MPEG-2/4 AAC-LC Encoder and Decoder

Low Complexity Advanced Audio Coder C64x

The AAC-LC (Advanced Audio Coding) is a natural audio coding algorithm that can handle 48 channels and sample rates up to 96kHz. This coding technique uses a perceptual filter bank, a sophisticated masking model, noise-shaping techniques, and channel coupling. It provides the highest possible quality at smaller bit-rates. AAC is an audio data compression format. The encoder/decoder is an implementation of the MPEG-2 (ISO/IEC 13818-3) and MPEG-4 (ISO/IEC 14496-3) standard. AAC provides both a higher quality and superior performance than MP3 at the same bit rate. It supports the coding of multi-channel audio, with up to 48 main channels and 16 low-frequency channels. AAC offers multiple profiles/object types to meet the requirements of a wide range of applications making it one of the most popular audio compression standards across wide spectrum of application ranging from portable player, cell phones, music systems, to the internet.

FEATURES

Encoding and decoding of AAC-LC (Low Complexity) bit-streams
MPEG4 AAC Low Complexity (LC) object type implementations supported
MPEG2 AAC Low Complexity (LC) object type implementations supported
Decoding of mono and stereo streams supported
Mono, stereo, and dual mono input file supported
RAW data input format supported
Sampling frequency range of 8 khz – 96 khz (input sampling frequencies supported)
Bit Rate : 16 - 576 kbps
C callable API for initialization, decoding
Full accuracy with ISO/IEC 13818 and ISO/IEC 14496 audio standards
Multi-channel, Reentrant implementation
Tested on a variety of AAC bit streams
16/24 bit PCM output / High sound quality
Optimized implementation
Only AAC-LC output format supported
eXpressDSP™ Digital Media compliant

AVAILABILITY

Currently supported on ARM (ARM9E™, ARM11™), TI DM320™, TI C64x™ (TMS320C64x™), TI C64x+™/DaVinci™ (DM644x™)

APPLICATIONS

Digital Audio Broadcasting
Portable Audio Systems
Internet Streaming
Internet Appliances
Cell Phones