An efficient DSP-based implementation of a fast convolution approach with non uniform partitioning

Finite impulse response convolution is one of the most widely used operation in digital signal processing field for filtering operations. In this context, low computationally demanding techniques become essential for calculating convolutions with low input/output latency in real scenarios, considering that the real time requirements are strictly related to the impulse response length. In this context, an efficient DSP implementation of a fast convolution approach is presented with the aim of lowering the workload required in applications like reverberation. It is based on a non uniform partitioning of the impulse response and a psychoacoustic technique derived from the human ear sensitivity. Several results are reported in order to prove the effectiveness of the proposed approach also introducing comparisons with the existing techniques of the state of the art.