Abstract:
The invention provides improved ambient noise reduction for ear-worn devices, such as earphones and headphones and for other devices worn upon or used in close proximity to the ear, such as cellular telephone handsets, and it provides, in particular, improvements to "feed-forward" ambient noise-reduction systems. Most feed-forward noise-reduction systems available hitherto purport to operate only below about 1 kHz and, even then, provide only relatively modest amounts of noise reduction. In accordance with this invention, predetermined filter parameters, such as the gain and cut-off frequency of a selected filter stage used in the noise-reduction processing, are mathematically modelled and the model is adjusted in real-time, in response to user-interpretation of a graphical display of a predicted residual noise amplitude spectrum. This allows the user to inspect the predicted residual noise level amplitude spectrum and to iteratively adjust the filter parameters to minimise residual noise in a chosen environment. Instead of being made manually by a user, the iterative adjustments may be automated and implemented under computer control, using known data-fitting methods and/or neural networks.
Abstract:
Noise reduction circuitry for a communication apparatus can apply different noise reduction transfer functions, depending on whether a listening device is connected to the apparatus. If no listening device is connected, the noise reduction transfer function can be adapted for use with microphones (12) and speakers (28) that form an integral part of the communication apparatus, which may be a cellular telephone. If a listening device is connected, the noise reduction transfer function can be adapted for use with microphones (12) and speakers (28) that form a part of the listening device. This allows the noise reduction circuitry to provide improved noise reduction performance.
Abstract:
An adaptive, feed-forward, ambient noise-reduction system includes a reference microphone for generating first electrical signals representing incoming ambient noise, and a connection path including a circuit for inverting these signals and applying them to a loudspeaker directed into the ear of a user. The system also includes an error microphone for generating second electrical signals representative of sound (including that generated by the loudspeaker in response to the inverted first electrical signals) approaching the user's ear. An adaptive electronic filter is provided in the connection path, together with a controller for automatically adjusting one or more characteristics of the filter in response to the first and second electrical signals. The system is configured to constrain the operation of the adaptive filter such that it always conforms to one of a predetermined family of filter responses, thereby restricting the filter to operation within a predetermined and limited set of amplitude and phase characteristics.
Abstract:
The invention relates to telephonic transmission of 3D sound. Existing video conferencing systems suffer from the disadvantage that following transmission of a person speaking, the speaker's voice tends to become 'disembodied'. That is, if a person moves with respect to a microphone, the reproduced voice tends not to move with the speaker. The invention overcomes or reduces this effect by obtaining left and right monophonic signals, modifying the signals to compensate for head related air-to-ear transfer functions and performing equalisation and cross-talk cancellation on the signals. Eventually signals are compressed to produce a compressed binaural signal for transmission along a telephone link so that frequencies are split into separate bands, but relative phase differences between signals in different frequency bands are preserved. 3D sounds are therefore able to be transmitted via telephone links, and reproduced more effectively, than was previously possible.
Abstract:
The invention is intended to facilitate the production of ambient noise- cancelling earphones and, to that end, provides a module (30) comprising a microspeaker (34) and an electret microphone (38) both carried on a common substrate (32) which is also configured to incorporate an acoustic resistor (33, 35). The module (30) is incorporated into an earpiece having electrical connections to noise-cancelling electronic circuitry that is provided separately from the earpiece and is housed, for example, in a separate pod (82, 102) or incorporated within the body of a cellular telephone. The performances of the microspeaker (34) and microphone (38) are classified against one or more predetermined operational criteria, enabling the noise-cancelling circuitry to be configured to allow for departures from such criteria. In some embodiments, the module (30) further comprises an information storage device (40) capable of recording data concerning departures from the aforementioned criteria and of providing, upon interrogation, information over the electrical connection to automatically compensate for such departures. The invention also comprises a method of producing ambient noise-cancelling earphones in which the components on the module (30) are classified inter alia by feeding known signals to the microphone (38) and noting the response of the speaker (34) thereto.
Abstract:
A method of audio signal processing for loudspeaker (1) located close to an ear (2), the method consisting of or including creating a reverberant signal from an original monophonic signal, modifying the spectral characteristics of the original signal using a first pinna transfer function, modifying the spectral characteristics of the reverberant signal using a further pinna transfer function, combining the modified reverberant signal and the modified original signal to form a combined signal, and feeding the combined signal to said loudspeaker, thereby providing cues for enabling the listener to perceive the source of the sound of the original audio signal to be located remote from said ear. The method is particularly advantageous for use in communications apparatus such as telephones or radio transceivers.
Abstract:
The invention is a method of making a mobile telephone more secure and includes generating an identification code, inserting the code into transmitted speech, detecting the code at a base station and comparing it with stored information to verify the authenticity of the mobile telephone. The identification code comprises two portions: the first portion (which stores the code) being produced during manufacture of a chip and the second portion being formed by a randomised process during commissioning of the telephone. The invention overcomes problems associated with similar, prior art systems because the chip containing the identification code has part of its code randomly selected because it is an irreversible process.
Abstract:
A mobile electronic device is programmed so that when the device is running a an application and an event occurs that the device needs to notify the user about, then the device alters the visual and/or sonic behaviour of the application according to, a pre- defined event notification profile. The user can select a desired event notification profile from a menu of available profiles stored in the device. For example, the event notification profile could be graphics and/or audio in the application gently fading to an alternative state using a pre-defined transition effect.
Abstract:
A method of audio signal processing for a loudspeaker located close to an ear in use, the method consisting of or including: creating one or more derived signal from an original monophonic input signal, the derived signals being representative of the original signal being scattered by one or more bodies remote from said ear (excluding room boundary reflection or reverberation), combining the derived signal or signals with said input signal to form a combined signal, and feeding the combined signal to said loudspeaker, thereby providing cues for enabling the listener to perceive the source of the sound of the original monophonic input signal to be located remote from said ear.
Abstract:
A method of generating a second decorrelated audio signal from a first audio signal, for use in synthesising a 3D sound field, includes: a) deriving from the first signal a first delayed signal; b) multiplying this first delayed signal by a gain factor between zero and minus one to give a first delayed gain-adjusted signal; c) deriving from the first audio signal a second delayed signal, having a different delay time from the first delayed signal; d) multiplying this second delayed signal by a gain factor between zero and plus one (such that the said gain factors sum to zero) to give a second delayed gain-adjusted signal; e) combining said first and said second delayed gain-adjusted signals with the first audio signal to provide a second decorrelated audio signal. The first and second delayed signals are delayed by time periods which change in a substantially random manner.