What is AAC? Advanced Audio Coding for Streaming Explained

Advanced Audio Coding (AAC) is an MPEG-standardized lossy perceptual codec that uses psychoacoustic masking to remove or reduce audio details listeners are less likely to notice, generally providing better quality than MP3 at comparable bitrates. AAC is a codec rather than a container, appearing as raw ADTS-framed audio in .aac files or within MP4-based .m4a and .mp4 files, and it is identified in streaming manifests through codec strings such as mp4a.40.2 for AAC-LC. AAC-LC is the most compatible profile and is recommended for HLS and RTMP delivery at roughly 96–160 kbps stereo, while HE-AAC variants use spectral band replication and parametric stereo for efficient low-bitrate audio at 64 kbps or below; AAC-LD and AAC-ELD target lower-delay communication use cases. HLS and RTMP commonly carry AAC, with Apple HLS requiring stereo AAC-family audio, whereas WebRTC requires Opus and does not require AAC, meaning platforms that bridge interactive WebRTC rooms to HLS broadcasts typically re-encode Opus to AAC once. FFmpeg’s native AAC encoder supports standard AAC-LC-oriented profiles, while the optional Fraunhofer FDK-based encoder adds HE-AAC and low-delay profiles, subject to build and licensing considerations.

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