Effect of talker and speaking style on the speech transmission index.
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Three experiments investigated factors that influence the creation of and release from informational masking in speech recognition. The target stimuli were nonsense sentences spoken by a female talker. In experiment 1 the masker was a mixture of three, four, six, or ten female talkers, all reciting similar nonsense sentences. Listeners' recognition performance was measured with both target and masker presented from a front loudspeaker (F-F) or with a masker presented from two loudspeakers, with the right leading the front by 4 ms (F-RF). In the latter condition the target and masker appear to be from different locations. This aids recognition performance for one- and two-talker maskers, but not for noise. As the number of masking talkers increased to ten, the improvement in the F-RF condition diminished, but did not disappear. The second experiment investigated whether hearing a preview (prime) of the target sentence before it was presented in masking improved recognition for the last key word, which was not included in the prime. Marked improvements occurred only for the F-F condition with two-talker masking, not for continuous noise or F-RF two-talker masking. The third experiment found that the benefit of priming in the F-F condition was maintained if the prime sentence was spoken by a different talker or even if it was printed and read silently. These results suggest that informational masking can be overcome by factors that improve listeners' auditory attention toward the target.
Three experiments investigated the roles of interaural time differences (ITDs) and level differences (ILDs) in spatial unmasking in multi-source environments. In experiment 1, speech reception thresholds (SRTs) were measured in virtual-acoustic simulations of an anechoic environment with three interfering sound sources of either speech or noise. The target source lay directly ahead, while three interfering sources were (1) all at the target's location (0 degrees,0 degrees,0 degrees), (2) at locations distributed across both hemifields (-30 degrees,60 degrees,90 degrees), (3) at locations in the same hemifield (30 degrees,60 degrees,90 degrees), or (4) co-located in one hemifield (90 degrees,90 degrees,90 degrees). Sounds were convolved with head-related impulse responses (HRIRs) that were manipulated to remove individual binaural cues. Three conditions used HRIRs with (1) both ILDs and ITDs, (2) only ILDs, and (3) only ITDs. The ITD-only condition produced the same pattern of results across spatial configurations as the combined cues, but with smaller differences between spatial configurations. The ILD-only condition yielded similar SRTs for the (-30 degrees,60 degrees,90 degrees) and (0 degrees,0 degrees,0 degrees) configurations, as expected for best-ear listening. In experiment 2, pure-tone BMLDs were measured at third-octave frequencies against the ITD-only, speech-shaped noise interferers of experiment 1. These BMLDs were 4-8 dB at low frequencies for all spatial configurations. In experiment 3, SRTs were measured for speech in diotic, speech-shaped noise. Noises were filtered to reduce the spectrum level at each frequency according to the BMLDs measured in experiment 2. SRTs were as low or lower than those of the corresponding ITD-only conditions from experiment 1. Thus, an explanation of speech understanding in complex listening environments based on the combination of best-ear listening and binaural unmasking (without involving sound-localization) cannot be excluded.
The SII model in its present form (ANSI S3.5-1997, American National Standards Institute, New York) can accurately describe intelligibility for speech in stationary noise but fails to do so for nonstationary noise maskers. Here, an extension to the SII model is proposed with the aim to predict the speech intelligibility in both stationary and fluctuating noise. The basic principle of the present approach is that both speech and noise signal are partitioned into small time frames. Within each time frame the conventional SII is determined, yielding the speech information available to the listener at that time frame. Next, the SII values of these time frames are averaged, resulting in the SII for that particular condition. Using speech reception threshold (SRT) data from the literature, the extension to the present SII model can give a good account for SRTs in stationary noise, fluctuating speech noise, interrupted noise, and multiple-talker noise. The predictions for sinusoidally intensity modulated (SIM) noise and real speech or speech-like maskers are better than with the original SII model, but are still not accurate. For the latter type of maskers, informational masking may play a role.
The speech intelligibility index (SII) (ANSI S3.5-1997) provides a means for estimating speech intelligibility under conditions of additive stationary noise or bandwidth reduction. The SII concept for estimating intelligibility is extended in this paper to include broadband peak-clipping and center-clipping distortion, with the coherence between the input and output signals used to estimate the noise and distortion effects. The speech intelligibility predictions using the new procedure are compared with intelligibility scores obtained from normal-hearing and hearing-impaired subjects for conditions of additive noise and peak-clipping and center-clipping distortion. The most effective procedure divides the speech signal into low-, mid-, and high-level regions, computes the coherence SII separately for the signal segments in each region, and then estimates intelligibility from a weighted combination of the three coherence SII values.
