CRITICAL GUIDELINES

Windows File Path Requirements

MANDATORY: Always Use Backslashes on Windows for File Paths

When using Edit or Write tools on Windows, you MUST use backslashes (\) in file paths, NOT forward slashes (/).

Quick Reference

Task	Command
Extract audio	`ffmpeg -i video.mp4 -vn -c:a copy audio.m4a`
Convert to MP3	`ffmpeg -i input.flac -c:a libmp3lame -q:a 2 output.mp3`
Normalize (EBU R128)	`-af loudnorm=I=-23:LRA=7:TP=-2`
Podcast standard	`-af loudnorm=I=-16:TP=-1.5`
Adjust volume	`-af "volume=1.5"` or `-af "volume=6dB"`
Mono to stereo	`-ac 2`

Codec	Recommended Bitrate	Use Case
AAC	128-192k (music), 64k (speech)	Streaming, mobile
MP3	192-320k (music), 128k (speech)	Universal compatibility
Opus	96-128k (music), 48k (speech)	WebM, VoIP, modern

When to Use This Skill

Use for audio-focused operations:

Extracting audio from video files
Loudness normalization for broadcast/streaming compliance
Podcast and audiobook processing
Audio format conversion
Audio effects (EQ, compression, noise reduction)

FFmpeg Audio Processing (2025)

Complete guide to audio encoding, normalization, and professional audio workflows with FFmpeg.

Audio Codec Reference

Codec Comparison

Codec	Encoder	Bitrate Range	Quality	Compatibility	Use Case
AAC	aac, libfdk_aac	64-320 kbps	Excellent	Universal	Streaming, mobile
MP3	libmp3lame	96-320 kbps	Good	Universal	Legacy, podcasts
Opus	libopus	32-256 kbps	Best	Modern	VoIP, WebM
FLAC	flac	~900 kbps	Lossless	Wide	Archival
ALAC	alac	~900 kbps	Lossless	Apple	Apple ecosystem
Vorbis	libvorbis	64-500 kbps	Very Good	Wide	WebM, games
AC3	ac3	192-640 kbps	Good	Universal	DVD, Blu-ray
EAC3	eac3	192-768 kbps	Very Good	Wide	Streaming
xHE-AAC	- (decode only)	12-64 kbps	Excellent	Emerging	Ultra-low bitrate

Recommended Bitrates

Use Case	AAC	MP3	Opus
Podcast/Speech	64-96k	96-128k	48-64k
Music (Standard)	128-192k	192-256k	96-128k
Music (High Quality)	256-320k	320k	160-256k
Transparent Quality	256k+	320k	192k+

