Editing MPEG-4 Part 3 (section)

==AAC-SSR==
'''AAC Scalable Sample Rate''' was introduced by Sony to the MPEG-2 Part 7 and MPEG-4 Part 3 standards.{{Citation needed|date=October 2009|reason=There should be a reference to Sony's work on this standard.}} It was first published in ISO/IEC 13818-7, Part 7: Advanced Audio Coding (AAC) in 1997.<ref name="iso13818-7-2004-pdf"/><ref name="iso13818-7-1997"/> 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 [[Modified discrete cosine transform|MDCTs]] with a size ''k'' of 32 or 256 samples. This is similar to normal AAC LC 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 efficiency around (1,2,3) * fs/8 is worse than with normal MPEG-4 AAC LC.{{Citation needed|date=February 2013}}

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

=== 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&nbsp;kbit/s, sample rate = 48&nbsp;kHz, f_lowpass = 20&nbsp;kHz
* 3 subbands: bitrate ~ 120&nbsp;kbit/s, sample rate = 48&nbsp;kHz, f_lowpass = 18&nbsp;kHz
* 2 subbands: bitrate ~ 100&nbsp;kbit/s, sample rate = 24&nbsp;kHz, f_lowpass = 12&nbsp;kHz
* 1 subband: bitrate ~  65&nbsp;kbit/s, sample rate = 12&nbsp;kHz, f_lowpass = 6&nbsp;kHz

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