering. The goal is to reduce non-speech sound while preserving the speaker's tone, timing, breaths, consonants and perceived volume. A cleaner signal must not become louder merely because its noise floor fell.

The local, open-source workflow

Preserve the original audio unchanged. Lock the source cuts, then export the exact cut-only audio as uncompressed PCM. Avoid unnecessary resampling; DeepFilterNet3 expects 48kHz.
```
ffmpeg -i final-cut.mp4 -vn -ac 2 -ar 48000 -c:a pcm_s24le voice-source.wav
```
Select and record representative noise-only and speech windows before processing. Measure their mean or RMS level and peak level so the result can be compared against the same windows later.
Run a local open-source speech enhancer. DeepFilterNet3 is the proven default:
```
deepFilter --pf --atten-lim 30 --output-dir enhanced voice-source.wav
```
Record the package version, model or checkpoint, license, attenuation limit and post-filter setting. Treat 30dB as a tested upper bound, not a universal value; reduce it if speech becomes metallic, watery, phasey or over-processed.
Listen to the complete enhanced track before EQ or mastering. Reject clipped consonants, missing word tails, pumping, metallic or watery tone, exaggerated sibilance, damaged plosives, or speech removed during pauses. Do not use a hard noise gate by default.

Apply only conservative corrective filtering when needed. These frequencies are starting points, not mandatory values:

ffmpeg -i enhanced/voice-source_DeepFilterNet3_pf.wav \
  -af 'highpass=f=70:p=2,lowpass=f=15500:p=2' \
  -ar 48000 -c:a pcm_s24le voice-clean.wav

Master loudness as a separate step. Measure first:
```
ffmpeg -i voice-clean.wav -af 'ebur128=peak=true' -f null -
```
Never apply a large fixed gain merely because denoising made the waveform quieter. A modest lift is allowed when it materially improves intelligibility; normally start between +1 and +3 LU relative to the source and confirm it by A/B listening. Unless the user or delivery platform specifies a target, use about -18 LUFS integrated as a conservative starting point for close-mic speech and prefer clear over loud or fatiguing. Keep final true peak at or below -1.5dBTP. Compression is optional; if needed, use a low ratio and level-match the before/after comparison so louder does not masquerade as cleaner.
Mux the exact processed voice master into the delivery. Do not add music, ambience or decorative sound unless requested.
Verify synchronization after processing and final encoding. Enhancement, delay compensation, AAC priming and container timing can introduce offsets. Correlate the decoded delivery audio against the voice master and target an absolute offset of 20ms or less. If timing changed, realign captions to the delivered audio.

Required QA

Keep the original audio immutable.
Use only locally runnable open-source components when the user prohibits paid or closed tools.
Record component, model, checkpoint and license provenance.
Compare the same noise-only and speech windows before and after cleanup.
A/B at matched perceived volume; judge cleanup independently from loudness.
Report integrated loudness, loudness range, true peak and the gain change from the cleaned signal.
Listen for artifacts across the full track, including pauses and word boundaries.
Confirm that silence remains natural and no gate clips consonants or breaths.
Decode the complete final stream and verify audio/video duration and sync.
Disclose any unavoidable residual noise or processing artifact.

Field note: one edit reached approximately -16.1 LUFS integrated and -1.3dBFS encoded peak. It was technically clean but the speaker judged the voice too loud. Therefore -16 LUFS is not a safe universal default; use only a modest lift when it improves clarity, and evaluate denoising success separately from delivery loudness.