Audio Modes
Reference Guide: evolution 5000 E5780 and E5782 Encoder
Page F-3
ST.RE.E10135.4
F.1
A Brief Introduction to Coding Standards
Where appropriate, the output transport stream can be made compliant with ATSC A53(E)
ATSC Digital Television Standard and DVB 101-154 v1.7.7.
F.1.1 MPEG
The Moving Pictures Experts Group (MPEG) was formed in 1988 to generate compression
techniques for audio and video. In the first version, ISO/IEC 11172-3 MPEG-1 audio, has a
selection of two separate algorithms. MPEG-1 Layer I and II were implementations of the
MUSICAM algorithm and MPEG-1 Layer III (mp3) was an implementation of the ASPEC
algorithm. The algorithms have since been improved and extended with other versions of
MPEG.
MPEG-1 Layer I/II
This algorithm is similar to MUSICAM and only really differs in the structure of the frame
headers. Layer I is a restricted version of the full algorithm to allow a reduced decoder to be
developed. Hence, over time as the processing power of decoders have increased by orders
of magnitude, Layer I is no longer used for broadcast.
The algorithm creates 3 frames of 384 samples. Each small frame is divided into subbands
and these subbands can be coded for each frame or for all 3. There is limited ability to
allocate bits to different bands and there is no entropy coding of the encoded samples so a
relatively high bitrate is required to obtain a reasonable quality.
This is selectable from the Audio A and B menu.
MPEG-2 AAC-LC (Advanced Audio Coding)
Further audio research after the completion of MPEG-1 and MPEG-2 standard allowed the
development of a significantly improved coding standard that was not compatible to the
previous standards. The target was to achieve good audio quality at 320 kbit/s for a 5.1
channel system. It has a different frame size that more closes represents the frequency
response of the error, more stereo encoding tools and advanced entropy coding of the
encoded samples.
There are 3 profiles or versions available:
•
Main (MP): includes all of the tools that improve encoding efficiency.
•
Low complexity (LC): some tools are not allowed and others are restricted to enable this
algorithm to fit into the broadcast space.
•
Scalable Sample Rate (SSR): maximises temporal resolution (getting the high frequency
sounds at the right time) at the expense of coding efficiency. This is similar to Sony’s
ATRAC.
If licenced, this is selectable from Advanced Audio 4A – 4D menus. The minimum allowable
delay is 100ms.
F.1.2 Dolby
Digital
Dolby Digital is an algorithm from Dolby that forms part of both the ATSC and DVB standard
for digital broadcasting. It is marketed under the name of Dolby Digital.
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