Across-frequency processing by common interaural time delay (ITD) in spatial unmasking was investigated by measuring speech reception thresholds (SRTs) for high- and low-frequency bands of target speech presented against concurrent speech or a noise masker. Experiment 1 indicated that presenting one of these target bands with an ITD of +500 micros and the other with zero ITD (like the masker) provided some release from masking, but full binaural advantage was only measured when both target bands were given an ITD of + 500 micros. Experiment 2 showed that full binaural advantage could also be achieved when the high- and low-frequency bands were presented with ITDs of equal but opposite magnitude (+/- 500 micros). In experiment 3, the masker was also split into high- and low-frequency bands with ITDs of equal but opposite magnitude (+/-500 micros). The ITD of the low-frequency target band matched that of the high-frequency masking band and vice versa. SRTs indicated that, as long as the target and masker differed in ITD within each frequency band, full binaural advantage could be achieved. These results suggest that the mechanism underlying spatial unmasking exploits differences in ITD independently within each frequency channel.
To examine spectral effects on declines in speech recognition in noise at high levels, word recognition for 18 young adults with normal hearing was assessed for low-pass-filtered speech and speech-shaped maskers or high-pass-filtered speech and speech-shaped maskers at three speech levels (70, 77, and 84 dB SPL) for each of three signal-to-noise ratios (+8, +3, and -2 dB). An additional low-level noise produced equivalent masked thresholds for all subjects. Pure-tone thresholds were measured in quiet and in all maskers. If word recognition was determined entirely by signal-to-noise ratio, and was independent of signal levels and the spectral content of speech and maskers, scores should remain constant with increasing level for both low- and high-frequency speech and maskers. Recognition of low-frequency speech in low-frequency maskers and high-frequency speech in high-frequency maskers decreased significantly with increasing speech level when signal-to-noise ratio was held constant. For low-frequency speech and speech-shaped maskers, the decline was attributed to nonlinear growth of masking which reduced the "effective" signal-to-noise ratio at high levels, similar to previous results for broadband speech and speech-shaped maskers. Masking growth and reduced "effective" signal-to-noise ratio accounted for some but not all the decline in recognition of high-frequency speech in high-frequency maskers.
Binaural release from informational masking (IM) was examined in a speech identification task. Target and masker sentences were processed into mutually exclusive frequency bands, thus limiting energetic masking (EM), and presented over headphones. In a baseline condition, both were presented monotically to the same ear (TmMm). Despite minimal frequency overlap between target and masker, the presence of the masker resulted in reduced performance, or IM. Presenting the target monotically and the masker diotically (TmMo) resulted in a release from IM. Release was also obtained by imposing interaural differences in level (ILDs) and in time (ITDs) on the maskers (T(m)M(ILD), T(m)M(ITD)). Any masker with a perceived lateral position that differed from that of a truly monaural stimulus resulted in a similar amount of release from IM relative to TmMm. For binaural targets and maskers (T(o)M(ILD), T(o)M(ITD)), release was seen whenever ITDs or ILDs differed between target and masker. These results suggest that binaural cues can be very effective in reducing IM. Because mechanisms based on differences in perceived location make predictions that are similar to those of nonlocation-based binaural mechanisms, a variant of the equalization-cancellation model is also considered.
In everyday listening, both background noise and reverberation degrade the speech signal. Psychoacoustic evidence suggests that human speech perception under reverberant conditions relies mostly on monaural processing. While speech segregation based on periodicity has achieved considerable progress in handling additive noise, little research in monaural segregation has been devoted to reverberant scenarios. Reverberation smears the harmonic structure of speech signals, and our evaluations using a pitch-based segregation algorithm show that an increase in the room reverberation time causes degraded performance due to weakened periodicity in the target signal. We propose a two-stage monaural separation system that combines the inverse filtering of the room impulse response corresponding to target location and a pitch-based speech segregation method. As a result of the first stage, the harmonicity of a signal arriving from target direction is partially restored while signals arriving from other directions are further smeared, and this leads to improved segregation. A systematic evaluation of the system shows that the proposed system results in considerable signal-to-noise ratio gains across different conditions. Potential applications of this system include robust automatic speech recognition and hearing aid design.