Basic Audio Operations

Extract Audio

bash

# Extract to original format (no re-encode)
ffmpeg -i video.mp4 -vn -c:a copy audio.m4a

# Extract to MP3
ffmpeg -i video.mp4 -vn -c:a libmp3lame -b:a 320k audio.mp3

# Extract to AAC
ffmpeg -i video.mp4 -vn -c:a aac -b:a 256k audio.m4a

# Extract to FLAC (lossless)
ffmpeg -i video.mp4 -vn -c:a flac audio.flac

# Extract to WAV (uncompressed)
ffmpeg -i video.mp4 -vn -c:a pcm_s16le audio.wav

Convert Audio Formats

bash

# MP3 to AAC
ffmpeg -i input.mp3 -c:a aac -b:a 256k output.m4a

# WAV to MP3
ffmpeg -i input.wav -c:a libmp3lame -b:a 320k output.mp3

# FLAC to MP3
ffmpeg -i input.flac -c:a libmp3lame -b:a 320k output.mp3

# Multiple files (batch)
for f in *.flac; do
  ffmpeg -i "$f" -c:a libmp3lame -b:a 320k "${f%.flac}.mp3"
done

Quality Settings

bash

# AAC VBR quality (1-5, higher = better)
ffmpeg -i input.wav -c:a aac -q:a 2 output.m4a

# MP3 VBR quality (0-9, lower = better)
ffmpeg -i input.wav -c:a libmp3lame -q:a 0 output.mp3

# Opus with target bitrate
ffmpeg -i input.wav -c:a libopus -b:a 128k output.opus

Audio Normalization

Loudness Standards

Standard	Target	TP (True Peak)	Use Case
EBU R128	-23 LUFS	-1 dBTP	European broadcast
ATSC A/85	-24 LKFS	-2 dBTP	US broadcast
Spotify	-14 LUFS	-1 dBTP	Streaming
YouTube	-14 LUFS	-1 dBTP	Video platform
Apple Music	-16 LUFS	-1 dBTP	Music streaming
Podcast	-16 to -19 LUFS	-1 dBTP	Podcast

EBU R128 Normalization (loudnorm)

Single-Pass (Live/Real-time)

bash

# Quick normalization (less accurate)
ffmpeg -i input.mp3 \
  -af loudnorm=I=-16:TP=-1.5:LRA=11 \
  output.mp3

Two-Pass (Recommended)

bash

# Pass 1: Analyze
ffmpeg -i input.mp3 \
  -af loudnorm=I=-16:TP=-1.5:LRA=11:print_format=json \
  -f null -

# Output will include:
# "input_i": "-25.23"
# "input_tp": "-0.50"
# "input_lra": "8.32"
# "input_thresh": "-35.87"
# "target_offset": "1.23"

# Pass 2: Normalize with measured values
ffmpeg -i input.mp3 \
  -af loudnorm=I=-16:TP=-1.5:LRA=11:measured_I=-25.23:measured_TP=-0.50:measured_LRA=8.32:measured_thresh=-35.87:offset=1.23:linear=true \
  -ar 48000 \
  output.mp3

Two-Pass Script

bash

#!/bin/bash
# loudnorm-2pass.sh

INPUT="$1"
OUTPUT="$2"
TARGET_I="${3:--16}"
TARGET_TP="${4:--1.5}"
TARGET_LRA="${5:-11}"

# Pass 1: Analyze
stats=$(ffmpeg -i "$INPUT" \
  -af loudnorm=I=${TARGET_I}:TP=${TARGET_TP}:LRA=${TARGET_LRA}:print_format=json \
  -f null - 2>&1 | grep -A 12 "Parsed_loudnorm")

# Extract values
input_i=$(echo "$stats" | grep input_i | tr -d '", ' | cut -d':' -f2)
input_tp=$(echo "$stats" | grep input_tp | tr -d '", ' | cut -d':' -f2)
input_lra=$(echo "$stats" | grep input_lra | tr -d '", ' | cut -d':' -f2)
input_thresh=$(echo "$stats" | grep input_thresh | tr -d '", ' | cut -d':' -f2)
offset=$(echo "$stats" | grep target_offset | tr -d '", ' | cut -d':' -f2)

# Pass 2: Normalize
ffmpeg -i "$INPUT" \
  -af "loudnorm=I=${TARGET_I}:TP=${TARGET_TP}:LRA=${TARGET_LRA}:measured_I=${input_i}:measured_TP=${input_tp}:measured_LRA=${input_lra}:measured_thresh=${input_thresh}:offset=${offset}:linear=true" \
  -ar 48000 \
  "$OUTPUT"

Peak Normalization

bash

# Normalize to peak level
ffmpeg -i input.mp3 \
  -af "volume=0dB:eval=once:precision=fixed" \
  -af "loudnorm=I=-16:TP=-1:LRA=11" \
  output.mp3

# Simple peak normalization
ffmpeg -i input.mp3 \
  -filter:a "volume=replaygain=peak" \
  output.mp3

RMS Normalization

bash

# Normalize to specific RMS level
ffmpeg -i input.mp3 \
  -af "loudnorm=I=-23:LRA=7:TP=-2" \
  output.mp3

ffmpeg-normalize Tool

The ffmpeg-normalize Python utility provides an easier interface:

bash

# Install
pip install ffmpeg-normalize

# Basic usage
ffmpeg-normalize input.mp3 -o output.mp3

# Custom target
ffmpeg-normalize input.mp3 -o output.mp3 -t -14

# Batch normalize (album mode - preserves relative loudness)
ffmpeg-normalize *.mp3 --batch -o normalized/

# Use built-in presets (v1.36.0+)
ffmpeg-normalize input.mp3 --preset podcast -o output.mp3
ffmpeg-normalize *.mp3 --preset music --batch -o normalized/

