Abstract
The Internet low bit-rate coder (iLBC) is now widely used for voice over Internet protocol (VoIP) applications. Unlike speech coders such as those based on code excited linear prediction (CELP), the iLBC achieves superior robustness to packet loss by avoiding inter-frame coding dependencies. While robustness to packet loss is essential, a VoIP codec should also possess the flexibility to change its source coding rate in order to counter network congestion and facilitate joint source channel coding for wireless channels. Previously, we presented a new variation of the iLBC encoding procedure which yielded a more efficient, rate-flexible result. In an effort to improve performance at lower source rates, we present various improvements to the original framework. Specifically, we reallocate bits from the adaptive codebook procedure; reduce the length of the start state vector; utilize an adaptive pulse gain quantization scheme; and extend the use of entropy coding. Overall, the various combined improvements result in the modified iLBC (with entropy coding) achieving a rate reduction of 2.0 to 2.9 kbps when compared to the original fixed-rate iLBC without any loss in quality. In comparisons with adaptive multi-rate (AMR), the modified iLBC coder remarkably exhibits equivalent perceptual evaluation of speech quality (PESQ) scores as the AMR coder at 10.2 and 12.2 kbps, and out-performs AMR for all packet loss rates. This is a significant result as the modified iLBC performs equivalent to AMR without exploiting inter-frame redundancies