Convert AMR to AIFF — Free Online Tool
Convert AMR audio files to AIFF format, decoding the compressed Adaptive Multi-Rate speech codec into uncompressed PCM audio (pcm_s16be) for full-fidelity playback and editing on macOS. Ideal for bringing mobile voice recordings into professional audio workflows where lossless storage is required.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AMR file here
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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
AMR files store audio using a highly compressed speech-optimized codec (libopencore_amrnb for narrowband or libopencore_amrwb for wideband) designed for mobile telephony, typically at bitrates between 4.75 and 12.2 kbps with a narrow frequency range tuned for voice. During conversion, FFmpeg fully decodes the AMR bitstream into raw PCM samples, then re-encodes them as 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFF container — Apple's uncompressed audio format. This is a lossy-to-lossless conversion in container terms: the AIFF file is lossless and uncompressed, but it cannot recover audio detail that was discarded during the original AMR encoding. The resulting AIFF will be dramatically larger than the source AMR file, since uncompressed audio at 16-bit/8kHz or 16kHz occupies far more space than the compressed speech data.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser, this runs via FFmpeg.wasm compiled to WebAssembly, executing entirely on your local machine without any server upload. |
-i input.amr
|
Specifies the input file — an AMR audio file encoded with the Adaptive Multi-Rate codec (typically AMR-NB at 8 kHz, narrowband). FFmpeg auto-detects the AMR container and selects the appropriate libopencore decoder. |
-c:a pcm_s16be
|
Sets the audio codec for the output to 16-bit signed big-endian PCM, the standard uncompressed audio encoding for AIFF files. This fully decodes the lossy AMR speech data into raw, editor-ready PCM samples compatible with macOS audio tools and DAWs. |
output.aiff
|
Defines the output filename and container format. The .aiff extension tells FFmpeg to wrap the pcm_s16be audio stream in Apple's Audio Interchange File Format — an uncompressed, lossless container natively supported by macOS, Logic Pro, GarageBand, and Final Cut Pro. |
Common Use Cases
- Importing voice memos or call recordings from older Nokia, Samsung, or feature phones into GarageBand, Logic Pro, or other macOS DAWs that natively read AIFF
- Preparing transcribed interview recordings captured on Android devices in AMR format for editing in Final Cut Pro or Audacity on a Mac, where AIFF is the preferred lossless import format
- Archiving mobile voicemail recordings from carrier systems that export in AMR to an uncompressed AIFF for long-term storage without further generational quality loss
- Feeding AMR voice recordings into audio forensics or transcription software that requires uncompressed PCM input and does not support the AMR codec natively
- Converting AMR speech samples collected in field research or linguistic studies into AIFF for analysis in tools like Praat or Adobe Audition on macOS
- Preparing voice-over scratch tracks originally recorded on a mobile device (saved as AMR) into AIFF so they can be synced and compared against studio recordings in a professional session
Frequently Asked Questions
No — the AIFF output will not recover audio quality lost during AMR encoding. AMR is a lossy codec designed for narrowband speech (typically 8 kHz sample rate for AMR-NB), meaning much of the audio spectrum above roughly 4 kHz was discarded when the original recording was made. The AIFF file is an exact, uncompressed representation of what the AMR decoder produces, which is already the highest fidelity you can get from that source. The benefit of AIFF here is lossless storage going forward: no further quality is lost during editing or re-saving.
AMR-NB operates at bitrates as low as 4.75 kbps up to 12.2 kbps, making it extremely compact. AIFF with 16-bit PCM at 8 kHz (typical for AMR-NB) produces approximately 128 kbps of uncompressed audio data — roughly 10 to 27 times larger than the AMR source. A 1-minute AMR file might be around 90 KB, while the equivalent AIFF could be 960 KB or more. This size increase is expected and is the trade-off for uncompressed, editor-friendly audio.
Yes. FFmpeg reads the sample rate embedded in the AMR file — 8,000 Hz for AMR-NB (narrowband) or 16,000 Hz for AMR-WB (wideband) — and preserves it in the AIFF output without resampling, unless you explicitly add a resampling flag like -ar to the command. If your downstream tool requires a specific sample rate such as 44,100 Hz or 48,000 Hz, you can append -ar 44100 to the FFmpeg command before the output filename.
Yes. The command uses -c:a pcm_s16be, which encodes 16-bit signed big-endian PCM into the AIFF container. You can substitute pcm_s24be for 24-bit, pcm_s32be for 32-bit integer, pcm_f32be for 32-bit float, or pcm_f64be for 64-bit float — all of which AIFF supports. For speech decoded from AMR-NB, 16-bit PCM is already well above the effective dynamic range of the source, so higher bit depths add file size without audible benefit unless your downstream workflow specifically requires them.
The single-file command shown can be adapted for batch processing in a shell. On macOS or Linux, you can run: for f in *.amr; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.amr}.aiff"; done — this loops over every AMR file in the current directory and converts each one to a matching AIFF file. On Windows Command Prompt, use: for %f in (*.amr) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". The browser-based tool processes one file at a time, so the FFmpeg command is especially useful for bulk conversions.
AMR files carry minimal metadata — the format was designed purely for telephony and generally stores little beyond basic codec parameters. AIFF supports standard metadata chunks (NAME, AUTH, ANNO), and FFmpeg will attempt to map any recognized metadata from the AMR source into the AIFF output. However, because AMR files rarely contain rich ID3-style tags, the AIFF output will typically also have minimal metadata. If you need embedded metadata in the output, you can add it manually using FFmpeg's -metadata flag, for example: -metadata title="Interview 2024".
Technical Notes
AMR-NB (narrowband), the default codec in most .amr files, operates at a fixed 8,000 Hz sample rate and mono channel — it was engineered for intelligible speech over mobile networks, not audio fidelity. AMR-WB (wideband, sometimes stored as .awb) uses 16,000 Hz and offers noticeably better clarity. FFmpeg uses the libopencore_amrnb and libopencore_amrwb libraries to decode these formats; these libraries must be present in the FFmpeg build, which is the case for this browser-based tool and most standard desktop FFmpeg distributions. The output codec pcm_s16be is the default and most compatible PCM format for AIFF — 'be' refers to big-endian byte order, which is native to the AIFF specification and required by macOS tools. AIFF does not support multiple audio tracks, subtitles, or chapter markers, but neither does AMR, so no data is lost on those fronts. One known limitation: if the AMR source is stereo (rare but possible with AMR-WB), FFmpeg will decode both channels correctly. Bit-for-bit, the decoded AIFF is the most accurate uncompressed representation of the AMR source achievable, making it a solid archival intermediate before any further processing.