MyMediaTools

Convert AMR to AIFC — Free Online Tool

Convert AMR speech recordings to AIFC format, decoding the narrow-band or wide-band Adaptive Multi-Rate audio and re-encoding it as 16-bit big-endian PCM (pcm_s16be) — the native uncompressed codec of Apple's AIFF/AIFC container. This is ideal for bringing mobile voice recordings into professional audio workflows on macOS or Logic Pro.

FFmpeg Command

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Estimated output:

Conversion Complete!

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How It Works

AMR files store audio using the libopencore_amrnb (narrow-band) or libopencore_amrwb (wide-band) codec, which are highly compressed speech codecs designed for mobile telephony — typically encoded at bitrates as low as 4.75 kbps and sampled at 8 kHz (NB) or 16 kHz (WB). During conversion, FFmpeg fully decodes the AMR bitstream into raw PCM audio in memory, then re-encodes it into the AIFC container using pcm_s16be — 16-bit signed PCM with big-endian byte ordering. This means the output is uncompressed and the AIFC file will be significantly larger than the source AMR. Importantly, the audio quality ceiling is limited by the original AMR encoding: any speech artifacts or bandwidth limitations baked into the AMR file are preserved in the output, since upsampling lossy compressed audio cannot recover lost frequency information.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg binary. In the browser-based version of this tool, this runs via FFmpeg.wasm compiled to WebAssembly — no files leave your device.
-i input.amr Specifies the input AMR file. FFmpeg reads the file and automatically detects whether it contains AMR-NB (narrow-band, 8 kHz) or AMR-WB (wide-band, 16 kHz) audio, selecting the appropriate libopencore decoder.
-c:a pcm_s16be Sets the audio codec to pcm_s16be — 16-bit signed PCM with big-endian byte ordering. This is the standard uncompressed audio codec for the AIFC/AIFF container and is natively supported by macOS, Logic Pro, and GarageBand without any additional decoders.
-b:a 128k Specifies a target audio bitrate of 128 kbps. For uncompressed PCM codecs like pcm_s16be, the actual bitrate is fixed by sample rate and bit depth, so this flag does not alter the output — it is included here for structural completeness and has no effect on the AIFC file's audio data.
output.aifc Defines the output filename with the .aifc extension. FFmpeg uses this extension to mux the decoded PCM audio into an AIFC container — the compressed-capable variant of Apple's AIFF format, distinguishable from plain .aiff by its support for non-PCM codecs.

Common Use Cases

  • Importing voice memos or call recordings from older Nokia, Samsung, or feature phones into Logic Pro or GarageBand on macOS, which natively read AIFC files
  • Archiving mobile voice recordings in an uncompressed format for long-term preservation, replacing the fragile AMR codec dependency with standard PCM audio
  • Preparing interview or field recordings captured on Android devices as AMR files for editing in professional audio workstations that expect AIFF/AIFC input
  • Converting AMR voicemail attachments downloaded from telecom providers into a format compatible with macOS audio tools and Automator workflows
  • Delivering speech recordings to post-production teams working on Apple platforms, where AIFC is the expected interchange format for uncompressed audio assets
  • Batch-processing a library of AMR voice recordings into AIFC so they can be imported and normalized in batch within a DAW project

Frequently Asked Questions

No — converting from AMR to AIFC does not improve quality. AMR is a lossy codec, so any compression artifacts, limited frequency range (8 kHz for narrow-band or 16 kHz for wide-band), and speech-specific processing are permanently embedded in the source file. The AIFC output stores that audio as uncompressed PCM, which is an exact lossless representation of the decoded AMR signal — but it cannot reconstruct audio information that was discarded during the original AMR encoding. You will get a larger file with the same audio fidelity ceiling as the AMR source.
AMR is an extremely aggressive compression codec designed for mobile networks, typically operating at 4.75–12.2 kbps. The AIFC output uses pcm_s16be, which is uncompressed PCM audio — a one-minute recording at 16-bit/8 kHz mono translates to roughly 960 KB of raw PCM, compared to perhaps 90 KB in AMR. This size increase is expected and is the cost of converting to an uncompressed, universally readable format. If file size is a concern, consider using a compressed AIFC codec like pcm_alaw or pcm_mulaw, which are also telephony-oriented and can reduce the output size significantly.
To use a different PCM codec supported by AIFC, replace pcm_s16be with an alternative such as pcm_s24be (24-bit, higher dynamic range), pcm_alaw, or pcm_mulaw — for example: ffmpeg -i input.amr -c:a pcm_alaw output.aifc. The -b:a flag has limited effect on uncompressed PCM codecs since their bitrate is determined by sample rate and bit depth, but it is included for compatibility. For pcm_mulaw or pcm_alaw, the output file will be much smaller while remaining in an AIFC container.
Yes. On Linux or macOS, you can use a shell loop: for f in *.amr; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.amr}.aifc"; done. On Windows Command Prompt, use: for %f in (*.amr) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc". Each file is processed sequentially, and the output AIFC files are named to match the source AMR filenames.
AMR files carry minimal metadata — typically just basic tags if any. FFmpeg will attempt to copy any ID3-style or file-level tags present in the AMR container into the AIFC output, but AIFC has its own metadata chunk structure (ANNO, NAME, AUTH chunks) that may not map cleanly from AMR tags. In practice, most AMR voice recordings from mobile phones contain no embedded metadata at all, so this is rarely a concern. If metadata preservation is critical, check the output with a tool like ffprobe.
FFmpeg automatically detects whether the AMR file uses narrow-band (AMR-NB, 8 kHz sample rate, libopencore_amrnb) or wide-band (AMR-WB, 16 kHz sample rate, libopencore_amrwb) and decodes it accordingly. The AIFC output will reflect the original sample rate — an AMR-NB file produces an AIFC file sampled at 8 kHz, while an AMR-WB file produces 16 kHz output. If you need a specific sample rate in the output, add -ar 44100 (or your target rate) to the command to resample: ffmpeg -i input.amr -c:a pcm_s16be -ar 44100 -b:a 128k output.aifc.

Technical Notes

AMR's two sub-formats — narrow-band (AMR-NB) and wide-band (AMR-WB) — have fundamentally different frequency responses: AMR-NB tops out at 4 kHz effective bandwidth (8 kHz sample rate), while AMR-WB reaches 7 kHz (16 kHz sample rate). Neither approaches the 20 kHz ceiling of CD-quality audio, which means the resulting AIFC file, while technically uncompressed, will sound like telephony-grade audio. The pcm_s16be codec used in this conversion stores 16-bit samples in big-endian byte order, which is the native endianness of the AIFC/AIFF format and why Apple platforms handle these files natively. AIFC technically supports compressed audio codecs (hence the 'C' in its name), including pcm_alaw and pcm_mulaw which are also speech-oriented codecs used in telephony — making them semantically aligned with AMR source material. FFmpeg does not support embedding chapter markers or multiple audio tracks in AIFC, and AMR carries neither, so no data loss occurs on those axes. The -b:a flag is carried in the command for structural consistency but has no practical effect on pcm_s16be since uncompressed PCM bitrate is determined entirely by sample rate, bit depth, and channel count.

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