Abstract
This paper presents a new scalable coder based on the ITU-T G.723.1 standard which is one of the most famous speech coders for VoIP applications. In order to support both bit-rate scalability and bandwidth scalability, the proposed coder adopts a split-band approach, where the input signal, sampled at 16 kHz, is decomposed into two equal frequency bands. The lower-band speech is coded with the standard coder such as the G.723.1 standard. In addition, the low-band enhancement layer for the lower-band speech improves the perceptual quality of decoded speech by employing additional coding units based on a cascaded codebook approach. The higher-band signal is encoded using an MDCT-based transform coding scheme. The proposed coder at a bit-rate of 19.4 kbit/s provides speech quality comparable to the ITU-T 24 kbit/s G.722.1 coder, while it also has interoperability with G.723.1.
| Original language | English |
|---|---|
| Pages (from-to) | I285-I288 |
| Journal | ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings |
| Volume | 1 |
| Publication status | Published - 2004 |
| Event | Proceedings - IEEE International Conference on Acoustics, Speech, and Signal Processing - Montreal, Que, Canada Duration: 2004 May 17 → 2004 May 21 |
All Science Journal Classification (ASJC) codes
- Software
- Signal Processing
- Electrical and Electronic Engineering