Wireless technologies, characterized by their dynamically varying bandwidth and signal quality, provide a harsh environment for audio and video applications. Restricted bandwidth leads to losses and delays, which significantly reduce the quality of playback at the receiver. In this paper, we present a coding and transport technique, which allows the bandwidth used by real-time multicast audio streams to adapt to current network conditions. The Hierarchical Audio for Mobile Hosts (HIERAM) coding scheme divides an audio stream into stream components representing multiple levels of audio resolution. A receiver may select the number of components she wishes to receive, where a larger number of streams provide higher resolution at the cost of higher bandwidth usage. We have implemented this coding and transport mechanism and present results showing its advantages.