MyMediaTools

Compress AIFC Online — Free File Size Reducer

Compress or re-encode an AIFC audio file to another AIFC file using PCM big-endian encoding, giving you control over the audio codec and bitrate within Apple's AIFF-C container. This is useful for standardizing AIFC files to a consistent PCM format or reducing bitrate on compressed AIFC streams.

FFmpeg Command

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How It Works

AIFC (AIFF-C) is a superset of Apple's AIFF format that can store both uncompressed PCM audio (in big-endian byte order) and compressed audio streams such as A-law or μ-law. During this conversion, FFmpeg reads the input AIFC file — which may contain any supported AIFC codec including pcm_s24be, pcm_f32be, pcm_alaw, or pcm_mulaw — and re-encodes the audio stream to 16-bit signed big-endian PCM (pcm_s16be) at 128k bitrate, writing the result back into an AIFC container. Because both source and destination are AIFC and the output codec is pcm_s16be, this is a full re-encode rather than a remux. If the source was higher bit-depth (e.g., 24-bit or 32-bit float), the output will be downconverted to 16-bit, which reduces file size but also reduces dynamic range headroom.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg binary, the open-source multimedia processing engine that powers this conversion. In the browser version, this runs as a WebAssembly module via FFmpeg.wasm with no server upload required.
-i input.aifc Specifies the input AIFC file to be processed. FFmpeg will detect the AIFC container and identify the internal audio codec — which could be pcm_s16be, pcm_s24be, pcm_alaw, pcm_mulaw, or another supported AIFC codec — before applying the conversion.
-c:a pcm_s16be Sets the output audio codec to 16-bit signed big-endian PCM, the standard uncompressed format within the AIFC container. This re-encodes whatever codec was in the source AIFC — including compressed formats like A-law or higher bit-depth PCM — into uniform 16-bit big-endian samples.
-b:a 128k Specifies a target audio bitrate of 128 kilobits per second. For uncompressed pcm_s16be audio, the actual bitrate is fixed by sample rate and channel count rather than this parameter, so this flag primarily takes effect if the command is modified to use a compressed AIFC codec such as pcm_alaw or pcm_mulaw.
output.aifc Defines the output filename and tells FFmpeg to write the result into an AIFC (AIFF-C) container. The .aifc extension ensures the file is recognized by Apple tools, samplers, and professional audio software that expect the AIFF-C container structure.

Common Use Cases

  • Standardizing a batch of AIFC files from a legacy Mac audio system where some files use 24-bit or float PCM to a uniform 16-bit PCM AIFC format for compatibility with older Apple software or hardware samplers.
  • Converting a compressed AIFC file that uses A-law or μ-law encoding (common in telephony workflows) to uncompressed pcm_s16be for editing in a DAW that only handles uncompressed AIFC.
  • Reducing the file size of 32-bit or 64-bit float AIFC master files to 16-bit AIFC for archival distribution while keeping the AIFC container intact for downstream Apple ecosystem tools.
  • Re-encoding an AIFC file with an unrecognized or problematic codec variant to a clean pcm_s16be AIFC that is universally supported by professional audio applications like Logic Pro and Pro Tools.
  • Preparing AIFC audio assets from a game or multimedia project for import into a platform that requires standard 16-bit big-endian PCM within the AIFF-C container structure.

Frequently Asked Questions

Yes, downconverting from a higher bit depth to 16-bit PCM is a lossy operation in terms of dynamic range. A 24-bit AIFC has up to 144 dB of theoretical dynamic range, while 16-bit offers around 96 dB. For most listening purposes this is inaudible, but if you are working on a mastering chain or need to preserve every bit of the original resolution, you should choose pcm_s24be as your output codec instead by modifying the -c:a flag in the FFmpeg command.
This tool is specifically designed to process AIFC-to-AIFC — useful when you need to change the internal codec, bit depth, or bitrate of an AIFC file while keeping it within the AIFF-C container. The AIFC container is required by certain Apple legacy tools, hardware samplers, and professional audio systems that expect the AIFF-C file structure. If you need a different output container such as WAV or standard AIFF, you would use a different conversion tool.
Yes, FFmpeg will still run the audio through its encode/decode pipeline rather than stream-copying it, because the command specifies an explicit -c:a pcm_s16be codec. This means a lossless pcm_s16be-to-pcm_s16be re-encode occurs, which is mathematically lossless for PCM — every sample value is preserved exactly. The only practical effect is a small amount of additional processing time.
Yes. In the displayed FFmpeg command, replace pcm_s16be with any codec supported by the AIFC container: pcm_s24be, pcm_s32be, pcm_f32be, pcm_f64be, pcm_alaw, or pcm_mulaw. For example, use -c:a pcm_s24be to preserve 24-bit depth, or -c:a pcm_alaw to produce a telephony-style compressed AIFC file. Note that if you choose a compressed codec like pcm_alaw, the -b:a bitrate flag has less effect since those codecs operate at fixed rates.
On Linux or macOS, you can wrap the command in a shell loop: for f in *.aifc; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "compressed_${f}"; done. On Windows Command Prompt, use: for %f in (*.aifc) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "compressed_%f". This will process every AIFC file in the current directory and write compressed output files with a prefixed filename.
FFmpeg will attempt to copy recognized metadata from the AIFC container, but AIFC-specific chunks such as MIDI instrument data, loop point markers (MARK and INST chunks), and application-specific chunks may not be preserved during re-encoding. If your AIFC files contain loop or instrument metadata critical to a sampler workflow, verify the output file in a dedicated audio tool before discarding the originals.

Technical Notes

AIFC (AIFF-C) uses big-endian byte ordering for all its PCM variants, which distinguishes it from WAV-based PCM formats that use little-endian storage. The codec pcm_s16be specifically means signed 16-bit integer samples stored in big-endian order — this is the most universally supported format within the AIFC container and is directly compatible with classic Mac OS, Logic Pro, and many hardware samplers. The AIFC format supports a wide range of bit depths (16, 24, 32-bit integer and 32/64-bit float), so re-encoding from a higher bit depth to pcm_s16be will truncate the lower-significance bits, introducing quantization noise at an amplitude roughly 144 dB below full scale — well below audible thresholds for most content. The -b:a 128k flag is technically a target bitrate hint, but for uncompressed PCM codecs like pcm_s16be, the actual bitrate is determined entirely by sample rate and channel count, not by the -b:a value. This flag has meaningful effect only if you switch to a compressed AIFC codec such as pcm_alaw or pcm_mulaw. AIFC does not support video, subtitles, chapters, or multiple audio tracks, so this tool operates solely on the single audio stream. Files using proprietary AIFC compression types outside FFmpeg's supported codec list (e.g., Apple's MACE 3:1 or MACE 6:1) may fail to decode and will require specialized tools.

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