In speech communications, signal processing algorithms for near end listening enhancement allow to improve the intelligibility of clean (far end) speech for the near end listener who perceives not only the far end speech but also ambient background noise. A typical scenario is mobile telephony in acoustical background noise such as traffic or babble noise. In these situations, it is often not acceptable/possible to increase the audio power amplification.
In this contribution we use a theoretical analysis of the Speech Intelligibility Index (SII) to develop an algorithm which numerically maximizes the SII under the constraint of an unchanged average power of the audio signal.
These audio samples are 24bit PCM wav files at 48kHz sampling rate. They are 20s long and leveled such that full scale corresponds to 120dBspL.
Processing was performed at 8kHz sampling rate. The speech signal is added only to the right channel in order to simulate a telephone situation. The signal-to-noise ratio before processing is about -1.5dB.
Without processing |
5.5 M |
sauert10_maxtransfer_120db.wav Maximal power transfer algorithm as described in [sauert06b] |
5.5 M |
sauert10_siioptimized_120db.wav Proposed SII optimized algorithm |
5.5 M |
Clean speech only |
2.7 M |
Noise only |
5.5 M |
None so far.
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