The increased mobility in society has led to a need for convenient mobile communication in many different type of environments. Environments such as a motorcycle helmet, engine rooms and most industrial sites share a common challenge in that they often offer significant acoustic background noise. Noise reduces the speech intelligibility and consequently limits the potential of mobile speech communications. Existing single channel solutions for speech enhancement may perform adequately when the level of noise is moderate. When the noise level becomes significant, additional use of the spatial domain in order to successfully perform speech enhancement is a potential solution. This is achieved by including several microphones in an array placed in the vicinity of the person speaking. A beamforming algorithm is hereby used to combine the microphones such that the desired speech signal is enhanced. The interest in using microphone arrays for broadband speech and audio processing has increased in recent years. There have been a considerable amount of interesting applications published using beamforming techniques for hands-free voice communication in cars, hearing-aids, teleconferencing and multimedia applications. Most of proposed solutions deal exclusively with environments where the noise is moderate. This thesis is a study of noise reduction in a helmet communication system on a moving motorcycle. The environment is analyzed under different driving conditions and a speech enhancement solution is proposed that operates successfully in all driving conditions. The motorcycle environment can exhibit extremely high noise levels, when driving at high speed, while it can produce a low noise levels at moderate speeds. This fact implies that different solutions are required. It is demonstrated in the thesis that a cascaded combination of a calibrated subband beamforming technique, together with a single channel solution provides good results at all noise levels. The proposed solution operates in the frequency domain, where all microphone signals are decomposed by a subband filter bank prior to the speech enhancement processing. Since the subband transformation is an important component of the overall system performance, a method for filter bank design is also provided in the thesis. The design is such that the aliasing effects in the transformations are minimized while a small delay of the total system is maintained.