Advanced Audio Coding

masonchung

Advanced Audio Coding (AAC) is a standardized, lossy digital audio compression scheme. It's a way of compressing audio files, such as WAV, AIFF or imported from a CD. AAC gets a better quality sound than MP3 for the same compression level or smaller file sizes for the same quality audio. For example: A 192 kbps song compressed by MP3 will be the same quality as a 128 kbps AAC file, which will be smaller in overall size. It's most commonly used as the standard format for compressing audio CD's for Apple's iPod and iTunes. Apple also uses this as their format for selling from the iTunes Store, although these files are restricted using Apple's own DRM, known as FairPlay. It was also used as the standard audio file, for SONY's Playstation 2. It's also used as the audio for the .M4V format that Apple uses for it's iPod with Video.

History
AAC 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 1997.

Standardization
It is specified both as Part 7 of the MPEG-2 standard, and Part 3 of the MPEG-4 standard.
) Y9 c# s3 ]" `6 d; n5 [
% L( g. w1 i2 ^/ ]" PAs such, it can be referred to as MPEG-2 Part 7 and MPEG-4 Part 3 depending on its implementation, however it is most often referred to as MPEG-4 AAC, or AAC for short.
, E% K% l+ t0 U
AAC's improvements over MP3
AAC was designed to have better performance than MP3 (which was specified in MPEG-1 and MPEG-2) by the ISO/IEC in 11172-3 and 13818-3.
9 W# f7 M3 A5 A2 n/ J  [+ x5 l
Improvements include:
) ]" \+ }$ {9 a. K" x4 y6 k) k
More Sample frequencies (from 8 kHz to 96 kHz) than MP3 (16 kHz to 48 kHz)
Up to 48 channels (MP3 supports up to two channels in MPEG-1 mode and up to 5.1 channels in MPEG-2 mode)
$ p+ u& b4 M. q' @/ {( X% r1 z! fHigher efficiency and simpler filterbank (hybrid → pure MDCT)
% ]9 P7 q& K, k; m( M. ^9 PHigher 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
. ]! W$ j3 q" J" ^1 ]4 o3 W( a# KMore flexible joint stereo (separate for every scale band)
This gives developers more flexibility to design codecs that offer more efficient compression as compared to MP3. However in terms of whether AAC is better than MP3, the advantages of AAC are not entirely conclusive, and the MP3 specification, while outdated, has proven surprisingly robust. AAC and HE-AAC are better than MP3 at very low bitrates, however at medium to higher bitrates, the two formats are more comparable.
# Q' v, O2 U, E' G- Z
Marketing aspects
& L2 w: _; x  l, O8 u8 kAAC 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 Apple 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).
# Q7 H% X; K( u9 H4 z
How AAC works
  n8 a4 v7 E; L% D; L) c8 NAAC 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.

Signal components that are perceptually irrelevant are discarded;
Redundancies in the coded audio signal are eliminated;
The signal is processed by a modified discrete cosine transform (MDCT) according to its complexity;
1 `4 h6 I# G% BInternal error correction codes are added;
; k7 ]& v; [, v3 Q' d7 E8 GThe 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.

" \" Z: o) [5 YThe 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.
# n7 `" T$ g# Z4 ?5 j2 SIn contrast to MP3's hybrid filter bank, AAC uses the modified discrete cosine transform (MDCT) together with the increased window lengths of 1024 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 1024 points or 8 blocks of 128 points.

If a signal change or a transient occurs, 8 shorter windows of 128 points each are chosen for their better temporal resolution.
By default, the longer 1024-point window is otherwise used because the increased frequency resolution allows for a more sophisticated psychoacoustic model, resulting in improved coding efficiency.
# q. ?7 m1 }; I1 d2 K! d3 k; S
" Y* i2 k2 Q8 F' e. \ Modular encoding
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:
. V3 z. t4 @" h1 U
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);
3 i0 \! h- }; i- V+ jLong 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.[1]
% }' \' b( j3 R
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.
- A. M& m: P9 v) Q+ g9 X4 C* d0 }1 ^
0 _3 {/ y4 c0 k, G- p, P, oThe 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.

/ `3 q3 Y3 k2 N' b! qAAC Low Delay compared to normal AAC codecs and ITU speech audio compression systems.
# b" i6 p0 z" x+ L# G0 W& V
AAC 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.
" Z( S; x% b7 g1 V9 _3 l
  [! T: a$ G/ U- o: @0 G2 wThe AAC payload can be subdivided into parts with different error sensitivities.
) s( \+ |* o9 f! V  Y" P1 B+ w
Independent error correcting codes can be applied to any of these parts using the Error Protection (EP) tool defined in MPEG-4 Audio.
This toolkit provides the error correcting capability to the most sensitive parts of the payload in order to keep the additional overhead low.
$ _- q# J# D9 n( o$ L& o
Error Resilient (ER) AAC
$ q- ^9 Q  d. Z3 c% A- s6 y5 K  }% E+ Y5 _Error Resilience (ER) techniques can be used to make the coding scheme itself more robust against errors.
+ a1 ~. a* R4 d3 d. \
For AAC, three custom-tailored methods were developed and defined in MPEG-4 Audio

3 |  r; a% U$ f* `: E7 E' D0 sHuffman Codeword Reordering (HCR) to avoid error propagation within spectral data;
! P. I( t" J, K5 dVirtual Codebooks (VCB11) to detect serious errors within spectral data;
( J% i: J9 a: I& ?Reversible Variable Length Code (RVLC) to reduce error propagation within scale factor data.

  T0 `6 Y' F* x' MAAC 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.
# {- L: E; g4 w' I, x
[ 本帖最後由 masonchung 於 2007-1-20 02:43 PM 編輯 ]