Abstract: Electronic devices having digital reverberators are disclosed, together with a method of reproducing sound for a user with the digital reverberator. The digital reverberator uses digital surface absorption filters positioned in the reverberator to simulate absorption of energy as digital audio data samples are reflected from virtual surfaces. The position of the digital surface absorption filters enables known frequency-dependent surface absorption characteristics of real materials to be directly implemented using the filter coefficients of each digital surface absorption filter. This enables virtual acoustic spaces to be designed quickly without the need for the digital reverberator to be manually tuned for each space.

Abstract: Electronic devices having digital reverberators are disclosed, together with a method of reproducing sound for a user with the digital reverberator. The digital reverberator uses digital surface absorption filters positioned in the reverberator to simulate absorption of energy as digital audio data samples are reflected from virtual surfaces. The position of the digital surface absorption filters enables known frequency-dependent surface absorption characteristics of real materials to be directly implemented using the filter coefficients of each digital surface absorption filter. This enables virtual acoustic spaces to be designed quickly without the need for the digital reverberator to be manually tuned for each space.

Abstract: The invention makes use of impulse responses of the performance venue to process a recording or other signal so as to emulate that recording having being recorded in the performance venue. In particular, by measuring or calculating the impulse responses of a performance venue such as an auditorium between an instrument location within the venue and one or more soundfield sampling locations, it then becomes possible to process a “dry” signal, being a signal which has little or no reverberation or other artifacts introduced by the location in which it is captured (such as, for example, a close microphone studio recording) with the impulse response or responses so as to then make the signal seem as if it was produced at the instrument location in the performance venue, and captured at the soundfield sampling location.

Abstract: Embodiments of the invention improve upon the prior art array by having more carefully defined directivity functions designed to meet two criteria, being firstly to minimise cross-talk between non-adjacent microphones in the array, and secondly to design the array response such that it approximates stereophonic panning curves that have been shown to provided for good auditory localisation. One embodiment therefore provides a microphone array, comprising N microphones, wherein N is greater than or equal to 3. The microphones are substantially equiangularly arranged over a circular arc subtending an angle ?, wherein ? is less than or equal to 2?, with the directional axes of the N microphones facing substantially radially outwards.

Abstract: The invention makes use of impulse responses of the performance venue to process a recording or other signal so as to emulate that recording having being recorded in the performance venue. In particular, by measuring or calculating the impulse responses of a performance venue such as an auditorium between an instrument location within the venue and one or more soundfield sampling locations, it then becomes possible to process a “dry” signal, being a signal which has little or no reverberation or other artifacts introduced by the location in which it is captured (such as, for example, a close microphone studio recording) with the impulse response or responses so as to then make the signal seem as if it was produced at the instrument location in the performance venue, and captured at the soundfield sampling location.

Abstract: A non-uniform filter bank is created by joining sections of oversampled uniform filter bands that are based on complex exponential modulation (as opposed to cosine modulation). Each filter bank handles a given, non-overlapping frequency band. The bands are not of uniform bandwidth, and the filters of different banks have different bandwidths. The frequency bands of the different filter banks cover the frequency of interest with gaps in the neighborhoods of the filter band edges. A set of transition filters fills those gaps.

Abstract: A non-uniform filter bank is created by joining sections of oversampled uniform filter bands that are based on complex exponential modulation (as opposed to cosine modulation). Each filter bank handles a given, non-overlapping frequency band. The bands are not of uniform bandwidth, and the filters of different banks have different bandwidths. The frequency bands of the different filter banks cover the frequency of interest with gaps in the neighborhoods of the filter band edges. A set of transition filters fills those gaps.

Abstract: A non-uniform filter bank is created by joining sections of oversampled uniform filter bands that are based on complex exponential modulation (as opposed to cosine modulation). Each filter bank handles a given, non-overlapping frequency band. The bands are not of uniform bandwidth, and the filters of different banks have different bandwidths. The frequency bands of the different filter banks cover the frequency of interest with gaps in the neighborhoods of the filter band edges. A set of transition filters fills those gaps.

Abstract: A non-uniform filter bank is created by joining sections of oversampled uniform filter bands that are based on complex exponential modulation (as opposed to cosine modulation). Each filter bank handles a given, non-overlapping frequency band. The bands are not of uniform bandwidth, and the filters of different banks have different bandwidths. The frequency bands of the different filter banks cover the frequency of interest with gaps in the neighborhoods of the filter band edges. A set of transition filters fills those gaps.

Abstract: A/D conversion is achieved by employing a piecewise continuous dither signal such that a signal that results from combining the dither signal with the signal to be converted, has a zero-crossing within each interval Ij=(jT,(j+1)T), where j is an integer, T=&pgr;/&lgr;B, B is the bandwidth of the signal to be converted, and &lgr; is a constant that is greater than 1, and those zero crossings in adjacent intervals Ij are always separated by some minimal distance, &egr;, for all intervals Ij. Zero crossings of the combined signal are detected, and the instances where those crossings occur are encoded; one zero crossing for each interval.