Method of synthesizing an approximate impulse response function

Details Method of synthesizing an approximate impulse response function Method of synthesizing an approximate impulse response function

2000-11-14

2004-05-25

Harvey, Minsun Oh (Department: 2644)

Electrical audio signal processing systems and devices

Binaural and stereophonic

Stereo speaker arrangement

C381S061000

Reexamination Certificate

active

06741711

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to a method of synthesising an approximate impulse response function from a measured first impulse response function in a given sound field. It relates particularly, though not exclusively, to impulse responses in sound fields in which scattering is present.
2. Background
A first aspect of the present invention relates to 3D-audio signal-processing based on Head-Related Transfer Functions (HRTFs), in which recorded sounds can be reproduced so as to appear to originate in full, three-dimensional space around the listener, using only a single pair of audio channels, and reproduced via either a conventional pair of loudspeakers or headphones.
A second aspect of the present invention relates to headphone “virtualisation” technology, in which an audio signal is processed such that, when it is auditioned using headphones, the source of the sound appears to originate outside the head of the listener. (At present, conventional stereo audio creates sound-images which appear—for the most part—to originate inside the head of the listener, because it does not contain any three-dimensional sound-cues.) This application includes single channel virtualisation, in which a single sound source is positioned at any chosen point in space, and two-channel virtualisation, where a conventional stereo signal-pair are processed so as to appear to originate from a virtual pair of loudspeakers in front of the listener. This method also extends to the virtualisation of multi-channel cinema surround-sound, in which it is required to create the illusion that the headphone listener is surrounded by five or more virtual loudspeakers.
Another aspect of the invention relates to its application in virtual 3D-reverberation processing.
A co-pending patent application, filed together with the present application, provides a comprehensive explanation of the difficulty in creating effective headphone “externalisation” (including prior art), and describes the method by which it can be successfully achieved. Essentially, the inventor found that wave-scattering effects are critical for achieving adequate headphone externalisation. What is meant by this is that, when sound is emitted in a scattering environment (and most practical environments do contain physical clutter which scatters sound-waves), then the wavefront can be considered as becoming fragmented into a multitude of elemental units, each of which is scattered (i.e. reflected, diffracted and partially absorbed) differently by the objects and surfaces present in the room. This multiplicity of elemental components eventually arrive irregularly at the listener's head after different time periods have elapsed (depending on their scattered path-lengths). Consequently, the incoming waves to the listener are characterised by a clean “first-arrival”, straight from the source itself in a direct line to the listener, closely followed by a period of “turbulence” created by the arrival of the multiplicity of scattered elemental waves. Note that this effect occurs both inside rooms, and outside rooms. For example, in a forest, wave-scattering would be predominant; there would be ground-reflections, but no reverberation. In a partially-cluttered room (most real world rooms), then the scattered signals would be experienced before any reflections or reverberation from the walls, and hence scattering is still the dominant effect. The present inventor has discovered that it is the turbulent period which is critical to sound image externalisation for headphone users. In practise, this period begins within a few milliseconds after the first-arrival, builds to a maximum value over a slightly longer time period, and then decays exponentially over a period of tens of milliseconds. This is consistent with the relative scattering path lengths (compared to the direct sound path) lying in the range from one meter to ten or more meters. The maximum amplitude of the envelope of the turbulent signal is typically 5 to 20% of the amplitude of the direct signal.
Our co-pending patent application describes practical examples of various embodiments of applications in which the synthesis of wave-scattering effects is required. However, a common feature of these embodiments is the requirement for a “wave-scattering” filter, which would simulate the turbulent period of scattered-wave arrivals. This can be accomplished in a conventional manner by means of a digital finite-impulse response (FIR) filter, in which the impulse response of the scattering environment could be measured and replicated, sample by sample. However, at a typical audio sampling rate of 44.1 kHz, then in order to simulate a sufficiently long period of turbulence (say, 100 ms in duration), then a single filter would need to be 4,100 taps in length (and two of these would be needed for many applications). This is impracticably long, by almost two orders of magnitude. For comparison, when HRTF processing is carried out on the CPU of a computer, it is common to use pairs of 25-tap FIR filters, and no more than eight of these can be tolerated in interactive computer applications at present (i.e. 200 taps), otherwise the CPU becomes excessively burdened. As a rule of thumb, it would be useful if the turbulent period of wave-scattering could be simulated using a signal-processing engine having a processing requirement which corresponds to a 100-tap (or less) FIR filter.
SUMMARY OF THE INVENTION
In summary, what is required is a processing-efficient means of reproducing the turbulent features of audio wave-scattering effects as they occur at the ears of the listener. It is an aim of the present invention to provide a method for achieving this goal.
According to a first aspect of the present invention there is provided a method as specified in claims
1
-
13
.
According to a second aspect of the present invention there is provided a method as specified in claims
14
-
15
.
According to a third aspect of the present invention there is provided a impulse response function as specified in claim 16.
According to a fourth aspect of the present invention there is provided an audio signal as specified in claim 17.
According to a fifth aspect of the present invention there is provided signal processing apparatus as specified in claim 18.
According to a sixth aspect of the present invention there is provided a portable audio system as specified in claim 19.
According to a seventh aspect of the present invention there is provided a mobile or cellular telephone handset as specified in claim 20.
According to an eighth aspect of the present invention there is provided an electronic musical instrument as specified in claim 21.
According to a ninth aspect of the present invention there is provided a signal processing system for adding reverberation to an audio signal as claimed in claim 22.


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