Since the prior-art is based upon analog circuitry, it uses sinusoidal frequency and/or amplitude modulation to simulate a rotating speaker at a reasonable cost. This invention uses a process based upon theoretically derived frequency modulation (FM) and experimentally measured amplitude modulation (AM) to simulate the rotating speaker. The main FM equation is based upon the Doppler effect and is equal to one over one plus a sinusoidal velocity coefficient. The main AM equation has a much narrower peak than sinusoidal modulation. This invention also contains several novel methods to control the angular velocity of the speakers, including changing the horn's speed dependent upon the original audio, modeling the speaker's acceleration to allow the physically realistic transitions between angular velocities, and adding noise to simulate natural variations in rotation. The digital apparatus that implements this invented process includes a digital processor and memory. In summary, the invented process is much more realistic sounding than prior-art.
A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The gain versus frequency function is a set of pure real numbers, so the inverse frequency domain transform gives a set of time-domain filter coefficients. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.
A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.
