Abstract: A digital hearing aid according to the present invention is capable of measuring its own performance. The hearing aid includes a test signal generator for feeding a test signal into the hearing aid amplifier. The response to the test signal is acquired at a specific point in the hearing aid, depending upon what aspect of performance is to be measured. Various elements of the hearing aid and/or the hearing aid feedback may be bypassed. The hearing aid further includes the capability of initializing hearing aid parameters based upon the performance measurements. The measurement and initialization capability may be entirely integral to the hearing aid, or an external processor may be used to download the measurement program and the run time program, and assist in computing the parameters.

Abstract: The present invention describes a device and methods for synthesizing a musical audio signal. The invention includes a device for storing a collection of sound segments taken from idiomatic musical performances. Some of these sound segments include transitions between musical notes such as the slur from the end of one note to the beginning of the next. Much of the complexity and expressivity in musical phrasing is associated with the complex behavior of these transition segments. The invention further includes a device for generating a sequence of sound segments in response to an input control sequence—e.g. a MIDI sequence. The sound segments are associated with musical gesture types. The gesture types include attack, release, transition, and sustain. The sound segments are further associated with musical gesture subtypes. Large upward slur, small upward slur, large downward slur, and small downward slur are examples of subtypes of the transition gesture type.

Abstract: Tonal audio signals can be modeled as a sum of sinusoids with time-varying frequencies, amplitudes, and phases. An efficient encoder and synthesizer of tonal audio signals is disclosed. The encoder determines time-varying frequencies, amplitudes, and, optionally, phases for a restricted number of dominant sinusoid components of the tonal audio signal to form a dominant sinusoid parameter sequence. These components are removed from the tonal audio signal to form a residual tonal signal. The residual tonal signal is encoded using a residual tonal signal encoder (RTSE). In one embodiment, the RTSE generates a vector quantization codebook (VQC) and residual codebook sequence (RCS). The VQC may contain time-domain residual waveforms selected from the residual tonal signal, synthetic time-domain residual waveforms with magnitude spectra related to the residual tonal signal, magnitude spectrum encoding vectors, or a combination of time-domain waveforms and magnitude spectrum encoding vectors.

Abstract: A hearing aid has a built in or internal test tone generator for providing test tones and noise for diagnostic tests to a user through the receiver of the hearing aid. Alternatively an external test tone generator may be coupled to the hearing aid and selectively coupled to the receiver of the hearing aid for the diagnostic tests. A memory internal to the hearing aid may store real world sounds for diagnostic tests to simulate actual usage of the hearing aid.

Abstract: The present invention describes methods and means for estimating the time-varying spectrum of an audio signal based on a conditional probability density function (PDF) of spectral coding vectors conditioned on pitch and loudness values. Using this PDF a time-varying output spectrum is generated as a function of time-varying pitch and loudness sequences arriving from an electronic music instrument controller. The time-varying output spectrum is converted to a synthesized output audio signal. The pitch and loudness sequences may also be derived from analysis of an input audio signal. Methods and means for synthesizing an output audio signal in response to an input audio signal are also described in which the time-varying spectrum of an input audio signal is estimated based on a conditional probability density function (PDF) of input spectral coding vectors conditioned on input pitch and loudness values.

Abstract: A digital signal processing hearing aid is disclosed having a plurality of digital signal processing means for processing input digital signals, and a selector switch manipulable by a user for choosing which of the processing means to utilize. Each of the digital signal processing means is designed to provide optimal results in a particular listening environment. Since the user is allowed to choose which of the plurality of processing means to invoke, and since each processing means is specifically designed to operate in a particular listening environment, the hearing aid is capable of providing excellent results in a plurality of listening environments.

Abstract: An improved multiband audio compressor is well behaved for both wide band and narrow band signals, and shows no undesirable artifacts at filter crossover frequencies. The compressor includes a heavily overlapped filter bank, which is the heart of the present invention. The filter bank filters the input signal into a number of heavily overlapping frequency bands. Sufficient overlapping of the frequency bands reduces the ripple in the frequency response, given a slowly swept sine wave input signal, to below about 2 dB, 1 dB, or even 0.5 dB or less with increasing amount of overlap in the bands. Each band is fed into a power estimator, which integrates the power of the band and generates a power signal. Each power signal is passed to a dynamic range compression gain calculation block, which calculates a gain based upon the power signal. Each band is multiplied by its respective gain in order to generate scaled bands. The scaled bands are then summed to generate an output signal.