To examine spectral and threshold effects for speech and noise at high levels, recognition of nonsense syllables was assessed for low-pass-filtered speech and speech-shaped maskers and high-pass-filtered speech and speech-shaped maskers at three speech levels, with signal-to-noise ratio held constant. Subjects were younger adults with normal hearing and older adults with normal hearing but significantly higher average quiet thresholds. A broadband masker was always present to minimize audibility differences between subject groups and across presentation levels. For subjects with lower thresholds, the declines in recognition of low-frequency syllables in low-frequency maskers were attributed to nonlinear growth of masking which reduced "effective" signal-to-noise ratio at high levels, whereas the decline for subjects with higher thresholds was not fully explained by nonlinear masking growth. For all subjects, masking growth did not entirely account for declines in recognition of high-frequency syllables in high-frequency maskers at high levels. Relative to younger subjects with normal hearing and lower quiet thresholds, older subjects with normal hearing and higher quiet thresholds had poorer consonant recognition in noise, especially for high-frequency speech in high-frequency maskers. Age-related effects on thresholds and task proficiency may be determining factors in the recognition of speech in noise at high levels.
Speech recognition in noisy environments improves when the speech signal is spatially separated from the interfering sound. This effect, known as spatial release from masking (SRM), was recently shown in young children. The present study compared SRM in children of ages 5-7 with adults for interferers introducing energetic, informational, and/or linguistic components. Three types of interferers were used: speech, reversed speech, and modulated white noise. Two female voices with different long-term spectra were also used. Speech reception thresholds (SRTs) were compared for: Quiet (target 0 degrees front, no interferer), Front (target and interferer both 0 degrees front), and Right (interferer 90 degrees right, target 0 degrees front). Children had higher SRTs and greater masking than adults. When spatial cues were not available, adults, but not children, were able to use differences in interferer type to separate the target from the interferer. Both children and adults showed SRM. Children, unlike adults, demonstrated large amounts of SRM for a time-reversed speech interferer. In conclusion, masking and SRM vary with the type of interfering sound, and this variation interacts with age; SRM may not depend on the spectral peculiarities of a particular type of voice when the target speech and interfering speech are different sex talkers.
The Speech Reception Threshold for sentences in stationary noise and in several amplitude-modulated noises was measured for 8 normal-hearing listeners, 29 sensorineural hearing-impaired listeners, and 16 normal-hearing listeners with simulated hearing loss. This approach makes it possible to determine whether the reduced benefit from masker modulations, as often observed for hearing-impaired listeners, is due to a loss of signal audibility, or due to suprathreshold deficits, such as reduced spectral and temporal resolution, which were measured in four separate psychophysical tasks. Results show that the reduced masking release can only partly be accounted for by reduced audibility, and that, when considering suprathreshold deficits, the normal effects associated with a raised presentation level should be taken into account. In this perspective, reduced spectral resolution does not appear to qualify as an actual suprathreshold deficit, while reduced temporal resolution does. Temporal resolution and age are shown to be the main factors governing masking release for speech in modulated noise, accounting for more than half of the intersubject variance. Their influence appears to be related to the processing of mainly the higher stimulus frequencies. Results based on calculations of the Speech Intelligibility Index in modulated noise confirm these conclusions.
Chinese sentence recognition strongly relates to the reception of tonal information. For cochlear implant (CI) users with residual acoustic hearing, tonal information may be enhanced by restoring low-frequency acoustic cues in the nonimplanted ear. The present study investigated the contribution of low-frequency acoustic information to Chinese speech recognition in Mandarin-speaking normal-hearing subjects listening to acoustic simulations of bilaterally combined electric and acoustic hearing. Subjects listened to a 6-channel CI simulation in one ear and low-pass filtered speech in the other ear. Chinese tone, phoneme, and sentence recognition were measured in steady-state, speech-shaped noise, as a function of the cutoff frequency for low-pass filtered speech. Results showed that low-frequency acoustic information below 500 Hz contributed most strongly to tone recognition, while low-frequency acoustic information above 500 Hz contributed most strongly to phoneme recognition. For Chinese sentences, speech reception thresholds (SRTs) improved with increasing amounts of low-frequency acoustic information, and significantly improved when low-frequency acoustic information above 500 Hz was preserved. SRTs were not significantly affected by the degree of spectral overlap between the CI simulation and low-pass filtered speech. These results suggest that, for CI patients with residual acoustic hearing, preserving low-frequency acoustic information can improve Chinese speech recognition in noise.
