How to Use the afconvert Command on Mac to Batch Convert and Compress Audio Files in Terminal

Transforming uncompressed high-resolution audio files (such as 24-bit 96 kHz WAV or AIFF) into compact, streaming-ready formats like AAC, MP3, or Apple Lossless (ALAC) typically leads users to install third-party applications like Audacity, HandBrake, or FFmpeg. However, macOS includes an enterprise-grade, hardware-accelerated audio conversion engine built directly into the operating system: afconvert (Audio File Convert). Available from any Terminal shell, afconvert provides deep access to Apple’s Core Audio framework for precision audio transcoding, bit-depth dithering, and automated batch scripting.

Understanding afconvert and the Core Audio Pipeline

Unlike third-party converters that rely on external software codecs, afconvert connects directly to Apple’s proprietary AudioToolbox and Core Audio libraries. This integration provides several key benefits:

• Apple AAC Codec Excellence: Core Audio’s AAC encoder is widely recognized by sound engineers as one of the highest-fidelity perceptual audio encoders available, outperforming standard open-source AAC encoders at identical bitrates.
• Zero External Dependencies: Present out-of-the-box on every modern macOS installation (from OS X through macOS Sequoia) with no Homebrew or Xcode requirements.
• Broad Format Compatibility: Reads and writes standard container formats including AIFF (.aif), WAVE (.wav), Core Audio Format (.caf), MPEG-4 Audio (.m4a), and NeXT/Sun audio (.au).

Basic Command Syntax and Format Flags

The standard syntax for converting an audio file requires specifying the output file format (-f) and the underlying audio data format (-d):

afconvert -f [file_format] -d [data_format] [options] input_file [output_file]

To inspect all available container file formats supported by your system’s Core Audio engine:

afconvert -hf

To view all available encoding data formats (such as AAC, ALAC, Linear PCM, and IMA 4:1):

afconvert -hd

Converting WAV or AIFF to High-Quality AAC (M4A)

To compress an uncompressed master recording into a high-quality AAC file wrapped inside an MPEG-4 container:

afconvert -f m4af -d aac -b 256000 input.wav output.m4a

The parameters operate as follows:

• -f m4af: Sets the container format to MPEG-4 Audio (.m4a).
• -d aac: Applies the Advanced Audio Coding (AAC) compression algorithm.
• -b 256000: Sets the target constant bitrate to 256 kbps (Apple Music standard quality).

For even greater fidelity and dynamic efficiency, enable True Variable Bitrate (VBR) encoding by specifying the -u (user property) flag:

afconvert -f m4af -d aac -u pgcm 2 -q 127 -b 256000 master.wav master_vbr.m4a

Where -q 127 commands the Core Audio encoder to operate at maximum algorithmic precision.

Converting to Apple Lossless Audio Codec (ALAC)

If you need to compress large studio masters without discarding a single bit of acoustic data, transcode into Apple Lossless (ALAC):

afconvert -f m4af -d alac input.wav output_lossless.m4a

This typically halves the physical file footprint compared to raw Linear PCM while maintaining mathematically identical waveforms upon decompression.

Resampling and Sample-Rate Conversion

Studio audio recorded at 96 kHz or 192 kHz frequently requires downsampling to standard red-book 44.1 kHz or broadcast 48 kHz:

afconvert -f WAVE -d LEI16@44100 --src-complexity bats input.wav output_44k.wav

Where:

• LEI16@44100: Specifies Little-Endian 16-bit integer PCM at a 44,100 Hz sample rate.
• --src-complexity bats: Instructs the Sample Rate Converter (SRC) to use “Best Quality (Bats)” sinc-interpolation filters, ensuring zero audible aliasing or phase distortion.

Automating Batch Audio Conversion with a Bash Loop

Because afconvert processes a single file per execution, transcode an entire directory of WAV files into AAC using a simple terminal shell loop:

for f in *.wav; do
    echo "Transcoding ${f}..."
    afconvert -f m4af -d aac -b 256000 "${f}" "${f%.wav}.m4a"
done

This shell script iterates across all .wav files in the current folder, generates corresponding .m4a files, and preserves the original lossless master files untouched.