FIELD OF THE INVENTION

The present invention relates to methods and systems for applying time-based effects in a multi-channel audio reproduction system. Time-based effects means processing based on, but not limited to, delays and/or reverberation. These effects can be obtained via various techniques known in the art guaranteeing signal causality. The effects may be processed in the time-domain, for example feedback delay networks, or the Fourier domain, for example partitioned convolution.

BACKGROUND AND PRIOR ART

Multi-channel audio systems used in large venues, such as a concert hall, may have more the two loudspeakers to provide a more even sound pressure over the area where the audience is located. For instance, loudspeakers may be provided to the side and rear of an audience area to prevent sound pressure levels being lower for audience members further from the stage. This is known as “sound reinforcement” and consists of reproducing the same audio channels at the sides and rear of the audience are as are being reproduced at the stage or front of the audience area. The term loudspeaker may refer to a single enclosure or a number of drivers and enclosures working from the same input signal, so that a multi-channel audio system has two or more signals that each are reproduced on loudspeakers.

Alternatively, or in addition, signal processing may be applied to the audio channels reproduced by loudspeakers in multi-channel audio systems used in large venues. Such signal processing may contribute to an “acoustics enhancement” of the sound in the audience area. For instance, reverberation or “reverb”, echo and other signal processing may be applied to one or more channels reproduced by side or read loudspeakers. Reverb, echo and other signal processing effects are well known in the art. For instance, US patent application US2011/0261966 to Dolby International AB describes a system for applying reverb to down-mixed channels which are then up-mixed for reproduction on loudspeakers.

In larger venues, it is possible that audience members remote from the stage may hear the sound from one of the side or rear loudspeakers before the sound from the front loudspeakers. This has the undesirable consequence that affected audience members will hear the sound coming from the rear or side of the venue while seeing the performance occurring at the front. To avoid this, a fixed time delay or “predelay” is applied to the audio signal reproduced by loudspeakers spaced from the stage. The time delay is chosen so that sound from the front loudspeakers arrives at the audience at least 15 ms before the sound from the rear or side loudspeakers to maintain the perceived direction of sound as emanating from the front/stage. One term used to describe this is maintaining “precedence” in the sound.

SUMMARY OF THE INVENTION

In accordance with one aspect of this invention there is provided a signal processing system for applying time-based effects to an N-channel audio input signal for reproduction on a set of loudspeakers having a predetermined configuration, comprising:a first subsystem that receives the N-channel audio input signal and produces therefrom a first M-channel audio signal;at least one second subsystem, each of which receives the first M-channel audio signal, each second subsystem comprising:an effect unit for applying a time-based effect to each channel of an M-channel audio signal, wherein the time-based effect comprises a minimum delay value;a signal distribution unit that:associates each channel of the first M-channel audio signal with a subset of the loudspeakers; andproduces a second M-channel audio signal from the first M-channel audio signal according to an M×M matrix, each element aijin the M×M matrix including a delay term, wherein the signal distribution unit determines a minimum value of the delay term in each element aijaccording to a distance between at least two loudspeakers in at least one of the i and j subsets of loudspeakers and according to the minimum delay value;the effect unit configured to apply a time-based effect to each channel of the second M-channel audio signal;a mixing unit that produces a K-channel audio signal from the or each second M-channel audio signal.

Preferably, the signal distribution unit determines a minimum value of the delay term in each element aijto be at least the time for sound to travel a maximum distance between loudspeakers in the i and j subsets of loudspeakers.

Preferably, the signal processing system further comprises a plurality of second subsystems.

Preferably, each second subsystem's effect unit is configured to apply a plurality of time-based effects having either a first minimum delay value or a second minimum delay value.

Preferably, each second subsystem's signal distribution unit determines a minimum value of the delay term in each element aijaccording to one of:(a) a distance between adjacent loudspeakers in the j subset of loudspeakers;(b) a time for sound to travel a maximum distance between loudspeakers in the i and j subsets of loudspeakers.

Preferably, each second subsystem's signal distribution unit is configured to determine a minimum value of the delay term in each element aijaccording to criteria (a) if that second subsystem's effect unit's minimum delay value is less than a predetermined threshold value.

Preferably, each second subsystem's signal distribution unit is configured to add predetermined fixed delay value to the minimum value of the delay term in each element aij.

In accordance with another aspect of this invention there is provided a digital signal processing method for applying time-based effects to an N-channel audio input signal for reproduction on a set of loudspeakers having a predetermined configuration, comprising the following processor-implemented steps:producing a first M-channel audio signal from the N-channel audio input signal;associating each channel of the first M-channel audio signal with a subset of the loudspeakers;producing at least one second M-channel audio signal from the first M-channel audio signal according to an M×M matrix, each element aijin the M×M matrix including a delay term, further comprising determining a minimum value of the delay term in each element aijaccording to a distance between at least two loudspeakers in at least one of the i and j subsets of loudspeakers and according to a minimum delay value;applying a time-based effect to each channel of the second M-channel audio signal, wherein the time-based effect comprises the minimum delay value;producing a K-channel audio signal from the or each second M-channel audio signal.

Preferably, the minimum value of the delay term in each element aijis determined to be at least the time for sound to travel a maximum distance between loudspeakers in the i and j subsets of loudspeakers.

Preferably, the method further comprises producing a plurality of second M-channel audio signals from the first M-channel audio signal according to a corresponding M×M matrix for each second M-channel audio signal.

Preferably, the time-based effect comprises either a first minimum delay value or a second minimum delay value.

Preferably, the minimum value of the delay term in each element aijis determined according to one of:a) a distance between adjacent loudspeakers in the j subset of loudspeakers;b) a maximum distance between loudspeakers in the i and j subsets of loudspeakers.

Preferably, the minimum value of the delay term in each element aijis determined according to criteria (a) if the minimum delay value applied to that channel by the time-base effect is less than a predetermined threshold value.

Preferably, the method further comprises adding predetermined fixed delay value to the minimum value of the delay term in each element aij.

