MPEG-4 Part 3

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MPEG-4 Part 3 (formally ISO/IEC 14496-3) is the third part of the ISO/IEC MPEG-4 international standard. It specifies audio coding methods.

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[edit] Bifurcation in the AAC technical standard

The Advanced Audio Coding in MPEG-4 Part 3 was enhanced relative to what was previously specified in MPEG-2 Part 7, in order to provide better sound quality relative to the bit rate used for the encoding.

It is assumed that any Part 3 and Part 7 differences will be ironed out by the ISO standards body in the near future to avoid the possiblity of future bitstream incompatabilites. At present there are no known player or codec incompatibilities due to the newness of the standard.

AAC's multiple codecs:

  • Low Complexity Advanced Audio Coding (LC-AAC)
  • High-Efficiency Advanced Audio Coding (HE-AAC)
  • Scalable Sample Rate Advanced Audio Coding (AAC-SSR)
  • Bit Sliced Arithmetic Coding (BSAC)

[edit] HE-AAC

It has been suggested that this section be split into a new article entitled HE-AAC. (Discuss)

High Efficiency AAC is a lossy data compression scheme for streaming audio. It combines three techniques: Advanced Audio Coding (AAC), Spectral Band Replication (SBR), and Parametric Stereo (PS) (only in v2).

The aacPlus codec, sometimes written as AAC+, was standardized by the MPEG under the High Efficiency AAC (HE-AAC) name. The codec can operate at very low bitrates and is good for Internet radio streaming. It is claimed that a 48 kilobit-per-second stream will produce higher quality output than MP3 at 128 kbit/s. While listening tests[1] suggest that this claim is unfounded, aacPlus generally outperforms other popular codecs in the 32-64 kbit/s range. Its main competition for delivering acceptable quality at these bitrates comes from MP3pro, which also employs SBR.

AacPlus is supported in the open source FAAD/FAAD2 decoding library (and all players incorporating it), Winamp, foobar2000, and Sony's latest SonicStage version 4.

iTunes does not yet support aacPlus,so the only players for Mac OS X are VLC, MPlayerOSX and Songbird.

Nero has released a free of charge command line HE-AAC encoder, and also support HE-AAC inside the Nero software suite.

The 3GPP consortium released source code of a reference HE-AACv2 encoder that appears to be quite competitive regarding quality[2]. No community-supported Free Software encoder is available yet.

[edit] External links

[edit] AAC-SSR

AAC Scalable Sample Rate was introduced by Sony to the MPEG-4 standard. The audio signal is first split into 4 bands using a 4 band polyphase quadrature filter bank. Then these 4 bands are further split using MDCTs with a size k of 32 or 256 samples. This is similar to normal MPEG-4 AAC which uses MDCTs with a size k of 128 or 1024 directly on the audio signal.

The advantage of this technique is that short block switching can be done separately for every PQF band. So high frequencies can be encoded using a short block to enhance temporal resolution, low frequencies can be still encoded with high spectral resolution. However, due to aliasing between the 4 PQF bands coding efficiencies around (1,2,3) * fs/8 is worse than normal MPEG-4 AAC.

MPEG-4 AAC-SSR is very similar to ATRAC and ATRAC-3.

[edit] Why AAC-SSR was introduced

The idea behind AAC-SSR was not only the advantage listed above, but also the possibility of reducing the data rate by removing 1, 2 or 3 of the upper PQF bands. A very simple bitstream splitter can remove these bands and thus reduce the bitrate and sample rate.

Example:

  • 4 subbands: bitrate = 128 kbit/s, sample rate = 48 kHz, f_lowpass = 20 kHz
  • 3 subbands: bitrate ~ 120 kbit/s, sample rate = 48 kHz, f_lowpass = 18 kHz
  • 2 subbands: bitrate ~ 100 kbit/s, sample rate = 24 kHz, f_lowpass = 12 kHz
  • 1 subband: bitrate ~ 65 kbit/s, sample rate = 12 kHz, f_lowpass = 6 kHz

Note: although possible, the resulting quality is much worse than typical for this bitrate. So for normal 64 kbit/s AAC a bandwidth of 14-16 kHz is achieved by using intensity stereo and reduced NMRs. This degrades audible quality less than transmitting 6 kHz bandwidth with perfect quality.

[edit] BSAC

Bit Sliced Arithmetic Coding is an MPEG-4 standard (ISO/IEC 14496-3 subpart 4) for scalable audio coding. BSAC uses an alternative noiseless coding to AAC, with the rest of the processing being identical to AAC. This support for scalability allows for nearly transparent sound quality at 64 kbit/s and graceful degradation at lower bit rates. BSAC coding is best performed in the range of 40 kbit/s to 64 kbit/s, though it operates in the range of 16 kbit/s to 64 kbit/s. The AAC-BSAC codec is used in Digital Multimedia Broadcasting (DMB) applications.

[edit] See also

[edit] External links

Multimedia compression formats
Video compression formats ISO/IEC MPEG-1 | MPEG-2 | MPEG-4 | MPEG-4/AVC ITU-T H.261 | H.262 | H.263 | H.264 Others AVS | Dirac | Indeo | MJPEG | RealVideo | VC-1 | Theora | VP6 | VP7 | WMV
Audio compression formats ISO/IEC MPEG MPEG-1 Layer III (MP3) | MPEG-1 Layer II | AAC | HE-AAC ITU-T G.711 | G.722 | G.722.1 | G.722.2 | G.723 | G.723.1 | G.726 | G.728 | G.729 | G.729.1 | G.729a Others AC3 | ATRAC | FLAC | iLBC | Monkey's Audio | Musepack | RealAudio | SHN | Speex | Vorbis | WavPack | WMA
Image compression formats ISO/IEC/ITU-T JPEG | JPEG 2000 | JPEG-LS | JBIG | JBIG2 -- -- Others BMP | GIF | ILBM | PCX | PNG | TGA | TIFF | WMP
Media container formats General 3GP | ASF | AVI | FLV | Matroska | MP4 | MXF | NUT | Ogg | Ogg Media | QuickTime | RealMedia Audio only AIFF | AU | WAV -- --
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