Convert AIFC to AMR — Free Online Tool
Convert AIFC audio files to AMR format using the libopencore_amrnb codec, optimized for speech and mobile telephony. This tool transforms Apple's professional PCM or compressed audio container into a low-bitrate AMR file designed for voice transmission on mobile networks.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
AIFC files store audio using big-endian PCM codecs (such as pcm_s16be or pcm_s24be) or compressed variants like A-law and μ-law, typically at high sample rates and bit depths suited for professional audio work. During conversion, FFmpeg decodes the AIFC audio stream fully — regardless of whether it was stored as lossless PCM or a compressed variant — and then re-encodes it using the libopencore_amrnb codec. AMR-NB operates at a fixed 8kHz sample rate and is engineered specifically for speech intelligibility at very low bitrates (as low as 4.75 kbps). This means the output is significantly downsampled and optimized for voice, so any music, wide-frequency audio, or stereo content in the original AIFC will be reduced to mono 8kHz audio — a deliberate tradeoff for mobile compatibility.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all decoding, resampling, and re-encoding in this conversion pipeline. |
-i input.aifc
|
Specifies the input AIFC file. FFmpeg reads the AIFC container and decodes whichever audio codec is present — whether lossless PCM (pcm_s16be, pcm_s24be) or a compressed variant like pcm_alaw — into raw audio for processing. |
-c:a libopencore_amrnb
|
Selects the libopencore_amrnb encoder to produce AMR-NB (Narrowband) output. This codec forces the audio to 8kHz mono and encodes it using the CELP speech model, which is the standard format used in mobile telephony and voice messaging systems. |
-b:a 12200
|
Sets the AMR-NB bitrate to 12200 bps, which is the highest mode available in AMR-NB (Mode 7, also called MR122). This provides the best speech quality the AMR-NB format can deliver while remaining compatible with all AMR-capable devices. |
output.amr
|
Defines the output filename with the .amr extension, which tells FFmpeg to wrap the encoded libopencore_amrnb audio in the standard AMR file container used by mobile phones, VoIP systems, and multimedia messaging applications. |
Common Use Cases
- Archiving voice memos or dictation recordings originally captured in AIFC format into a compact AMR file for storage on a mobile device or older phone
- Preparing professional studio voice-over recordings exported as AIFC for use in mobile telephony systems or IVR platforms that require AMR input
- Converting Apple Logic Pro or GarageBand AIFC voice session exports into AMR for embedding in multimedia messaging (MMS) on mobile networks
- Reducing the file size of AIFC speech recordings captured at 24-bit or 32-bit depth for transmission over bandwidth-constrained channels such as satellite phones
- Repurposing AIFC-format interview or podcast audio into AMR for compatibility with legacy VoIP or mobile softphone applications
- Converting μ-law or A-law AIFC telephony recordings back into a mobile-compatible AMR file for cross-system interoperability in telecom workflows
Frequently Asked Questions
Yes — this conversion involves substantial quality reduction by design. AIFC stores audio at high fidelity (often 16-bit, 24-bit, or 32-bit PCM at 44.1kHz or higher), while AMR-NB downsamples everything to 8kHz mono and targets speech intelligibility rather than audio fidelity. Music, stereo imaging, and high-frequency content will be lost. For voice recordings or speech, the result is perfectly usable on mobile systems, but it is not appropriate for music or broadcast audio.
AMR-NB (Adaptive Multi-Rate Narrowband) was designed specifically for mobile telephony, where network bandwidth is extremely limited. The 8kHz sample rate captures only the frequency range critical for human speech intelligibility (roughly 300Hz–3400Hz), which is sufficient for phone calls. FFmpeg automatically resamples your AIFC file from its original sample rate down to 8kHz during this conversion — there is no way to retain a higher sample rate within the AMR-NB format.
Replace the value after '-b:a' with any AMR-NB supported bitrate: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200 (bps). The command already uses the highest AMR-NB mode at 12200 bps, which gives the best speech quality the format allows. Lowering it to 4750 saves more space but may introduce more compression artifacts on complex speech. For example: 'ffmpeg -i input.aifc -c:a libopencore_amrnb -b:a 7950 output.amr'.
Yes, AMR-WB (Wideband) supports 16kHz audio and offers noticeably better speech quality than AMR-NB. To switch, change the codec flag in the command to '-c:a libopencore_amrwb' and use a compatible bitrate such as 23850 bps. AMR-WB is supported by most modern smartphones and VoIP systems, but some older mobile devices only support AMR-NB. The output file extension remains .amr.
No. AMR is a bare audio format with minimal metadata support, and most AIFC metadata fields — including embedded markers, loop points, track names, and instrument data — will not carry over to the AMR file. If metadata preservation is important, consider converting to a format like M4A or MP3 before archiving, and keep the original AIFC file as the authoritative version.
Yes, on the command line you can use a shell loop to batch process files. On Linux or macOS, run: 'for f in *.aifc; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.aifc}.amr"; done'. On Windows PowerShell, use: 'Get-ChildItem *.aifc | ForEach-Object { ffmpeg -i $_.FullName -c:a libopencore_amrnb -b:a 12200 ($_.BaseName + ".amr") }'. This processes every AIFC file in the current directory with the same settings used by this browser tool.
Technical Notes
AIFC supports a range of big-endian PCM codecs (pcm_s16be, pcm_s24be, pcm_s32be, pcm_f32be, pcm_f64be) as well as telephony-oriented codecs like pcm_alaw and pcm_mulaw, making it a flexible professional container. When converting to AMR, all of this is decoded to raw PCM first and then re-encoded — there is no stream copying possible between these formats. The libopencore_amrnb encoder enforces hard constraints: mono channel, 8000 Hz sample rate, and one of eight fixed bitrate modes. FFmpeg will automatically downmix stereo AIFC content to mono and resample to 8kHz, which can cause phasing artifacts if the stereo channels contained significantly different audio. The AMR format does not support chapters, multiple audio tracks, or subtitles. Because AMR-NB uses a codebook-based speech model (CELP), it degrades gracefully on clean speech but can produce buzzy or robotic artifacts on non-speech audio such as music or ambient sound from AIFC files. The output file size will be dramatically smaller than the source — a 10MB AIFC file at 24-bit/44.1kHz will typically compress to well under 100KB in AMR-NB at 12200 bps.