AAC Definition

ACC is short for Advanced Audio Coding, AAC is an audio codec that has better audio quality and compression when compared with MP3.

Advanced Audio Coding (AAC) is a standardised, lossy digital audio compression scheme. It was developed with the cooperation and contributions of companies mainly including Dolby, Fraunhofer (FhG), AT&T, Sony and Nokia, and was officially declared an international standard by the Moving Pictures Experts Group in April of 1997. It was written into specification as Part 7 of the MPEG-2 standard, and again into Part 3 of the MPEG-4 standard. As such, AAC can be referred to as MPEG-2 Part 7 and MPEG-4 Part 3 depending on its implementation, but is most often referred to as MPEG-4 AAC, or AAC for short. AAC was designed as an improved-performance codec relative to MP3 (which was specified in MPEG-1 and MPEG-2) by the ISO/IEC in 11172-3 and 13818-3.

AAC was promoted as the successor to MP3 for audio coding at medium to high bitrates. Its popularity is currently maintained by it being the default iTunes codec, the media player which powers iPod, the most popular digital audio player on the market. [1] Furthermore, the iTunes Store, whose sales account for 85% of the market for legal online downloads, [2] sells AAC-encoded songs (encapsulated with FairPlay Digital Rights Management).

How ACC Works

AAC is a wideband audio coding algorithm that exploits two primary coding strategies to dramatically reduce the amount of data needed to represent high-quality digital audio.

1. Signal components that are perceptually irrelevant are discarded;
2. Redundancies in the coded audio signal are eliminated;
3. The signal is processed by a modified discrete cosine transform (MDCT) according to its complexity;
4. Internal error correction codes are added;
5. The signal is stored or transmitted.

The MPEG-4 audio standard does not define a single or small set of highly efficient compression schemes but rather a complex toolbox to perform a wide range of operations from low bitrate speech coding to high-quality audio coding and music synthesis.

• The MPEG-4 audio coding algorithm family spans the range from low bitrate speech encoding (down to 2 Kbit/s) to high-quality audio coding (at 64 Kbit/s per channel and higher).
• AAC offers sampling frequencies between 8 kHz and 96 kHz and any number of channels between 1 and 48.
• In contrast to MP3's hybrid filter bank, AAC uses the modified discrete cosine transform (MDCT) together with the increased window lengths of 2048 points. AAC is much more capable of encoding audio with streams of complex pulses and square waves than MP3 or MP2.

AAC encoders can switch dynamically between a single MDCT block of length 2048 points or 8 blocks of 256 points.

• If a single change or transient occurs, the short window of 256 points is chosen for better temporal resolution.
• By default, the longer 2048-point window is used to improve the coding efficiency because of better frequency resolution.

Know matter if you need to know "what is a ACC", the definition of a "ACC", or the meaning of a "ACC", you can find it here at Network Liquidators. There's quite a bit of information out there to learn, and it all starts by you having the initiative to seek out that information.

Modular Coding

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. The standard offers four default profiles:

• Low Complexity (LC) - the simplest and most widely used and supported;
• Main Profile (MAIN) - like the LC profile, with the addition of backwards prediction;
• Sample-Rate Scalable (SRS), a.k.a. Scalable Sample Rate (MPEG-4 AAC-SSR);
• Long Term Prediction (LTP); added in the MPEG-4 standard - an improvement of the MAIN profile using a forward predictor with lower computational complexity.

Depending on the AAC profile and the MP3 encoder, 96 kbit/s AAC can give nearly the same or better perceptional quality as 128 kbit/s MP3.

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. In this way, the AAC-LD coding scheme bridges the gap between speech coding schemes and high quality audio coding schemes.

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.

Error Protection Toolkit

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 Resilient (ER) AAC
Error Resilience (ER) techniques can be used 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.

AAC's improvements over MP3

Some of its advances:

• Sample frequencies from 8 kHz to 96 kHz (official MP3: 16 kHz to 48 kHz)
• Up to 48 channels
• Higher efficiency and simpler filterbank (hybrid → pure MDCT)
• Higher coding efficiency for stationary signals (blocksize: 576 → 1024 samples)
• Higher coding efficiency for transient signals (blocksize: 192 → 128 samples)
• Can use Kaiser-Bessel derived window function to eliminate spectral leakage at the expense of widening the main lobe
• Much better handling of frequencies above 16 kHz
• More flexible joint stereo (separate for every scale band)

The result is a specification that allows developers more flexibility to design codecs that offer efficient compression compared to MP3. However, the advantages are not entirely decisive, and the MP3 specification, while outdated, has proven surprisingly robust. Although AAC and HE-AAC are far better than MP3 at very low bitrates, at medium to higher bitrates the two formats are more comparable. In the future as developers learn to better exploit the AAC format, AAC is expected to gain additional ground and perhaps overtake MP3.

AAC ISO Standard

AAC, which was first specified in the standard known formally as ISO/IEC 13818-7, was published in 1997 as a new "part" (distinct from ISO/IEC 13818-3) in the MPEG-2 family of international standards.

Know matter if you need to know "what is a AAC", the definition of a "AAC", or the meaning of a "AAC", you can find it here at Network Liquidators. There's quite a bit of information out there to learn, and it all starts by you having the initiative to seek out that information.

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