AAC Encoder
Introduction
AAC stands for Advanced Audio Coding, a part of MPEG4 (ISO/IEC 14496-3) and MPEG2 (ISO/IEC 13818-3) standards published by ISO/IEC. AAC supports sampling frequency from 8 to 96 khz and bitrates up to 576 kbps.
AAC encoding process consists of the following steps:

  • The signal is converted from time-domain to frequency-domain using forward modified discrete cosine transform (MDCT)
  • The frequency domain signal is quantised based on a psychoacoustic model to discard perceptually irrelevant parts of the signal.
  • The quantised signal is coded using lossless encoding and transmitted.
Salient Features
  • MPEG4 low Complexity profile
  • Supports output ADTS streams
  • Sampling rates – 24kHz, 32kHz, 44.1kHz, 48kHz
  • Bitrates – 64 kbps to 128 kbps
  • Little endian
  • Ported and tested on hardware platform with Linux OS
Applications
  • Portable audio players
  • Streaming
  • Mobile phones
  • Gaming consoles
  • Broadcast audio
Platform
  • MIPS
    • MIPS 74kf
    • Base core, DSP and FPU version available

Case studies

Developing an Asynchronous Sample Rate Converter

Designing an Asynchronous Sample Rate converter that offers high THD and low ripple across a range of frequencies is no mean achievement. We not only designed the ASRC but implemented it with low MHz on a fixed point processor.

Video codecs on a multi-core highly-parallel custom core

We worked with SiliconHive (now part of Intel) to develop High Definition video codecs that are designed to run optimally on a multi-core environment. Our contribution also included efficient coding for a VLIW core and algorithmic innovations to address memory bandwidth constraints.

Showcase

Android Auto projection on Telechip TCC8930 running WinCE 7.0
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RACE Media & Connectivity IVI suite
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