Convert ALAC to AMR — Free Online Tool
Convert ALAC audio files (.m4a) to AMR format using the libopencore_amrnb codec, optimized for speech encoding at low bitrates. This conversion is ideal for reducing lossless Apple audio to compact, mobile-telephony-compatible voice files.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
ALAC (Apple Lossless Audio Codec) stores audio in an MPEG-4 container with full lossless fidelity, meaning every sample from the original recording is preserved. During conversion to AMR, the ALAC audio stream is fully decoded from its lossless representation and then re-encoded using the libopencore_amrnb (Adaptive Multi-Rate Narrowband) codec. AMR-NB operates at fixed bitrates between 4.75 kbps and 12.2 kbps and is designed specifically for speech — it models voice frequencies (roughly 300–3400 Hz) and discards audio information outside that range. This means music, wide-frequency-range audio, and stereo content in your ALAC file will be significantly degraded or downmixed to mono, as AMR-NB only supports mono audio at 8000 Hz sample rate. The output .amr file will be dramatically smaller than the source .m4a, but quality loss is substantial and irreversible.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In the browser version of this tool, this runs via FFmpeg.wasm (WebAssembly) entirely client-side — no data leaves your machine. |
-i input.m4a
|
Specifies the input file — an MPEG-4 container (.m4a) holding a lossless ALAC audio stream. FFmpeg reads the ALAC codec data and container metadata from this file. |
-c:a libopencore_amrnb
|
Sets the audio encoder to libopencore_amrnb, which implements the AMR Narrowband codec used in GSM mobile telephony. This re-encodes the decoded ALAC audio into AMR-NB format, enforcing mono output at 8000 Hz. |
-b:a 12200
|
Sets the AMR-NB audio bitrate to 12,200 bits per second (12.2 kbps), which is the highest quality mode defined by the AMR-NB standard. This corresponds to AMR mode 7 (MR122) and produces the most intelligible speech output within the format's constraints. |
output.amr
|
Defines the output filename with the .amr extension. FFmpeg uses this extension to select the AMR file container, which holds the libopencore_amrnb-encoded audio stream in a format natively supported by mobile phones and telephony systems. |
Common Use Cases
- Preparing voice memos or spoken-word recordings originally saved in Apple's ALAC format for use in GSM or VoIP telephony systems that require AMR-encoded audio
- Archiving interview recordings from an Apple ecosystem (iPhone, Logic Pro) into AMR for storage on legacy mobile devices or embedded systems with limited storage
- Converting ALAC-encoded audiobook narration or podcast dialogue tracks into AMR for playback on feature phones or older Android devices that natively support AMR
- Generating AMR audio clips from ALAC source files for integration into IVR (Interactive Voice Response) systems that require AMR-NB formatted prompts
- Compressing losslessly recorded voice-over sessions from a DAW that exported ALAC, down to a minimal-size AMR file for MMS messaging or low-bandwidth delivery
- Extracting and re-encoding a spoken commentary track from an ALAC audio file into AMR format for compatibility with automotive or industrial embedded audio playback systems
Frequently Asked Questions
Almost certainly not, if music is the primary content. AMR-NB is a speech codec engineered to encode the human voice at 8000 Hz mono, discarding everything outside a narrow telephone-quality frequency band. Music with instruments, stereo imaging, or wide dynamic range will sound severely degraded — muffled, lossy, and mono. AMR is the right choice only when your ALAC file contains speech or voice content and file size is the overriding concern.
AMR-NB (libopencore_amrnb) is inherently a mono codec — it was designed for GSM mobile telephony where a single audio channel is sufficient for voice calls. FFmpeg will automatically downmix your stereo ALAC channels to mono during the conversion. If your source has distinct left/right content, that spatial information is permanently lost in the output .amr file.
The AMR container format does not support metadata tags, iTunes-style chapter markers, or embedded artwork. Any ID3-style tags, album art, or chapter data present in your ALAC .m4a file will be stripped during conversion and will not appear in the output .amr file. If preserving metadata is important, keep the original .m4a file as your archival copy.
Replace the value after -b:a in the command with any of the fixed AMR-NB bitrates: 4750, 5150, 5900, 6700, 7400, 7950, or 10200 (in bits per second). For example, use -b:a 4750 for the smallest possible file at the cost of lower speech intelligibility, or -b:a 7950 for a balance of size and voice clarity. The default in this tool is 12200 (12.2 kbps), which is the highest quality AMR-NB mode.
Yes — AMR-WB uses the libopencore_amrwb codec and operates at 16000 Hz sample rate instead of 8000 Hz, giving noticeably better speech quality. To use it, change the codec flag in the command to -c:a libopencore_amrwb and set a compatible bitrate such as -b:a 23850. Note that AMR-WB (.awb) has less universal support than AMR-NB on legacy devices, so confirm your target platform supports it before switching.
Yes. On Linux or macOS, you can run a shell loop: for f in *.m4a; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.m4a}.amr"; done. On Windows Command Prompt, use: for %f in (*.m4a) do ffmpeg -i "%f" -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". This is especially useful for large batches since the browser-based tool processes one file at a time.
Technical Notes
ALAC files carry 16-bit or 24-bit lossless audio at sample rates up to 384 kHz, stored in an MPEG-4 (.m4a) container. AMR-NB, by contrast, enforces a fixed 8000 Hz sample rate and mono channel layout, with only eight possible bitrates defined by the GSM/3GPP standard. FFmpeg handles the sample rate conversion (downsampling from whatever rate your ALAC uses to 8000 Hz) and stereo-to-mono downmix automatically as part of this conversion. Because AMR uses a Codec Excited Linear Prediction (CELP) model tuned for speech formants, non-speech audio suffers severe perceptual degradation. The resulting .amr file size will be roughly 90–98% smaller than the source ALAC file — a 10 MB .m4a might produce an .amr under 200 KB. There is no way to recover the original lossless quality from an AMR file; always retain your ALAC source. Note also that FFmpeg must be compiled with libopencore-amrnb support — most standard FFmpeg builds include this, but self-compiled builds may omit it if the library was not present at compile time.