SYSTEM FOR AUTOMATIC RECEPTION ENHANCEMENT OF HEARING ASSISTANCE DEVICES
Method and apparatus for automatic reception enhancement of hearing assistance devices. The present subject matter relates to methods and apparatus for automatic reception enhancement in hearing assistance devices. It provides a power estimation scheme that is reliable against both steady and transient input. It provides a TSM estimation scheme that is effective and efficient both in terms of storage size and computational efficiency. The embodiments employing a decision tree provide a weight factor between the omnidirectional and compensated directional signal. The resulting decision logic improves speech intelligibility when talking under noisy conditions. The decision logic also improves listening comfort when exposed to noise. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.
1 . An apparatus, comprising:
an omnidirectional microphone having a first reception profile;
a directional microphone having a second reception profile;
an omni input adapted to receive digital samples representative of signals received by the omnidirectional microphone;
a directional input adapted to receive digital samples representative of signals received by the directional microphone;
a mixing module connected to the omni input, the mixing module providing a mixing ratio for a block of digital samples, α(k);
a compensation filter connected to the directional input and the mixing module, the compensation filter adapted to output a third reception profile which substantially matches the first reception profile;
a first multiplier receiving the omni input and a signal value of (1−α(k)) from the mixing module;
a second multiplier receiving the directional input and a signal value of α(k) from the mixing module; and
a summing stage adding outputs of the first multiplier and the second multiplier,
wherein the output signal for sample n of block k, s c (n,k), is provided by:
s c ( n,k )=(1−α( k ))* s O ( n,k )+α( k ) s D ( n,k ),
where s O (n,k) is the output of the omni microphone for sample n of block k and s D (n,k) is the output of the compensation filter for sample n of block k, and α(k)=C*α(k−1)+(1−C)*β(k), and where C is a constant between 0 and 1 and β(k) is an output from the compensation filter for block k.