audio-edit

Audio editing (ffmpeg)

Loudness normalize, denoise, and background-music mixing. These are plain ffmpeg one-liners — no CLI command wraps them. Every recipe stream-copies the video (-c:v copy, fast + lossless) and re-encodes only the audio to aac.

Pick the operation:

Audio too quiet / loud / inconsistent across clips → normalize.
Background hiss / hum / fan noise → denoise.
Lay a music bed under narration → add music (use --duck-style sidechain so music drops under speech).
Cutting um/uh hesitations → use the filler-removal skill, not this one.

Normalize loudness

ffmpeg -y -i in.mp4 -af loudnorm=I=-16:TP=-1.5:LRA=11 -c:v copy -c:a aac out.mp4

-16 LUFS / -1.5 dBTP is a good general/web target. The single pass above is fine for most clips. For precise targets (or batch consistency), do two-pass: run once with loudnorm=...:print_format=json, read the measured values, then pass them back as measured_I/measured_TP/measured_LRA/measured_thresh on a second run.

Even out inconsistent levels (loud speaker vs quiet audience)

The most common "fix the audio" complaint on talks/panels/interviews isn't noise — it's level swings: the presenter is loud and near the mic, audience questions are faint, peaks nearly clip. Compress to tighten the dynamic range, then normalize:

ffmpeg -y -i in.mp4 -af \
  "acompressor=threshold=-21dB:ratio=3:attack=15:release=250:makeup=3,\
   equalizer=f=2800:t=q:w=2:g=2,\
   loudnorm=I=-14:TP=-1.5:LRA=9" \
  -c:v copy -c:a aac out.mp4

acompressor pulls the loud parts down; makeup lifts everything so the quiet parts come up — net result is consistent loudness. Lower threshold / higher ratio = more leveling.
The gentle equalizer presence bump (~2.5–3.5 kHz) adds intelligibility.
loudnorm with a tight LRA (7–9) finishes the leveling; check the output isn't pumping.
Diagnose first with astats (RMS level dB / Peak level dB) — a wide RMS-to-peak gap or a peak near 0 dB confirms it needs compression, not denoise.

Denoise (hiss / hum)

ffmpeg -y -i in.mp4 -af afftdn=nr=12 -c:v copy -c:a aac out.mp4

afftdn is an FFT denoiser — no extra deps. nr is noise reduction in dB (10–20 typical; higher = more aggressive but more artifacts/underwater sound — start at 12 and listen). Combine with normalize in one pass: -af "afftdn=nr=12,loudnorm=I=-16:TP=-1.5:LRA=11".

Add background music

If the edit is an EDL, put the bed in the EDL instead — "music": {"src": …, "gain_db": -18, "duck": true, "fade_out": 2.0} mixes it in the same render pass (looped to cover the whole cut), so you don't pay for a second re-encode. The recipes below are for a video that's already finished, or for a one-off mix outside an EDL.

Plain mix (music sits at a fixed level under the existing audio):

ffmpeg -y -i in.mp4 -i bg.mp3 -filter_complex \
  "[1:a]volume=-18dB[bg];[0:a][bg]amix=inputs=2:duration=first[a]" \
  -map 0:v -map "[a]" -c:v copy -c:a aac -shortest out.mp4

Music ducked under speech (music automatically drops when someone talks) — usually what you want for narration:

ffmpeg -y -i in.mp4 -i bg.mp3 -filter_complex \
  "[1:a]volume=-18dB[bg];\
   [bg][0:a]sidechaincompress=threshold=0.03:ratio=8:attack=20:release=300[bgd];\
   [0:a][bgd]amix=inputs=2:duration=first[a]" \
  -map 0:v -map "[a]" -c:v copy -c:a aac -shortest out.mp4

The sidechain compresses the music ([bg]) using the original speech ([0:a]) as the trigger: louder speech → more music attenuation. Tune ratio (duck depth), attack (how fast it ducks, ms), release (how fast music returns, ms).

Gotchas

Always -shortest when adding music, or a longer music track extends the clip past the video and everything downstream desyncs.
amix can lower overall level — if the result sounds quiet, append ,loudnorm=I=-16:TP=-1.5:LRA=11 after the amix output, or raise the speech with volume before mixing.
Re-encode audio to aac (-c:a aac); don't -c:a copy after a filter.
duration=first ties output length to the first amix input (the speech) — keep speech first in the amix chain.