This paper presents a digital signal processing algorithm for efficient and high-quality audio equalization. In this approach, the original full-band audio signal is first downsampled and separated into two or more sub-band signals using a multirate filterbank, after which the equalization is performed in the downsampled domains. After the equalization, the signal is upsampled and combined back to a full-band audio signal. Linear-phase FIR filters, designed based on user-controlled parameters, are used to implement the actual equalization. The method presented in this paper helps in designing an implementation that results in computational savings, while still preserving optimal sound quality with any equalization parameter setting.
Authors:
Hiipakka, Jarmo; Väänänen, Riitta
Affiliation:
Nokia Research Center
AES Convention:
122 (May 2007)
Paper Number:
7110
Publication Date:
May 1, 2007
Subject:
Signal Processing, Sound Quality Design
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.
To be notified of new comments on this paper you can subscribe to this RSS feed. Forum users should login to see additional options.
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.