Abstract
In recent years, methods for speech processing in cochlear implant systems have been proposed in which frequency modulation is extracted as well as amplitude modulation in different bands. In this paper besides considering one of the above strategies, methods for improving sample reduction and computational efficiency are also presented. In this case adaptive method according to processing parameters and an input signal has been suggested that due to the use of fewer samples to synthesize the signal is better than the other methods and is significantly efficient.