Audio Filters

Volume Control

bash

# Increase volume by 50%
ffmpeg -i input.mp3 -af "volume=1.5" output.mp3

# Increase by 6dB
ffmpeg -i input.mp3 -af "volume=6dB" output.mp3

# Decrease by 3dB
ffmpeg -i input.mp3 -af "volume=-3dB" output.mp3

Fade In/Out

bash

# Fade in 3 seconds, fade out last 3 seconds
ffmpeg -i input.mp3 \
  -af "afade=t=in:ss=0:d=3,afade=t=out:st=57:d=3" \
  output.mp3

# Calculate fade out start automatically
duration=$(ffprobe -v error -show_entries format=duration -of default=noprint_wrappers=1:nokey=1 input.mp3)
fadeout_start=$(echo "$duration - 3" | bc)
ffmpeg -i input.mp3 \
  -af "afade=t=in:ss=0:d=3,afade=t=out:st=${fadeout_start}:d=3" \
  output.mp3

Equalization

bash

# Bass boost
ffmpeg -i input.mp3 \
  -af "equalizer=f=100:width_type=o:width=2:g=5" \
  output.mp3

# Treble reduction
ffmpeg -i input.mp3 \
  -af "equalizer=f=8000:width_type=o:width=2:g=-3" \
  output.mp3

# Multi-band EQ
ffmpeg -i input.mp3 \
  -af "equalizer=f=100:width_type=o:width=2:g=3,equalizer=f=1000:width_type=o:width=2:g=-2,equalizer=f=8000:width_type=o:width=2:g=2" \
  output.mp3

High-Pass / Low-Pass Filters

bash

# High-pass filter (remove below 80Hz)
ffmpeg -i input.mp3 -af "highpass=f=80" output.mp3

# Low-pass filter (remove above 8kHz)
ffmpeg -i input.mp3 -af "lowpass=f=8000" output.mp3

# Band-pass filter
ffmpeg -i input.mp3 -af "highpass=f=80,lowpass=f=12000" output.mp3

Noise Reduction

bash

# FFT-based noise reduction
ffmpeg -i input.mp3 \
  -af "afftdn=nf=-25" \
  output.mp3

# With noise floor adjustment
ffmpeg -i input.mp3 \
  -af "afftdn=nf=-20:tn=1" \
  output.mp3

Compressor / Limiter

bash

# Dynamic range compression
ffmpeg -i input.mp3 \
  -af "acompressor=threshold=-20dB:ratio=4:attack=5:release=50" \
  output.mp3

# Limiter
ffmpeg -i input.mp3 \
  -af "alimiter=limit=0.9:attack=5:release=50" \
  output.mp3

# De-esser
ffmpeg -i input.mp3 \
  -af "deesser=i=0.4:f=4000:w=0.5" \
  output.mp3

Silence Detection/Removal

bash

# Detect silence
ffmpeg -i input.mp3 \
  -af silencedetect=noise=-30dB:d=0.5 \
  -f null -

# Remove silence
ffmpeg -i input.mp3 \
  -af "silenceremove=start_periods=1:start_silence=0.5:start_threshold=-50dB:stop_periods=1:stop_silence=0.5:stop_threshold=-50dB" \
  output.mp3

