Most lossless audio-coding algorithms are designed for PCM input sound formats. Little work has been done on the lossless compression of IEEE floating-point audio files. An efficient lossless-coding algorithm that handles IEEE floating-point format data as well as PCM format data is described in this paper. In the worst-case scenario, where the algorithm was applied to artificially generated 48-kHz sampling frequency and 32-bit floating-point sound files, an average compression ratio of 65% was still achieved for sound files with 48kHz sampling frequency. Moreover, the proposed algorithm is easily extensible to lossless/variable-lossy operation, which will provide scalability to accommodate the requirements of a wider range of applications and platforms.

Authors: Yang, Dai; Moriya, Takehiro
Affiliation: TT Cyber Space Laboratory, Musashio-Shi, Tokyo, Japan
AES Convention: 115 (October 2003) Paper Number: 5987
Publication Date: October 1, 2003
Session Subject: Psychoacoustics; Audio Coding

Click to purchase paper as a non-member or you can login as an AES member to see more options.

No AES members have commented on this paper yet.

Subscribe to this discussion

To be notified of new comments on this paper you can subscribe to this RSS feed. Forum users should login to see additional options.

Start a discussion!

If you would like to start a discussion about this paper and are an AES member then you can login here:
Username:
Password:

If you are not yet an AES member and have something important to say about this paper then we urge you to join the AES today and make your voice heard. You can join online today by clicking here.