Sound Source Localization and Beamforming for Teleconferencing Solutions
In teleconferencing the audio quality is key to conducting successful meetings. The conference room setting imposes various challenges on the speech signal processing, such as noise and interfering signals, reverberation, or participants positioned far from the telephone unit. This work aims at improving the received speech signal of a conference telephone by implementing sound source localization and beamforming. The implemented microphone array signal processing techniques are compared to the performance of an existing multi-microphone solution and evaluated under various conditions using a planar uniform circular array. Recordings of test-sequences for the evaluation were performed using a custom-built array mockup. The implemented algorithms did not show good enough performance to motivate the increased computational complexity compared to the existing solution. Moreover, an increase in number of microphones used was concluded to have little or no effect on the performance of the methods. The type of microphone used was, however, concluded to have impact on the performance and a subjective listening evaluation indicated a preference for omnidirectional microphones which is recommended to investigate further.
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