Apparatus and techniques for audio and voice transmission over noisy wireless links

The widespread of audio and voice transmission over wireless networks imposes several challenges upon the design and implementation of efficient transmission schemes. Some of these challenges include reducing the packetization overhead of small audio and voice frames as well as protecting their content against both bit errors introduced by wireless fading channels and packet erasures introduced by network buffering. In this dissertation, we present several optimization frameworks that address the afore-mentioned challenges.;First, we looked into the problem of transmitting a stored audio sequence. To that end, we present an optimization framework designed to optimally protect a stored audio sequence transmitted over a wireless link while considering the packetization overhead of audio frames. Utilizing rate compatible punctured RS codes and dynamic programming, it identifies the optimal assignment of parity to audio frames according to their perceptual importance such that the Segmented SNR of the received audio sequence is maximized. Our framework covers two cases. In the first case, a frame grouping technique is proposed to packetize audio frames and protect them against temporarily correlated bit errors introduced by a fading wireless channel. In this case, each packet is treated as a channel coding codeword. In the second case, a one-dimensional RS coder is applied vertically to a sequence of horizontally formed packets associated with an audio sequence in order to protect the sequence against both bit errors introduced by fading wireless channels and packet erasures introduced by network buffering.;Second, we looked into the problem of transmitting live voice sequences over tandem links, we present two novel frameworks that aim at unequal protection of voice frames according to their perceptual importance such that the expected distortion of a received voice sequence is minimized. The first framework proposes a hybrid media dependent and media independent error correction scheme to mitigate tandem loss. The main objective of the framework is to reduce the buffering delay at both transmitter and receiver, in turn reducing the overall end-to-end delay. The second framework utilizes a rate-adaptation mechanism identifying the optimal allocation of source- and channel-coding components for each voice frame. While the first frame-work offers a better end-to-end delay profile and a better performance in moderate as well as good channel conditions, the second framework provides a stronger error correction capability making it the preferred choice for bad channel conditions and for voice applications that can tolerate a higher transmission delay. Our simulation results showed that the proposed audio and voice transmission schemes have better performance compared to their literature counterpart.