Convert AMR to AC3 — Free Online Tool
Convert AMR audio files — the compressed speech format used in mobile voice recordings and telephony — to AC3 (Dolby Digital), the professional surround-sound codec used in DVDs, Blu-rays, and broadcast TV. This tool re-encodes the narrow-band AMR speech stream using the AC3 codec at 192k bitrate, making it compatible with Dolby Digital playback systems.
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FFmpeg Command
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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 (Adaptive Multi-Rate) audio is a highly compressed, speech-optimized codec designed for mobile telephony, typically sampled at 8kHz (narrowband AMR-NB) or 16kHz (wideband AMR-WB) with bitrates as low as 4.75 kbps. AC3 (Dolby Digital) is a perceptual audio codec supporting up to 5.1 surround channels at much higher bitrates (typically 192k–640k). During this conversion, FFmpeg decodes the AMR bitstream using the libopencore_amrnb or libopencore_amrwb decoder, then re-encodes the resulting PCM audio using the AC3 encoder. Because AMR source audio is mono or stereo speech sampled at low rates, the output AC3 file will reflect the original channel count and frequency content — it won't gain surround channels or high-frequency detail that wasn't in the source. The conversion is lossy at both ends, so some speech quality may shift as the AMR perceptual encoding is decoded and then re-compressed by the AC3 encoder.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. This command runs entirely in your browser via FFmpeg.wasm (WebAssembly) — no files leave your device. You can also run this same command locally on your desktop if you have FFmpeg installed, which is recommended for files over 1GB. |
-i input.amr
|
Specifies the input AMR file. FFmpeg detects the AMR container and selects the appropriate decoder (libopencore_amrnb for narrowband AMR or libopencore_amrwb for wideband AMR) based on the file's internal format header. |
-c:a ac3
|
Instructs FFmpeg to encode the audio stream using the AC3 (Dolby Digital) encoder. This replaces the AMR speech codec with the AC3 perceptual audio codec used in DVDs, Blu-rays, and broadcast television — a far more widely supported format for professional and home playback systems. |
-b:a 192k
|
Sets the AC3 audio bitrate to 192 kilobits per second, which is the standard baseline bitrate for stereo Dolby Digital audio. For a mono AMR speech source, 96k or 128k would also be sufficient; increase to 384k or higher only if you need compatibility with strict broadcast quality specifications. |
output.ac3
|
Defines the output filename with the .ac3 extension, telling FFmpeg to write a raw AC3 elementary stream. This format is directly accepted by DVD authoring tools and broadcast muxers. If you need the AC3 audio inside a video container (such as .mkv or .mp4), change the output filename extension accordingly. |
Common Use Cases
- Embedding a voice memo or field recording (originally captured on a mobile phone as AMR) into a DVD authoring project that requires Dolby Digital AC3 audio tracks
- Preparing mobile telephony recordings for broadcast or post-production workflows where the ingest system only accepts AC3-encoded audio
- Converting AMR voice recordings archived from older Nokia or feature phones into AC3 for inclusion in a video editing timeline that mandates Dolby Digital compatibility
- Transcoding AMR interview recordings into AC3 so they can be muxed into a VOB or transport stream for set-top box playback without format rejection errors
- Standardizing a batch of mixed-format audio files — some AMR, some other formats — into a single AC3 deliverable for a broadcast archive system
- Testing Dolby Digital playback pipelines using real speech content originally captured as AMR, without needing to source a separate AC3 test file
Frequently Asked Questions
No — converting from AMR to AC3 will not recover audio quality that was lost when the original AMR file was encoded. AMR is a lossy speech codec that discards frequency information above 4kHz (narrowband) or 7kHz (wideband), and those details cannot be restored by re-encoding to AC3. The AC3 output at 192k will faithfully represent whatever audio is in the AMR source, but the characteristic telephony sound of AMR — limited frequency range, occasional codec artifacts — will still be present in the output.
AMR is engineered for extreme compression of speech, operating at bitrates between 4.75 kbps and 12.2 kbps — far below typical audio codecs. AC3 at the default 192k bitrate is roughly 15 to 40 times the bitrate of a typical AMR file. This dramatic size increase is purely a result of the higher target bitrate, not any increase in audio quality or content. If file size is a concern, you can lower the AC3 bitrate in the FFmpeg command (e.g., to 96k), though 192k is the common baseline for Dolby Digital compatibility.
The output will match the channel layout of your AMR source. Most AMR files are mono (single channel), since AMR is a telephone speech codec. AC3 supports up to 5.1 surround channels, but FFmpeg will not fabricate surround channels from a mono or stereo source — the AC3 file will simply be mono or stereo AC3. If you need a 5.1 AC3 file for DVD authoring, you would need to upmix the audio as a separate step, which is beyond what this direct conversion does.
Replace the value after -b:a in the command. For example, to encode at 384k, use: ffmpeg -i input.amr -c:a ac3 -b:a 384k output.ac3. AC3 supports bitrates of 96k, 128k, 192k, 256k, 320k, 384k, 448k, and 640k. For a mono AMR speech source, 96k or 128k is usually more than sufficient. Higher bitrates like 384k or 640k are typically reserved for multichannel or music content and offer no meaningful improvement for speech converted from AMR.
Yes — on the command line you can use a shell loop to process multiple files. On Linux or macOS: for f in *.amr; do ffmpeg -i "$f" -c:a ac3 -b:a 192k "${f%.amr}.ac3"; done. On Windows Command Prompt: for %f in (*.amr) do ffmpeg -i "%f" -c:a ac3 -b:a 192k "%~nf.ac3". This browser-based tool processes one file at a time, so the FFmpeg command is especially useful for batch jobs involving many files.
AMR files generally carry minimal metadata — typically just the audio stream with no embedded timestamps, chapter markers, or subtitle tracks. AC3 also does not support chapters or subtitles, so there is no metadata loss in that sense. However, AMR's frame-based timing (each AMR frame is 20ms) is decoded to raw PCM before being re-encoded by the AC3 encoder, so the overall duration should be preserved accurately. Sync issues are more likely to arise when the resulting AC3 is muxed into a video container — ensure your muxer correctly handles the AC3 frame boundaries.
Technical Notes
AMR-NB (narrowband) encodes speech at 8kHz sample rate with one of eight fixed bitrates from 4.75 to 12.2 kbps; AMR-WB (wideband) operates at 16kHz. FFmpeg uses the libopencore_amrnb and libopencore_amrwb libraries to decode these streams. The decoded PCM is then passed to the native FFmpeg AC3 encoder, which implements the ATSC A/52 (Dolby Digital) standard. One practical limitation: AC3 requires a sample rate of 32kHz, 44.1kHz, or 48kHz, so FFmpeg will automatically resample the AMR source (8kHz or 16kHz) to 48kHz during conversion — this resampling is transparent but means the output sample rate will differ from the input. The AC3 format does not support embedded metadata fields like artist or album tags, so any ID3-style tags present in some AMR container variants will not carry over. AC3 is supported natively by virtually all DVD players, Blu-ray players, AV receivers with Dolby Digital decoding, and most video editing applications, making it a reliable delivery format even when the source content is narrow-band speech.