AAC - Advanced Audio

Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for digital audio. Designed to be the successor of the MP3 format, AAC generally achieves better sound quality than MP3 at similar bit rates AAC has been standardized by ISO and IEC, as part of the MPEG-2 and MPEG-4 specifications. Part of the AAC known as High-Efficiency Advanced Audio Coding (HE-AAC) which is part of MPEG-4 Audio is also adopted into digital radio standards like DAB+ and Digital Radio Mondiale, as well as mobile television standards DVB-H and ATSC-M/H.AAC supports inclusion of 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low frequency effects (LFE, limited to 120 Hz) channels, up to 16 "coupling" or dialog channels, and up to 16 data streams. The quality for stereo is satisfactory to modest requirements at 96 kbit/s in joint stereo mode; however, hi-fi transparency demands data rates of at least 128 kbit/s (VBR). The MPEG-2 audio tests showed that AAC meets the requirements referred to as "transparent" for the ITU at 128 kbit/s for stereo, and 320 kbit/s for 5.1 audio. AAC is also the default or standard audio format for iPhone, iPod, iPad, Nintendo DSi, iTunes, DivX Plus Web Player and PlayStation 3. It is supported on PlayStation Portable, Wii (with the Photo Channel 1.1 update installed for Wii consoles purchased before late 2007), Sony Walkman MP3 series and later, mobile phones made by Sony Ericsson; Nokia; Android; and WebOS - based mobile phones. AAC has also seen moderate adoption on in-dash car audio especially on high-end units such as the Pioneer AVIC series.

AAC stands for advanced audio coding and is the logical successor to MP3 (ISO/MPEG Audio Layer-3) for audio coding at medium to high bit rates.The idea of AAC's algorythm is to exploit two primary coding strategies to dramatically reduce the amount of data needed to convey high-quality digital audio. AAC has the sampling frequencies between 8 Hz and 96 kHz and any number of channels between 1 and 48. AAC uses the modified discrete cosine transform (MDCT) together with the increased window lengths of 2,048 points. AAC is much more capable of encoding audio with streams of complex pulses and square waves than MP3 or Musicam. AAC can be switched dynamically between MDCT block lengths of 2,048 points to 256 points.

AAC takes a modular approach to encoding. Depending on the complexity of the bitstream to be encoded, the desired performance and the acceptable output, implementers may create profiles to define which of a specific set of tools they want use for a particular application. So the AAC LD coding scheme bridges the gap between speech coding schemes and high quality audio coding schemes. Low Delay AAC (AAC-LD) and its successor, Enhanced Low Delay AAC (AAC-ELD), are state-of-the-art MPEG-4 audio codecs for maximum speech and audio quality at very low coding delay. The codecs are designed for high-quality IP-communication applications and devices.

AAC-LD is derived from MPEG AAC-LC, while AAC-ELD is a combination of MPEG AAC-LD and Spectral Band Replication (SBR).Quality    AAC-LD: CD-like audio quality at full audio bandwidth and superior to ITU-T G.722/G.722.1-C, G.729.1 at identical bit-rates AAC-ELD: CD-like audio quality at full audio bandwidth Bitrates   
AAC-LD: Large range of bit-rates (32 kbit/s - 64 kbit/s and more)
AAC-ELD: optimized for bit-rates from 24 to 64 kbit/s and more

Algorithmic delay    AAC-LD: down to 20 ms
AAC-ELD: down to 15 ms at 64 kbit/s, and up to 32 ms at 24 kbit/s
Channels    mono, stereo, multi-channel
Used in    e.g. Cisco TelePresence, Telos Zephyr/IP, Comrex Access,

Codian MCUs, Texas Instruments VoIP HD Voice Platform, Digigram Special feature    Suitable for all kinds of audio signals, including speech, music and ambience