Convert AC3 to AMR — Free Online Tool
Convert AC3 (Dolby Digital) surround sound audio to AMR format, transcoding from a multi-channel broadcast codec down to a speech-optimized mobile telephony format using the libopencore_amrnb encoder. This is ideal when you need to extract voice dialogue or speech content from AC3 audio tracks for use on mobile devices or telephony systems.
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FFmpeg Command
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DownloadHow It Works
AC3 is a lossy multi-channel audio format capable of encoding up to 5.1 surround sound at bitrates up to 640kbps, designed for high-fidelity home theater and broadcast use. AMR-NB (Adaptive Multi-Rate Narrowband), encoded here via libopencore_amrnb, is a fundamentally different codec designed exclusively for speech on mobile networks, operating at very low bitrates between 4.75kbps and 12.2kbps and supporting only mono audio at an 8kHz sample rate. During this conversion, FFmpeg decodes the AC3 stream, downmixes any surround channels to mono, resamples the audio from the AC3 source rate (typically 48kHz) down to 8kHz, and re-encodes the result using the AMR-NB codec. Because AMR-NB is a narrowband speech codec, the frequency range is heavily limited to the 300–3400Hz telephone band, meaning music or ambient sound effects from the original AC3 track will sound degraded — this conversion is most meaningful when the source material is primarily speech or dialogue.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles the full pipeline of demuxing the AC3 container, decoding the AC3 audio stream, resampling and downmixing, re-encoding with AMR-NB, and writing the output AMR file. |
-i input.ac3
|
Specifies the input file, an AC3 Dolby Digital audio file. FFmpeg reads and decodes the AC3 audio bitstream from this file as the source for the conversion. |
-c:a libopencore_amrnb
|
Selects the libopencore_amrnb encoder for the audio stream, which implements the AMR-NB (Adaptive Multi-Rate Narrowband) codec. This encoder enforces mono output at 8kHz, so FFmpeg will automatically downmix the AC3 surround channels and resample the audio to meet these requirements. |
-b:a 12200
|
Sets the AMR-NB encoding mode to 12.2kbps (MR122), the highest quality mode available in the AMR-NB codec standard. This value must be one of the eight fixed AMR-NB codec modes — 12200 represents the best speech quality the format is capable of producing. |
output.amr
|
Defines the output filename with the .amr extension, which tells FFmpeg to wrap the encoded AMR-NB audio in the standard RFC 4867 AMR file container, compatible with mobile devices and telephony applications. |
Common Use Cases
- Extracting spoken dialogue from a Dolby Digital DVD or broadcast AC3 track to create AMR voice memos or ringtones compatible with older mobile handsets
- Converting AC3 audio commentary tracks or podcast recordings that were originally mastered in Dolby Digital for broadcast delivery into AMR for distribution via MMS or legacy mobile messaging systems
- Preparing voice-over or narration audio recorded in AC3 from a broadcast workflow for integration into IVR (Interactive Voice Response) telephony systems that require AMR-NB format
- Stripping speech content from AC3-encoded news broadcasts or interviews to produce compact AMR files for archiving spoken-word content with minimal storage footprint
- Converting AC3 radio drama or spoken-word recordings to AMR for playback on embedded or IoT devices with limited audio processing capabilities that support only AMR decoding
Frequently Asked Questions
No. AMR-NB only supports mono audio, so all surround sound channels from the AC3 source — including center, surround, and LFE (subwoofer) channels in a 5.1 mix — will be downmixed to a single mono channel during conversion. FFmpeg handles this downmix automatically. If your AC3 file contains important audio spread across multiple channels, expect some content to be blended or attenuated in the resulting mono AMR file.
AC3 can encode audio with a full 20Hz–20kHz frequency range at bitrates up to 640kbps, while AMR-NB is a narrowband codec that only captures the 300–3400Hz speech frequency band at a maximum of 12.2kbps. The conversion also resamples the audio to 8kHz, which discards all high-frequency content. If your AC3 source contains music, sound effects, or ambient audio, those will sound noticeably thin, muffled, or distorted in the AMR output. AMR-NB is engineered specifically to make human speech intelligible, not to reproduce full-range audio.
AMR-NB uses a fixed set of codec modes rather than arbitrary bitrates, and 12200 (12.2kbps) is the highest mode, called MR122, which provides the best speech quality the codec is capable of. The other available modes range down to 4750 (4.75kbps). Unless you have a strict file size or bandwidth constraint — such as a telephony system that requires a specific AMR mode — keeping the default 12200 is recommended, as lower modes will make speech sound more robotic and compressed.
Replace the value after -b:a in the command with one of the valid AMR-NB mode bitrates: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200. For example, to use the 7950 mode you would write: ffmpeg -i input.ac3 -c:a libopencore_amrnb -b:a 7950 output.amr. Note that AMR-NB only accepts these specific values — providing an arbitrary bitrate like 64000 will cause an error or fall back to the nearest valid mode.
Yes. On Linux or macOS you can run a shell loop such as: for f in *.ac3; do ffmpeg -i "$f" -c:a libopencore_amrnb -b:a 12200 "${f%.ac3}.amr"; done. On Windows Command Prompt, use: for %f in (*.ac3) do ffmpeg -i "%f" -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". Keep in mind that your FFmpeg installation must be compiled with libopencore_amrnb support — many standard builds include it, but you can check by running ffmpeg -codecs | grep amr.
Very little metadata transfers during this conversion. AC3 files can contain metadata such as dialog normalization levels and channel configuration data, but AMR is a bare audio container with extremely limited metadata support. Standard tags like title or artist are generally not preserved in AMR files. If metadata retention is important, consider whether AMR is the right target format for your use case, as formats like M4A or OGG have much richer metadata support.
Technical Notes
This conversion involves significant lossy re-encoding at two levels: AC3 is already a lossy format, and AMR-NB is a second lossy encoding stage, meaning audio quality degrades through both compression steps. The most impactful technical constraint is AMR-NB's fixed 8kHz sample rate, which is enforced by the libopencore_amrnb encoder — FFmpeg will automatically insert a resampler in the filter chain to downsample from the AC3 source's typical 48kHz sample rate. The libopencore_amrnb library implements the 3GPP TS 26.071 AMR-NB standard and is required to be compiled into FFmpeg for this command to work; it is not included in all FFmpeg distributions due to licensing considerations. The output .amr file will be wrapped in the RFC 4867 AMR file format header, which is broadly compatible with mobile devices and telephony software. Note that if you want higher audio quality for speech than AMR-NB provides, you could alternatively target AMR-WB (libopencore_amrwb) which operates at 16kHz sample rate and bitrates up to 23.85kbps, though AMR-WB write support requires the vo-amrwbenc library rather than libopencore_amrwb. File sizes for AMR-NB output will be dramatically smaller than the AC3 source — a 640kbps AC3 file will shrink to roughly 1–2% of its original size at 12.2kbps AMR-NB.