Channel Operations

Stereo to Mono

bash

# Average both channels
ffmpeg -i stereo.mp3 \
  -af "pan=mono|c0=0.5*c0+0.5*c1" \
  mono.mp3

# Use only left channel
ffmpeg -i stereo.mp3 -af "pan=mono|c0=c0" mono.mp3

# Downmix stereo to mono
ffmpeg -i stereo.mp3 -ac 1 mono.mp3

Mono to Stereo

bash

# Duplicate mono to both channels
ffmpeg -i mono.mp3 -af "pan=stereo|c0=c0|c1=c0" stereo.mp3

# Simple conversion
ffmpeg -i mono.mp3 -ac 2 stereo.mp3

Extract Specific Channels

bash

# Extract left channel
ffmpeg -i stereo.mp3 \
  -filter_complex "[0:a]channelsplit=channel_layout=stereo:channels=FL[left]" \
  -map "[left]" left.mp3

# Extract right channel
ffmpeg -i stereo.mp3 \
  -filter_complex "[0:a]channelsplit=channel_layout=stereo:channels=FR[right]" \
  -map "[right]" right.mp3

5.1 Surround Operations

bash

# Downmix 5.1 to stereo
ffmpeg -i surround.ac3 \
  -af "pan=stereo|FL=0.5*FC+0.707*FL+0.707*BL+0.5*LFE|FR=0.5*FC+0.707*FR+0.707*BR+0.5*LFE" \
  stereo.mp3

# Extract center channel
ffmpeg -i surround.ac3 \
  -filter_complex "[0:a]channelsplit=channel_layout=5.1:channels=FC[center]" \
  -map "[center]" center.mp3

Sample Rate & Bit Depth

Sample Rate Conversion

bash

# Convert to 44.1kHz
ffmpeg -i input.wav -ar 44100 output.wav

# Convert to 48kHz
ffmpeg -i input.wav -ar 48000 output.wav

# High-quality resampling
ffmpeg -i input.wav \
  -af "aresample=resampler=soxr:precision=33:cheby=1" \
  -ar 44100 output.wav

Bit Depth Conversion

bash

# Convert to 16-bit
ffmpeg -i input.wav -c:a pcm_s16le output.wav

# Convert to 24-bit
ffmpeg -i input.wav -c:a pcm_s24le output.wav

# Convert to 32-bit float
ffmpeg -i input.wav -c:a pcm_f32le output.wav

Speed & Pitch

Speed Change (Affects Pitch)

bash

# 2x speed (chipmunk effect)
ffmpeg -i input.mp3 -af "atempo=2.0" output.mp3

# 0.5x speed (slow motion)
ffmpeg -i input.mp3 -af "atempo=0.5" output.mp3

# For >2x, chain filters
ffmpeg -i input.mp3 -af "atempo=2.0,atempo=2.0" output.mp3  # 4x

Pitch Change (Preserves Speed)

bash

# Pitch shift using rubberband
ffmpeg -i input.mp3 \
  -af "rubberband=pitch=1.5" \
  output.mp3

# Pitch shift semitones
ffmpeg -i input.mp3 \
  -af "asetrate=44100*2^(2/12),aresample=44100" \
  output.mp3  # +2 semitones

Trimming & Concatenation

Trim Audio

bash

# Extract 30 seconds starting at 1 minute
ffmpeg -ss 00:01:00 -i input.mp3 -t 00:00:30 -c copy output.mp3

# Extract from 1:00 to 2:30
ffmpeg -ss 00:01:00 -to 00:02:30 -i input.mp3 -c copy output.mp3

Concatenate Audio

bash

# Create file list
echo "file 'part1.mp3'" > list.txt
echo "file 'part2.mp3'" >> list.txt

# Concatenate (same format)
ffmpeg -f concat -safe 0 -i list.txt -c copy output.mp3

# Concatenate with re-encode
ffmpeg -f concat -safe 0 -i list.txt -c:a aac -b:a 256k output.m4a

Crossfade

bash

# Crossfade two files (3 second overlap)
ffmpeg -i part1.mp3 -i part2.mp3 \
  -filter_complex "acrossfade=d=3:c1=tri:c2=tri" \
  output.mp3

