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How to convert audio with FFmpeg: WAV, MP3, AAC, FLAC, and batch processing

TL;DR

FFmpeg converts audio formats with one command: ffmpeg -i input.wav output.mp3. Use -b:a for bitrate, -ar for sample rate, and -ac for channel count. For batch conversion, loop over files in your shell or PowerShell. If the converted audio is sent to an AI audio API, validate the API request and response before wiring it into production.

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Introduction

Audio conversion is one of FFmpeg’s most common use cases. You might receive a WAV file from a recording workflow and need MP3 for web delivery, or get FLAC from a production pipeline and need AAC for a mobile app.

This guide shows how to:

  • Convert between common audio formats
  • Control bitrate, sample rate, and channel count
  • Batch-process files
  • Trim audio during conversion
  • Prepare audio for AI API workflows
  • Troubleshoot common FFmpeg errors

Installation

Install FFmpeg first if it is not already available on your system.

macOS 

brew install ffmpeg
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Ubuntu/Debian 

sudo apt install ffmpeg
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Windows

Download FFmpeg from ffmpeg.org, then add the bin folder to your PATH.

Verify the installation:

ffmpeg -version
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Basic conversion syntax

The general pattern is:

ffmpeg -i input_file.ext output_file.ext
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FFmpeg usually infers the output format from the file extension. For simple conversions, that is enough.

Example:

ffmpeg -i input.wav output.mp3
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Common audio conversions

WAV to MP3 

ffmpeg -i recording.wav -b:a 192k output.mp3
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The -b:a 192k flag sets the audio bitrate to 192 kbps.

Common bitrate values:

  • 128k — standard web quality
  • 192k — good quality, common for music streaming
  • 320k — high quality, larger file size

For spoken word audio such as podcasts or voice notes, 128k is usually enough. For music, use 192k or higher.

FLAC to AAC 

ffmpeg -i lossless.flac -c:a aac -b:a 256k output.aac
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The -c:a aac flag explicitly selects the AAC audio codec. FFmpeg can often infer this from the output extension, but setting the codec directly makes the command clearer and more reliable.

MP3 to WAV 

ffmpeg -i compressed.mp3 output.wav
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This decodes MP3 into PCM WAV. The output file will be much larger because WAV is uncompressed.

OGG to MP3 

ffmpeg -i podcast.ogg -b:a 192k output.mp3
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M4A to MP3 

ffmpeg -i track.m4a -b:a 192k output.mp3
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Audio quality flags

Set bitrate

Use -b:a to set a constant audio bitrate:

ffmpeg -i input.wav -b:a 192k output.mp3
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For variable bitrate MP3 output, use -q:a:

ffmpeg -i input.wav -q:a 2 output.mp3
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For MP3, -q:a ranges from 0 to 9:

  • 0 — highest quality
  • 2 — high quality
  • 5-7 — acceptable for speech in many cases
  • 9 — lowest quality

Set sample rate

Use -ar to set the output sample rate:

ffmpeg -i input.wav -ar 44100 output.mp3
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Common values:

  • 22050 — smaller files, lower quality
  • 44100 — standard CD-quality audio
  • 48000 — common for professional audio and video workflows

Set channel count

Use -ac to set the number of audio channels:

ffmpeg -i stereo.wav -ac 1 mono.mp3
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-ac 1 converts stereo audio to mono. This is useful for voice content where stereo does not add value.

Control output codec and quality

For MP3, use libmp3lame when you want explicit control over the encoder:

ffmpeg -i input.wav -c:a libmp3lame -b:a 192k -ar 44100 -ac 2 output.mp3
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This command:

  • Reads input.wav
  • Encodes with libmp3lame
  • Sets bitrate to 192k
  • Sets sample rate to 44100
  • Outputs stereo audio with -ac 2

For AAC:

ffmpeg -i input.wav -c:a aac -q:a 1.5 output.aac
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Batch processing

Batch convert WAV to MP3 on macOS/Linux 

for file in *.wav; do
  ffmpeg -i "$file" -b:a 192k "${file%.wav}.mp3"
done
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This converts every .wav file in the current directory to .mp3.

