As portable voice communications devices become more ubiquitous, they are increasingly used in such noisy environments as airports, outdoor street traffic situations, and bars and clubs.
These conditions make it difficult for the person speaking to be correctly heard and understood on the receiving end of the communications link. Additionally, many communication systems use computer voice recognition, command, and/or response systems. High levels of background acoustic noise can cause high error rates in these types of systems. There is much value, therefore, in being able to improve the speech signal to background acoustic noise ratio.
This two-part article explains the basic principle of using a microphone array for background acoustic noise suppression in a voice communications system.
Microphone arrays
A microphone array is any number of microphones spaced apart from each other in a particular pattern, which work together to produce a resultant output signal or signals. Each microphone is a sensor, or spatial window, for receiving (spatial sampling) the incoming signal. The overall response of the array is a superposition of the responses of each element in the array consistent with the processing algorithm used.
The multiple microphones signals undergo 'array processing' algorithms based on the microphone spacings and patterns, the number and type of microphones, and sound propagation principles. Microphone arrays are typically used to improve speech input signals in the presence of ambient noise in hearing aids, speech recognition equipment, and telecommunication products. But they may also be used to locate the direction and estimate the distance of sounds from the array.
The primary purpose of a microphone array for speech communications is to provide a high quality version of the desired speech signal while at the same time reducing the level of localized and ambient noise signals. The quality aspect means that the resulting speech signal is natural sounding, without any artifacts such as pops and clicks, un-intended muting, frequency distortions, echoes, or aperiodic changes in signal levels associated with the signal processing methods done to achieve the speech enhancement. Therefore, the measure of signal to background noise ratio improvement (SNRI) alone is NOT the only criteria to use in selection of a background noise suppression solution.
Sound Information
Sound Pressure Level
Sound Pressure Level (SPL) decreases proportionately with distance "x" from the sound source. Figures 1 and 2 show the SPL drop off, expressed in dB, as a function of distance "x" from the origin of the sound. For speech, the reference point is generally accepted as 96dB SPL approximately 1cm from the lips of a person talking. The equation plotted is:
dB = 96 " 20log(x/0.01), or alternatively, dB = 96 + 20log(0.01/x)
where the (0.01/x or x/0.01) term is the reference value distance of 0.01m, (1cm) relative to distance "x" in meters
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Both curves show a loss of 6dB for every doubling of distance. Figure 1 is for distances out to 200cm. Figure 2 is a magnified portion for distances out to 50cm, and shows how rapidly sound pressure drops as distance increases from the sound source, even for short distances. For example, at 10cm distance, the SPL drops ~20dB, from 96dB SPL to ~76dB SPL.
Near-Field versus Far-Field Sound
The near field of a sound source is generally considered to be within 1 wavelength of the lowest frequency signal of interest.
If the lowest frequency of interest for speech was 300Hz, then the wavelength λ is equal to c/f or 331.1/300, or 1.104 meters, where c is the speed of sound (331.3 meters/second) at sea level and 0 degrees Celsius. For a frequency of 3500Hz, λ is equal to c/f or 331.1/3500, or 0.0946 meters (9.46cm). Therefore, the general near-field range limit of speech signals extends from ∼9.5 cm to ∼1.1 meters from the source. Beyond ∼1 meter, the speech signal is generally considered to be in the far-field of the speech source.
For closely spaced microphones in an array, near-field sound sources present a spherical wavefront with strong signal amplitude, pressure gradient, and frequency dependent differences based upon the distance from the source for each microphone in the array. For example, assume 2 microphones were placed 3cm apart, at a distance of 5cm for the nearest microphone to the sound source.