Professional Workflows

Podcast Processing

bash

# Complete podcast processing chain
ffmpeg -i raw_podcast.wav \
  -af "highpass=f=80,\
       acompressor=threshold=-20dB:ratio=4:attack=5:release=50,\
       loudnorm=I=-16:TP=-1.5:LRA=11,\
       silenceremove=start_periods=1:start_silence=1:start_threshold=-50dB" \
  -c:a aac -b:a 96k \
  podcast.m4a

Music Mastering Chain

bash

# Mastering chain
ffmpeg -i mix.wav \
  -af "equalizer=f=60:width_type=o:width=1:g=1,\
       equalizer=f=12000:width_type=o:width=1:g=0.5,\
       acompressor=threshold=-12dB:ratio=2:attack=20:release=200,\
       alimiter=limit=0.95,\
       loudnorm=I=-14:TP=-1:LRA=9" \
  -c:a flac \
  master.flac

Broadcast Normalization

bash

# EBU R128 broadcast compliance
ffmpeg -i input.wav \
  -af "loudnorm=I=-23:TP=-1:LRA=11:dual_mono=true" \
  -ar 48000 \
  -c:a pcm_s24le \
  broadcast.wav

Troubleshooting

Common Issues

"loudnorm resamples to 192kHz"

bash

# Force output sample rate
ffmpeg -i input.mp3 \
  -af loudnorm=I=-16:TP=-1.5:LRA=11 \
  -ar 48000 \
  output.mp3

Audio/video sync after processing

bash

# Maintain sync with video
ffmpeg -i video.mp4 \
  -af "loudnorm=I=-16:TP=-1.5:LRA=11,aresample=async=1" \
  -c:v copy \
  output.mp4

Quality loss from re-encoding

Use lossless intermediate format (FLAC, WAV)
Avoid multiple lossy conversions
Use high bitrates for final lossy encode

Audio Analysis & Measurement

Audio Statistics (astats)

Measure comprehensive audio statistics including RMS, peak levels, crest factor, bit depth, and DC offset.

bash

# Full audio statistics
ffmpeg -i input.mp3 -af "astats=metadata=1:reset=1" -f null -

# Output includes:
# - RMS level and peak level
# - Crest factor
# - Dynamic range
# - DC offset
# - Min/Max sample values
# - Number of samples

# Per-channel statistics
ffmpeg -i input.mp3 -af "astats=metadata=1:reset=1:measure_perchannel=all" -f null -

# Measure specific metrics
ffmpeg -i input.mp3 -af "astats=measure_overall=RMS_level+Peak_level:reset=1" -f null -

astats Metrics:

Metric	Description
`DC_offset`	DC bias in signal
`Min_level`	Minimum sample value
`Max_level`	Maximum sample value
`Min_difference`	Minimum sample-to-sample difference
`Max_difference`	Maximum sample-to-sample difference
`Mean_difference`	Average sample-to-sample difference
`RMS_level`	Root Mean Square level (dB)
`Peak_level`	Peak level (dB)
`RMS_peak`	RMS peak level
`RMS_trough`	RMS trough level
`Crest_factor`	Peak to RMS ratio
`Flat_factor`	Flatness measure
`Peak_count`	Number of samples at peak
`Bit_depth`	Actual bit depth used
`Dynamic_range`	Dynamic range in dB

EBU R128 Loudness Measurement (ebur128)

Comprehensive loudness measurement per EBU R128 / ITU-R BS.1770 standards.

bash

# Basic loudness measurement
ffmpeg -i input.mp3 -af "ebur128=peak=true" -f null -

# Output includes:
# - Momentary loudness (M)
# - Short-term loudness (S)
# - Integrated loudness (I)
# - Loudness Range (LRA)
# - True Peak (dBTP)

# Frame-by-frame measurement
ffmpeg -i input.mp3 -af "ebur128=framelog=verbose:peak=true" -f null - 2>&1 | grep Summary

# Generate loudness graph (PNG)
ffmpeg -i input.mp3 \
  -filter_complex "ebur128=video=1:meter=18:gauge=1[v];[v]scale=1280:720[out]" \
  -map "[out]" \
  -frames:v 1 \
  loudness_graph.png

