Cooperative population coding facilitates efficient sound-source separability by adaptation to input statistics.

Our sensory environment changes constantly. Accordingly, neural systems continually adapt to the concurrent stimulus statistics to remain sensitive over a wide range of conditions. Such dynamic range adaptation (DRA) is assumed to increase both the effectiveness of the neuronal code and perceptual sensitivity. However, direct demonstrations of DRA-based efficient neuronal processing that also produces perceptual benefits are lacking. Here, we investigated the impact of DRA on spatial coding in the rodent brain and the perception of human listeners. Complex spatial stimulation with dynamically changing source locations elicited prominent DRA already on the initial spatial processing stage, the Lateral Superior Olive (LSO) of gerbils. Surprisingly, on the level of individual neurons, DRA diminished spatial tuning because of large response variability across trials. However, when considering single-trial population averages of multiple neurons, DRA enhanced the coding efficiency specifically for the concurrently most probable source locations. Intrinsic LSO population imaging of energy consumption combined with pharmacology revealed that a slow-acting LSO gain-control mechanism distributes activity across a group of neurons during DRA, thereby enhancing population coding efficiency. Strikingly, such "efficient cooperative coding" also improved neuronal source separability specifically for the locations that were most likely to occur. These location-specific enhancements in neuronal coding were paralleled by human listeners exhibiting a selective improvement in spatial resolution. We conclude that, contrary to canonical models of sensory encoding, the primary motive of early spatial processing is efficiency optimization of neural populations for enhanced source separability in the concurrent environment.

Benefits of triple acoustic beamforming during speech-on-speech masking and sound localization for bilateral cochlear-implant users.

Bilateral cochlear-implant (CI) users struggle to understand speech in noisy environments despite receiving some spatial-hearing benefits. One potential solution is to provide acoustic beamforming. A headphone-based experiment was conducted to compare speech understanding under natural CI listening conditions and for two non-adaptive beamformers, one single beam and one binaural, called "triple beam," which provides an improved signal-to-noise ratio (beamforming benefit) and usable spatial cues by reintroducing interaural level differences. Speech reception thresholds (SRTs) for speech-on-speech masking were measured with target speech presented in front and two maskers in co-located or narrow/wide separations. Numerosity judgments and sound-localization performance also were measured. Natural spatial cues, single-beam, and triple-beam conditions were compared. For CI listeners, there was a negligible change in SRTs when comparing co-located to separated maskers for natural listening conditions. In contrast, there were 4.9- and 16.9-dB improvements in SRTs for the beamformer and 3.5- and 12.3-dB improvements for triple beam (narrow and wide separations). Similar results were found for normal-hearing listeners presented with vocoded stimuli. Single beam improved speech-on-speech masking performance but yielded poor sound localization. Triple beam improved speech-on-speech masking performance, albeit less than the single beam, and sound localization. Thus, triple beam was the most versatile across multiple spatial-hearing domains.

The importance of processing resolution in "ideal time-frequency segregation" of masked speech and the implications for predicting speech intelligibility.

Ideal time-frequency segregation (ITFS) is a signal processing technique that may be used to estimate the energetic and informational components of speech-on-speech masking. A core assumption of ITFS is that it roughly emulates the effects of energetic masking (EM) in a speech mixture. Thus, when speech identification thresholds are measured for ITFS-processed stimuli and compared to thresholds for unprocessed stimuli, the difference can be attributed to informational masking (IM). Interpreting this difference as a direct metric of IM, however, is complicated by the fine time-frequency (T-F) resolution typically used during ITFS, which may yield target "glimpses" that are too narrow/brief to be resolved by the ear in the mixture. Estimates of IM, therefore, may be inflated because the full effects of EM are not accounted for. Here, T-F resolution was varied during ITFS to determine if/how estimates of IM depend on processing resolution. Speech identification thresholds were measured for speech and noise maskers after ITFS. Reduced frequency resolution yielded poorer thresholds for both masker types. Reduced temporal resolution did so for noise maskers only. Results suggest that processing resolution strongly influences estimates of IM and implies that current approaches to predicting masked speech intelligibility should be modified to account for IM.

Enhancing the perceptual segregation and localization of sound sources with a triple beamformer.

A triple beamformer was developed to exploit the capabilities of the binaural auditory system. The goal was to enhance the perceptual segregation of spatially separated sound sources while preserving source localization. The triple beamformer comprised a variant of a standard single-channel beamformer that routes the primary beam output focused on the target source location to both ears. The triple beam algorithm adds two supplementary beams with the left-focused beam routed only to the left ear and the right-focused beam routed only to the right ear. The rationale for the approach is that the triple beam processing exploits sound source segregation in high informational masking (IM) conditions. Furthermore, the exaggerated interaural level differences produced by the triple beam are well-suited for categories of listeners (e.g., bilateral cochlear implant users) who receive limited benefit from interaural time differences. The performance with the triple beamformer was compared to normal binaural hearing (simulated using a Knowles Electronic Manikin for Auditory Research, G.R.A.S. Sound and Vibration, Holte, DK) and to that obtained from a single-channel beamformer. Source localization in azimuth and masked speech identification for multiple masker locations were measured for all three algorithms. Taking both localization and speech intelligibility into account, the triple beam algorithm was considered to be advantageous under high IM listening conditions.