Free online ADTS to WAV converter

What the ADTS format is

ADTS, or Audio Data Transport Stream, is a framing format for AAC audio. An ADTS file stores a sequence of independently framed AAC packets, with headers carrying values such as the profile, sample rate, channel configuration, and frame length.

ADTS is used for raw AAC audio delivered outside an MP4 or other container. You may encounter it as .aac, occasionally as .adts, in broadcast and streaming workflows, audio extracted from transport streams, encoder output, and recordings made by devices that write AAC frames directly. FFmpeg, FAAC, and similar encoders can create ADTS output.

Unlike an MP4/M4A file, an ADTS stream normally has little or no container-level metadata such as a title, album, cover image, or chapter list. The filename extension alone does not guarantee that the content is valid ADTS AAC; some files use .aac for other AAC layouts.

What the WAV format is

WAV is an audio file format based on the RIFF container. It commonly stores uncompressed linear PCM samples, although the container can hold other codecs. The usual conversion target is standard PCM WAV, such as 16-bit or 24-bit little-endian audio.

PCM WAV is suitable for editing, mixing, measurement, speech-recognition input, game-audio tools, CD preparation, and applications that do not decode AAC. It preserves the decoded sample rate and channel count without adding another lossy compression stage, provides direct sample access, and is supported by most desktop audio software.

WAV files are much larger than ADTS files. For example, stereo 16-bit PCM at 44,100 Hz requires about 1,411 kbps before WAV headers, compared with a typical AAC stream using a fraction of that bitrate.

How to actually convert ADTS to WAV

Using FFmpeg

FFmpeg is the most reliable desktop command-line option for this conversion and runs on Windows, macOS, and Linux.

1. Install FFmpeg from your operating system’s package manager or the official FFmpeg distribution.
2. Open a terminal or command prompt in the folder containing the source file.
3. Run ffmpeg -i input.adts -c:a pcm_s16le output.wav. Replace input.adts with the actual filename; an AAC file with an .aac extension works the same way.
4. Check that FFmpeg reports the expected sample rate, channel count, and AAC profile, then open output.wav in the target application.

To retain the decoded stream’s sample rate and channels while selecting 24-bit PCM, use ffmpeg -i input.aac -c:a pcm_s24le output.wav. To set a specific output format, add options such as -ar 48000 for 48 kHz or -ac 2 for stereo: ffmpeg -i input.aac -ar 48000 -ac 2 -c:a pcm_s16le output.wav.

Use ffprobe input.aac before conversion when the file may be malformed or mislabeled. The output identifies the detected codec, sample rate, channel layout, duration, and stream errors.

Using an online converter

You can convert the file for free on 101convert.com without installing software: select the ADTS/AAC file, choose WAV as the output format, start the conversion, and download the resulting file. Do not upload recordings containing confidential, personal, or legally restricted audio.

Using a graphical audio editor

Audacity can open AAC files when its FFmpeg import support is installed. Import the file through File → Import → Audio, then choose File → Export Audio, select WAV, and choose a PCM encoding such as 16-bit or 24-bit. The available import behavior depends on the Audacity version and installed FFmpeg library.

Quality, compatibility, and limitations

Decoding AAC into WAV does not make the audio lossless. The AAC encoder has already discarded information; WAV only prevents additional loss during later editing or export. Avoid converting the WAV back to AAC repeatedly because each lossy encode can introduce more artifacts.

Keep the source sample rate and channel layout unless the destination specifically requires another format. Resampling or downmixing with -ar and -ac changes the audio data and may remove channels. A WAV file can also exceed the 4 GB limit of classic RIFF WAV; very long, high-resolution files may require RF64 support.

Some ADTS streams use HE-AAC, unusual profiles, damaged frame headers, or incomplete data. FFmpeg may decode them successfully, report errors while continuing, or reject them. If the output duration is wrong or contains gaps, obtain an intact source and inspect it with ffprobe; changing the extension does not repair an invalid stream.