The expression:

${file%.wav}
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removes the .wav extension before appending .mp3.

Batch convert into an output directory 

mkdir output

for file in *.flac; do
  ffmpeg -i "$file" -c:a aac -b:a 256k "output/${file%.flac}.aac"
done
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This keeps the original files in place and writes converted files to output/.

Batch convert on Windows PowerShell 

Get-ChildItem *.wav | ForEach-Object {
  $output = $_.BaseName + ".mp3"
  ffmpeg -i $_.FullName -b:a 192k $output
}
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Trim audio during conversion

You can trim and convert in the same command.

Start at 1 minute 30 seconds and export 60 seconds:

ffmpeg -i input.mp3 -ss 00:01:30 -t 60 -b:a 192k clip.mp3
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Flags:

  • -ss 00:01:30 — start at 1 minute 30 seconds
  • -t 60 — capture 60 seconds

Use -to when you want to specify the end timestamp instead of duration:

ffmpeg -i input.mp3 -ss 00:01:30 -to 00:02:30 output.mp3
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Common errors and fixes

No such file or directory

Check the input path. If the filename contains spaces, wrap it in quotes:

ffmpeg -i "my file.wav" output.mp3
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Invalid data found when processing input

The file may be corrupted or the extension may not match the actual format.

Inspect the file:

ffmpeg -i filename
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FFmpeg will print metadata about the file and detected streams.

Conversion failed! with codec errors

Specify the audio codec explicitly.

For MP3:

ffmpeg -i input.wav -c:a libmp3lame -b:a 192k output.mp3
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For AAC:

ffmpeg -i input.wav -c:a aac -b:a 256k output.aac
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Output file is larger than expected

You may be converting from a compressed format such as MP3 to an uncompressed format such as WAV.

For example:

ffmpeg -i input.mp3 output.wav
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This usually creates a much larger file because WAV is uncompressed.

Integrating with AI audio APIs

Audio conversion is often a preprocessing step before sending audio to AI APIs for transcription, sentiment analysis, or speaker identification.

A typical pipeline:

  1. Receive audio in any format, such as WAV, MP3, or M4A.
  2. Convert it to the format required by the AI API.
  3. Send the converted file with a POST request.
  4. Validate the API response before using it in your application.

Many speech APIs expect WAV audio with:

  • 16 kHz sample rate
  • Mono channel
  • 16-bit PCM encoding

Convert an M4A file into that format:

ffmpeg -i input.m4a -ar 16000 -ac 1 -c:a pcm_s16le processed.wav
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Flags:

  • -ar 16000 — sets the sample rate to 16 kHz
  • -ac 1 — converts to mono
  • -c:a pcm_s16le — outputs 16-bit little-endian PCM

Test the API request

After conversion, test the API endpoint before integrating it into application code.

Example request:

POST https://api.speech-service.example.com/v1/transcribe
Authorization: Bearer {{API_KEY}}
Content-Type: multipart/form-data

[audio_file: processed.wav]
[language: en-US]
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Add response checks such as:

Status code is 200
Response body has field transcript
Response body field confidence is greater than 0.7
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These checks help catch unexpected response structures before they cause silent production errors.

FAQ

Does FFmpeg lose quality when converting between compressed formats?

Yes. Converting from MP3 to AAC, for example, decodes and re-encodes the audio, which introduces generation loss.

If possible, convert from the original uncompressed source. For archival workflows, use a lossless format.

What is the best format for web audio?

MP3 has the widest browser support. AAC is slightly more efficient at the same bitrate. OGG Vorbis is open-source and provides good quality.

For most web use cases, MP3 at 128k to 192k is the safest choice.

How do I check what format an audio file actually uses?

Run:

ffmpeg -i filename.mp3
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FFmpeg prints file information without converting. Look for the Audio: line to see the codec, sample rate, bitrate, and channel layout.

Can FFmpeg convert audio and video together?

Yes. For example, this command copies the video stream and re-encodes only the audio:

ffmpeg -i input.mp4 -c:v copy -c:a aac output.mp4
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-c:v copy avoids re-encoding the video, which is faster than processing both streams.

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