# Create loudness monitoring video
ffmpeg -i input.mp3 \
  -filter_complex "ebur128=video=1:meter=18:gauge=1:scale=absolute[v]" \
  -map "[v]" -map 0:a \
  -c:a copy \
  loudness_monitor.mp4

# Measure with dual mono (speech)
ffmpeg -i input.mp3 -af "ebur128=dualmono=1:peak=true" -f null -

ebur128 Parameters:

Parameter	Description	Values
`video`	Generate video output	0 or 1
`meter`	Loudness meter level	9, 12, 18
`gauge`	Show gauge overlay	0 or 1
`scale`	Scale type	`absolute`, `relative`
`peak`	Measure true peak	`true`, `sample`, `none`
`dualmono`	Dual mono mode	0 or 1
`framelog`	Logging level	`quiet`, `info`, `verbose`

Speech Normalization (speechnorm)

Specialized normalizer for speech content that adapts to dynamic speech patterns.

bash

# Basic speech normalization
ffmpeg -i speech.mp3 -af "speechnorm" output.mp3

# Speech normalization with custom parameters
ffmpeg -i speech.mp3 \
  -af "speechnorm=p=0.95:m=10:r=0.0005:l=1" \
  output.mp3

# Aggressive speech normalization
ffmpeg -i speech.mp3 \
  -af "speechnorm=p=0.9:e=15:r=0.001:l=1" \
  output.mp3

# Podcast processing with speech normalization
ffmpeg -i podcast.wav \
  -af "highpass=f=80,speechnorm=p=0.95:e=12.5:r=0.0005,loudnorm=I=-16:TP=-1.5" \
  -c:a aac -b:a 96k \
  podcast.m4a

speechnorm Parameters:

Parameter	Description	Default	Range
`p`	Peak value target	0.95	0-1
`e`	Expansion factor	12.5	1-50
`r`	Rise time (speed of increase)	0.0005	0-1
`f`	Fall time (speed of decrease)	0.001	0-1
`c`	Compression factor	0	0-1
`l`	Link channels	0	0 or 1

Dialogue Enhancement (dialoguenhance) - FFmpeg 8.0+

Enhance dialogue clarity by separating and boosting voice frequencies.

bash

# Basic dialogue enhancement
ffmpeg -i input.mp4 -af "dialoguenhance" -c:v copy output.mp4

# Custom dialogue enhancement
ffmpeg -i input.mp4 \
  -af "dialoguenhance=original=0.3:enhance=0.7:voice=0.8" \
  -c:v copy output.mp4

# Heavy enhancement for poor recordings
ffmpeg -i input.mp4 \
  -af "dialoguenhance=original=0.2:enhance=0.9" \
  -c:v copy output.mp4

dialoguenhance Parameters:

Parameter	Description	Default	Range
`original`	Original signal mix	1	0-1
`enhance`	Enhanced signal mix	1	0-1
`voice`	Voice clarity boost	2	2-32

3D Audio / Binaural (sofalizer)

Apply HRTF (Head-Related Transfer Function) for 3D audio and binaural processing using SOFA files.

bash

# Basic binaural conversion (requires SOFA file)
ffmpeg -i surround.ac3 \
  -af "sofalizer=sofa=/path/to/hrtf.sofa" \
  binaural.mp3

# With custom gain and rotation
ffmpeg -i surround.ac3 \
  -af "sofalizer=sofa=/path/to/hrtf.sofa:gain=0:rotation=0" \
  binaural.mp3

# Binaural conversion with elevation
ffmpeg -i surround.ac3 \
  -af "sofalizer=sofa=/path/to/hrtf.sofa:elevation=0:radius=1" \
  binaural.mp3

SOFA Files:

Download HRTF databases from: https://sofacoustics.org/data/database/
Common formats: CIPIC, MIT KEMAR, LISTEN databases

sofalizer Parameters:

