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AAC Low Delay The MPEG-4 Low Delay Audio Coder (AAC-LD) is designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) format.

The most stringent requirements are a maximum algorithmic delay of only 20 ms and a good audio quality for all kind of audio signals including speech and music.

• The AAC-LD coding scheme bridges the gap between speech coding schemes and high quality audio coding schemes.

AAC Low Delay compared to normal AAC codecs and ITU speech audio compression systems.

Two-way communication with AAC-LD is possible on usual analog telephone lines and via ISDN connections. Compared to known speech coders, the codec is capable of coding both music and speech signals with good quality. Unlike speech coders, however, the achieved coding quality scales up with bitrate. Transparent quality can be achieved.

AAC LD can also process stereo signals by using the advanced stereo coding tools of AAC. Thus it is possible to transmit a stereo signal with a bandwidth of 7 kHz via one ISDN line or with a bandwidth of 15 kHz via two ISDN lines.

Applying error protection enables error correction up to a certain extent. Error correcting codes are usually applied equally to the whole payload.

But since different parts of an AAC payload show different sensitivity to transmission errors, this would not be a very efficient approach.

The AAC payload can be subdivided into parts with different error sensitivities. Independent error correcting codes can be applied to any of these parts using the Error Protection (EP) tool defined in MPEG-4 Audio. This provides the error correcting capability just the most sensitive parts of the payload in order to keep the additional overhead low.

Error Resilience (ER) techniques can be used in all versions of AAC CODECs to make the coding scheme itself more robust against errors.

For AAC, three custom-tailored methods were developed and defined in MPEG-4 Audio

• Huffman Codeword Reordering (HCR) to avoid error propagation within spectral data;
• Virtual Codebooks (VCB11) to detect serious errors within spectral data;
• Reversible Variable Length Code (RVLC) to reduce error propagation within scale factor data.

• Advanced Audio Coding (parent technology)
• MP3 (related older technology)
• Musicam (related older technology)