Free online ADTS to OPUS converter

Some players, browsers, and communication applications accept Opus but cannot open AAC audio carried in ADTS frames. Converting the stream produces an Ogg Opus file that is better suited to web delivery, voice communication, and smaller audio archives.

What the ADTS format is

ADTS (Audio Data Transport Stream) is a framing format for AAC audio, not a separate audio codec. Each AAC frame has an ADTS header containing synchronization data, codec information, sample rate, channel count, and frame length.

ADTS audio commonly uses the .aac extension and appears after AAC is extracted from an MPEG transport stream or generated for broadcast and streaming workflows. FFmpeg, GStreamer, FAAC, hardware encoders, and similar tools can create it. Some files use .adts, but the extension does not change the underlying AAC content.

What the OPUS format is

Opus is a modern lossy audio codec designed for both speech and music. It supports low-delay communication, variable bit rates, mono and multichannel audio, and sample rates up to 48 kHz.

Files ending in .opus normally contain Opus packets in an Ogg container. Opus usually provides smaller files than AAC at comparable perceived quality, especially for speech and interactive audio, but it is not supported by every older hardware player or editing application.

How to actually convert ADTS to OPUS

FFmpeg is the most direct desktop tool because it reads ADTS AAC and writes an Ogg Opus file without requiring an intermediate format.

1. Install a current FFmpeg build that includes the libopus encoder.
2. Check the source stream if necessary with ffprobe input.aac. Confirm the sample rate, channel layout, and duration.
3. Run ffmpeg -i input.aac -c:a libopus -b:a 128k output.opus. Replace input.aac with the actual ADTS filename; FFmpeg can usually identify ADTS content even when the extension is .adts.
4. Verify the result with ffprobe output.opus or by opening it in an Opus-capable player.

For speech, use a lower bit rate such as -b:a 32k to -b:a 64k. For music, -b:a 96k to -b:a 160k is a common range. The command ffmpeg -i input.aac -c:a libopus -b:a 128k -vbr on output.opus explicitly enables variable-bitrate encoding.

To convert without installing software, you can use the free converter on 101convert.com. Upload the ADTS/AAC file, select OPUS as the output format, start the conversion, and download the resulting file.

Quality, compatibility, and limitations

This is a lossy-to-lossy transcode: AAC is decoded and then encoded again as Opus. The conversion cannot restore detail already removed by AAC, so keep the original ADTS file for archival use and avoid repeated re-encoding.

FFmpeg normally writes an Ogg Opus container when the output filename ends in .opus. If a target application specifically requires WebM rather than Ogg, use ffmpeg -i input.aac -c:a libopus -b:a 128k output.webm. Some older applications recognize only .ogg or do not support Opus at all; check the target application's supported containers and codecs before converting.

Opus output is generally standardized at a 48 kHz internal sampling rate, even when the source AAC uses another rate. This is normal and does not indicate a failed conversion.