Todd Schneider

A flexible filterbank structure for extensive signal manipulations in digital hearing aids

Filterbanks for digital hearing aids must use significantly different criteria than those designed for coding applications. For digital hearing aids, the filterbank channel gains must be adjustable over a large dynamic range to compensate for the hearing loss. This adjustability violates the alias cancellation properties of critically sampled filterbanks designed for coding. This paper describes a filterbank designed exclusively for hearing aid applications. Consideration will be given to the extremely limited memory, low delay and low power requirements that must be met in a typical hearing aid application.

A multichannel compression strategy for a digital hearing aid

Multi-channel compression schemes are a practical methods of mapping the wide dynamic range of speech signals into the reduced dynamic range of hearing impaired listeners. These systems address two of the shortcomings of single-channel compression systems: (1) the reduction of gain as a result of narrow-band non-speech stimuli and (2) the reduction of gain that often occurs when high-frequency speech components are followed by intense low-frequency speech components. They also provide frequency-dependent compression ratios that are needed by many newer supra-threshold fitting strategies (e.g., DSL I/O). This paper presents a multichannel compression scheme that employs an oversampled, polyphase DFT filterbank. In each compressor channel, the gain is controlled by an adjustable combination of a overall, dual time-constant input signal level and the individual channel signal level that is measured with a short time-constant RMS detector. Informal listening tests have demonstrated that the design has very good audio quality and performs well in real-world listening situations. The design is suited for low-power, real-time operation.

Real-time speech synthesis on an ultra low-resource, programmable DSP system

An efficient implementation of a time-domain speech synthesis system on an ultra low-power, miniature, programmable block-floating-point DSP system is introduced. The DSP system, operating at a clock rate as low as 1.28 MHz, is well suited for speech and audio processing applications. Similar to the MBR-PSOLA technique, this time-domain synthesis method uses a normalized speech database generated by a high-quality harmonic synthesis. To reduce the memory usage and communication bandwidth, the normalized database is compressed using a block-adaptive, ADPCM approach. Listening tests comparing the synthetic speech quality on the DSP system and the same method implemented on a high-resource computer system show no degradations due to the memory, register length, or other low-resource limitations on the DSP system. The system consumes less than 1 mW at 1 volt.

A subband beamformer on an ultra low-power miniature DSP platform

This paper presents the design and implementation of a subband cardioid beamformer on an ultra low-power miniature DSP platform, using a 2-microphone endfire array. The subband beamformer extends the classical time-domain, narrow-band algorithm to a frequency-domain, broadband implementation, so it is suitable for general speech and audio applications. An oversampled, weighted overlap-add filterbank is used to allow wide gain and phase adjustments for low power, low group delay requirements. A subband IIR filter is proposed to overcome the non-zero bandwidth of the frequency bands, and to introduce a nearly linear phase adjustment across the bands. The subband implementation allows the flexibility of integrating the beamformer with additional algorithms at different frequency ranges. The beamformer has been implemented in real-time on Dspfactorys Toccata platform, which has been specifically designed for ultra low-power, miniature, head-mounted audio devices. At 1.25 Volts with a 5 MIPS DSP core, the Toccata consumes only about 800 micro Watts without microphones and receivers.

Using maximum length sequence coherence for broadband distortion measurements on hearing aids

Measurement procedures using broadband noise stimuli for testing hearing aids have recently been standardized in an attempt to ensure that test results more closely reflect the ‘‘real‐world’’ performance of hearing aids. Although a number of researchers have employed coherence measures made with broadband noise stimuli to characterize the broadband distortion of hearing aids, broadband distortion measurement methods have not yet been standardized. In this paper, it is demonstrated through simulations and measurements on hearing aids that similar results are obtained when coherence is measured using maximum length sequence (MLS) and conventional unbiased statistical‐based methods. The calculation of single‐ and dual‐channel MLS coherence are also formalized. Signal‐to‐distortion ratio (SDR) measurements for two automatic signal processing hearing aids are presented.

Electroacoustic characterization of hearing aids: a system identification approach

The accurate electroacoustic characterization of hearing aids is important for the design, assessment and fitting of these devices. With the prevalence of modern adaptive processing strategies (e.g., level-dependent frequency response, multi-band compression etc.) it has become increasingly important to evaluate hearing aids using test stimuli that are representative of the signals a hearing aid will be expected to process (e.g., speech). Nearly all current hearing aid tests use stationary test signals that can characterize only the steady-state performance of a hearing aid. The present research examines the characteristics of automatic signal processing hearing aids with natural-speech input signals that may cause the hearing aid response to time-vary. They have investigated a number of linear system identification techniques that can be used to develop time-varying models of hearing aids. Using these models, one can begin to characterize performance of hearing aids with real-world signals and explore speech-based transient distortion measures.

Speech‐based measures of hearing aid processing

The electroacoustic properties of many modern hearing aids change as a function of the input stimuli. One consequence is that electroacoustic measurements obtained using commercial hearing aid test systems may differ from those made with important, ‘‘real‐life’’ signals. As speech is the most important real‐life signal for most hearing aid users, tests using speech stimuli have been proposed. This paper describes one approach to such testing, using spectrograms to display speech signals in a three‐dimensional representation. Dual spectrogram displays prepared using the computerized speech research environment (CSRE) software allow speech signals at the input and output of a hearing aid to be compared. Such spectrograms are computed for a set of important phonemes extracted from running speech. The results of these analyses are evaluated for individual hearing aid users using a display which includes the user’s thresholds and loudness discomfort levels (LDLs), together with the target amplification levels ...

Maximum length sequence based testing of hearing aids

An automated dual‐channel maximum‐length sequence (MLS) test system for the electroacoustic characterization of hearing aids has been developed. This test system applies a speech‐shaped MLS acoustically to a hearing aid and measures the electro‐acoustic frequency response. This method provides results that compare favorably to those obtained using the method employed by Kates [J. M. Kates, J. Rehab. Res. Develop. 27, 255–278 (1990)] and the ANSI standard method [ANSI S3.42 (1992)]. Test results show that MLS‐based testing is significantly faster than noise‐based testing. Two signal‐biased MLS‐based testing methods have also been developed. These methods apply a bias signal to force the hearing aid into a mode of operation where the frequency response, with adaptive or subtractive bias signal cancellation, is measured using a low‐level, speech‐shaped MLS. This method has proved valuable for the characterization of automatic signal processing hearing aids. [Work supported by ORTC and OMH.]