Audio encoding:Low-delay AAC

Low-delay AAC

MPEG-4 audio toolbox contains a number of other techniques that may be used for other applications. One of these tools is low-delay AAC (AAC-LD).

While the MPEG-4 AAC provides very efficient coding of general audio signals at low bit rates, it has an algorithmic encoding/decoding delay of up to several 100 ms which is tolerable for broadcasting applications. As an example, for the general audio coder operating at 24 kHz sampling rate and 24 kbps, this results in an algorithmic coding delay of about 110 ms plus up to additional 210 ms for the use of the bit reservoir. Such long delays are unacceptable for such applications as real-time bidirectional communication. To enable coding of general audio signals with a delay not exceeding 20 ms, MPEG-4 specifies a low-delay audio coder. Compared with speech coding schemes, this coder allows compression of general audio signal types, including music, at a low delay. It operates at up to 48 kHz sampling rate and uses a frame length of 512 or 480 samples, com- pared with 1024 or 960 samples used in standard MPEG-2/4 AAC. Also, the size of the window used in the analysis and synthesis filter bank is reduced by a factor of 2. No block switching is used to avoid the ‘look- ahead’ delay due to the block switching decision. To reduce pre-echo arte- facts in case of transient signals, window shape switching is provided instead. For non-transient parts of the signal a sine window is used, while the low overlap KBD window is used in case of transient signals. Use of the bit reservoir is minimised in the encoder in order to reach the desired tar- get delay. As one extreme case, no bit reservoir is used at all. Verification tests have shown that the reduction in coding delay comes at a very mod- erate cost in compression performance.

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