Parameter	Description	Range
`sofa`	Path to SOFA file	-
`gain`	Additional gain (dB)	-20 to 40
`rotation`	Head rotation (degrees)	-360 to 360
`elevation`	Head elevation (degrees)	-90 to 90
`radius`	Distance scaling	0 to 3
`type`	Interpolation type	`time`, `freq`

Volume Detection

bash

# Detect volume levels
ffmpeg -i input.mp3 -af "volumedetect" -f null -

# Output includes:
# - mean_volume (average)
# - max_volume (peak)
# - histogram data

A-Weighting (aweighting)

Apply A-weighting curve for perceived loudness measurement.

bash

# Apply A-weighting filter
ffmpeg -i input.mp3 -af "aweighting" output_aweighted.wav

# Measure A-weighted levels
ffmpeg -i input.mp3 -af "aweighting,astats=measure_overall=RMS_level" -f null -

Audio Fingerprinting (chromaprint)

Generate audio fingerprints for identification.

bash

# Generate chromaprint fingerprint
ffmpeg -i input.mp3 -f chromaprint -fp_format raw fingerprint.txt

# Generate Base64 fingerprint
ffmpeg -i input.mp3 -f chromaprint -fp_format base64 - 2>&1 | grep FINGERPRINT

Complete Analysis Workflow

bash

#!/bin/bash
# audio-analysis.sh - Complete audio analysis

INPUT="$1"

echo "=== Audio Analysis Report ==="
echo "File: $INPUT"
echo ""

echo "--- Basic Statistics ---"
ffmpeg -i "$INPUT" -af "astats=measure_overall=all" -f null - 2>&1 | grep -A 50 "Parsed_astats"

echo ""
echo "--- Loudness (EBU R128) ---"
ffmpeg -i "$INPUT" -af "ebur128=peak=true" -f null - 2>&1 | grep -E "(Summary|I:|LRA:|Peak:)"

echo ""
echo "--- Volume Detection ---"
ffmpeg -i "$INPUT" -af "volumedetect" -f null - 2>&1 | grep -E "(mean_volume|max_volume)"

echo ""
echo "--- Silence Detection ---"
ffmpeg -i "$INPUT" -af "silencedetect=noise=-50dB:d=0.5" -f null - 2>&1 | grep silence

This guide covers FFmpeg audio processing. For video operations, see the fundamentals skill. For noise reduction details, see ffmpeg-noise-reduction.

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Install this agent skill to your Project

SKILL.md

CRITICAL GUIDELINES

Windows File Path Requirements

Quick Reference

When to Use This Skill

FFmpeg Audio Processing (2025)

Audio Codec Reference

Codec Comparison

Recommended Bitrates

Basic Audio Operations

Extract Audio

Convert Audio Formats

Quality Settings

Audio Normalization

Loudness Standards

EBU R128 Normalization (loudnorm)

Single-Pass (Live/Real-time)

Two-Pass (Recommended)

Two-Pass Script

Peak Normalization

RMS Normalization

ffmpeg-normalize Tool

Audio Filters

Volume Control

Fade In/Out

Equalization

High-Pass / Low-Pass Filters

Noise Reduction

Compressor / Limiter

Silence Detection/Removal

Channel Operations

Stereo to Mono

Mono to Stereo

Extract Specific Channels

5.1 Surround Operations

Sample Rate & Bit Depth

Sample Rate Conversion

Bit Depth Conversion

Speed & Pitch

Speed Change (Affects Pitch)

Pitch Change (Preserves Speed)

Trimming & Concatenation

Trim Audio

Concatenate Audio

Crossfade

Professional Workflows

Podcast Processing

Music Mastering Chain

Broadcast Normalization

Troubleshooting

Common Issues

Audio Analysis & Measurement

Audio Statistics (astats)

EBU R128 Loudness Measurement (ebur128)

Speech Normalization (speechnorm)

Dialogue Enhancement (dialoguenhance) - FFmpeg 8.0+

3D Audio / Binaural (sofalizer)

Volume Detection

A-Weighting (aweighting)

Audio Fingerprinting (chromaprint)

Complete Analysis Workflow