Real-Time Audio Capture, Compression & Streaming Service on a PDA

This paper shows how a PDA (Personal Digital Assstant)can be converted into an audio source within a private network or the web, providing capture, compression and streaming of audio as a real-time mobile service. Once the sound around the PDA has been captured and compressed to mp3 format, the service allows it to be broadcast to a Streaming Server. Once the audio reaches the Streaming Server, anyone with a network connection is able to receive and play it. The service provides different configuration parameters to control audio quality and broadcasting performance. For audio quality, different bitrate and frecuency values can be chosen. For broadcasting performance, different packet-length values can also be chosen, and the bitrate mode can be automatically controlled. The paper also includes the performance tunning of the compressor and experimental results using both wired and wireless networks. 1 SERVICE DESCRIPTION PDAs provide now a wide variety of mobile connections, from standard modem connections to wireless connections. Taking advantage of their mobility and connection capability, a PDA can be converted into an audio source within a private network or the web, providing capture, compression and streaming of audio as a real-time service. For instance, this kind of service could be useful to broadcast live interviews, news reports or press conferences. Once the sound around the PDA has been captured and compressed to mp3 format (Robinson and Hawksford, ) (Branderburg and Popp, ) (Rangachar, 2001), the service allows it to be broadcast to a streaming server. The service uses Real-time Transport Protocols (RTP/RTCP) and Session Description Protocol (SDP). These protocols allow the audio to reach the streaming server. The streaming server must be RTP and SDP compatible, which is the case of the Darwin Streaming Server (Apple) (Computer, ) and the Helix Server (RealNetworks) (RealNetworks, ). Once the audio reaches the streaming server, anyone with a network connection is able to receive and play it. Instead of broadcasting to a streaming server, the audio can be saved in a file and broadcast later. An improved use of free memory and long-time recording capability are the profits of saving voice messages in this way. The service provides different configuration parameters to control audio quality and broadcasting performance. For audio quality, different bitrate and frequency values can be chosen. For broadcasting performance, different packet-length values can also be chosen, and the auto-bitrate mode can be activated. One of the main advantages of the service is the auto adaptable control of the bitrate (in the autobitrate mode), which provides the best bitrate compression at all times. This means the bitrate will lower or rise as the available bandwidth does. The rest of the paper is structured in the following sections: service operation, audio compression (including compressor performance tuning), audio streaming, protocols (RTSP, RTP, RTCP & SDP), implementation details, experimental results (with both bluetooth and modem communication), concluding remarks and references.