Abstract: Apparatus for simultaneously decompressing and interpolating compressed audio data. The compressed audio data is stored in differential log format, meaning that the difference between each two consecutive data points is taken and the log of the difference calculated to form each compressed data point. To efficiently decompress and interpolate the compressed data, advantage is taken of the fact that addition of logs is equivalent to multiplication of linear values. Thus the log of an interpolation factor is added to each compressed data point prior to taking the inverse log of the sum. An integrator block completes the interpolation and decompression of the data.

Abstract: A detachable digital binaural processing hearing aid comprised of a digital signal processor (DSP), two microphones, two receivers, a bi-directional communications link between each microphone/receiver and the digital signal processor, an analog-to-digital converter, and a digital-to-analog converter. In one embodiment of the present invention, the user has the option of disabling the digital signal processor by either physically removing an external digital processing unit or by disabling a digital processor to permit an analog processor to provide audio enhancement. The user is also given the option of selecting from a variety of digital filters/compressors that generate binaural signals that are sent to both ears of the user. In a second embodiment, each hearing element comprises a digital signal processor and a communication link to the other hearing element.

Abstract: A parametric signal modeling musical tone synthesizer utilizes a multidimensional filter coefficient space consisting of many sets of filter coefficients to model an instrument. These sets are smoothly interpolated over pitch, intensity, and time. The filter excitation for a particular note is derived from a collection of single period excitations, which form a multidimensional excitation space, which is also smoothly interpolated over pitch, intensity and time. The synthesizer includes effective modeling of attacks of tones, and the noise component of a tone is modelled separately from the pitched component. The input control signals may include initial pitch and intensity, or the intensity may be time-varying. A variety of instruments may be specified.

Abstract: In this invention noise in a binaural hearing aid is reduced by analyzing the left and right digital audio signals to produce left and right signal frequency domain vectors and thereafter using digital signal encoding techniques to produce a noise reduction gain vector. The gain vector can then be multiplied against the left and right signal vectors to produce a noise reduced left and right signal vector. The cues used in the digital encoding techniques include directionality, short term amplitude deviation from long term average, and pitch. In addition, a multidimensional gain function based on directionality estimate and amplitude deviation estimate is used that is more effective in noise reduction than simply summing the noise reduction results of directionality alone and amplitude deviations alone. As further features of the invention, the noise reduction is scaled based on pitch-estimates and based on voice detection.

Abstract: An audio signal in a hearing aid is enhanced by detecting the power of the desired audio signal and the power of the total audio signal, generating a power value and making a noise-reduction adjustment or no noise-reduction adjustment based on the power value. In one embodiment, the power value is a function of the total power of the audio signal. In a second embodiment the power value is a function of the ratio of:the power of the desired audio signal to the power of the total audio signal.When the noise reduction is accomplished with beamforming, the invention uses a direction estimate vector in combination with a beam intensity vector, which is based on the power value, to generate a beamforming gain vector. The direction estimate vector is scaled by the beam intensity vector; the product of the vectors is the beamforming gain vector. The beamforming gain vector is multiplied with the left and right signal frequency domain vectors to produce noise reduced left and right signal frequency domain vectors.

Abstract: This invention relates to a hearing enhancement system having an ear device for each of the wearer's ears, each ear device has a sound transducer, or microphone, and a sound reproducer, or speaker, and associated electronics for the microphone and speaker. Further, the electronic enhancement of the audio signals is performed at a remote digital signal processor (DSP) likely located in a body pack worn somewhere on the body by the user. There is a down-link from each ear device to the (DSP) and an up-link from the DSP to each ear device. The DSP digitally interactively processes the audio signals for each ear based on both of the audio signals received from each ear device. In other words, the enhancement of the audio signal for the left ear is based on the both the right and left audio signals received by the DSP.In addition digital filters implemented at the DSP have a linear phase response so that time relationships at different frequencies are preserved.