The extension to the speech intelligibility index (SII; ANSI S3.5-1997 (1997)) proposed by Rhebergen and Versfeld [Rhebergen, K.S., and Versfeld, N.J. (2005). J. Acoust. Soc. Am. 117(4), 2181-2192] is able to predict for normal-hearing listeners the speech intelligibility in both stationary and fluctuating noise maskers with reasonable accuracy. The extended SII model was validated with speech reception threshold (SRT) data from the literature. However, further validation is required and the present paper describes SRT experiments with nonstationary noise conditions that are critical to the extended model. From these data, it can be concluded that the extended SII model is able to predict the SRTs for the majority of conditions, but that predictions are better when the extended SII model includes a function to account for forward masking.
Individuals with sensorineural hearing losses of both flat and sloping configuration evidence difficulty in identifying stop consonant place of articulation. To assess whether upward spread of masking is responsible for this difficulty, we presented hearing-impaired listeners with stimuli from a /ba da ga/ continuum in both monotic and dichotic (F1 to one ear; F2/F3 to the other ear) listening conditions. In the monotic conditions, listeners with similar audiograms evidence great variability in identification performance. In the dichotic conditions performance did not generally improve. For a few listeners, however, the improvement was striking. At moderate levels of signal presentation, upward spread of masking does not appear to be responsible for the poor identification of place by the majority of listeners with moderate hearing losses.
For 80 male subjects (age 60--90) and 30 female subjects (age 71--89) the monaural Speech-Reception Threshold (SRT) for sentences was investigated under five reverberation conditions at a constant noise level. The reverberation times used were between 0 and 2.3 s. The noise, with the long-term average speech spectrum, had a level of 52.5 dBA. Each subject was assigned to one of several subgroups formed on the basis of the maximum reverberation time at which the subject was still able to understand the sentences correctly The mean SRTs and the standard deviations are given, per subgroup, as a function of reverberation time. It is shown that, for each subgroup, the SRT in different reverberant sound fields can be expressed as a single number, namely the required Speech Transmission Index (STI) as introduced in room acoustics by Houtgast and Steeneken [Acustica 28, 66--73 (1973)]. Furthermore, it is shown that a model of SRT as a function of noise level, developed by Plomp [J. Acoust. Soc. Am. 63, 533--549 (1978)], can be combined with the STI model and can thus include the effect of reverberation.
This article deals with the combined effects of noise and reverberation on the speech-reception threshold for sentences. It is based on a series of current investigations on: (1) the modulation-transfer function as a measure of speech intelligibility in rooms, (2) the applicability of this concept to hearing-impaired persons, and (3) hearing loss for speech in quiet and in noise as a function of age. It is shown that, generally, in auditoria, classrooms, etc. the reverberation time T, acceptable for normal-hearing listeners, has to be reduced to (0.75)DT in order to be acceptable for elderly subjects with a hearing loss of D dB for speech in noise; for listening conditions as in lounges, restaurants, etc. the corresponding value is (0.82)DT.
An articulation index calculation procedure developed for use with individual normal-hearing listeners [C. Pavlovic and G. Studebaker, J. Acoust. Soc. Am. 75, 1606-1612 (1984)] was modified to account for the deterioration in suprathreshold speech processing produced by sensorineural hearing impairment. Data from four normal-hearing and four hearing-impaired subjects were used to relate the loss in hearing sensitivity to the deterioration in speech processing in quiet and in noise. The new procedure only requires hearing threshold measurements and consists of the following two modifications of the original AI procedure of Pavlovic and Studebaker (1984): The speech and noise spectrum densities are integrated over bandwidths which are, when expressed in decibels, larger than the critical bandwidths by 10% of the hearing loss. This is in contrast to the unmodified procedure where integration is performed over critical bandwidths. The contribution of each frequency to the AI is the product of its contribution in the unmodified AI procedure and a "speech desensitization factor." The desensitization factor is specified as a function of the hearing loss. The predictive accuracies of both the unmodified and the modified calculation procedures were assessed by comparing the expected and observed speech recognition scores of four hearing-impaired subjects under various conditions of speech filtering and noise masking. The modified procedure appears accurate for general applications. In contrast, the unmodified procedure appears accurate only for applications where results obtained under various conditions on a single listener are compared to each other.