Flexible playout adaptation for low delay AAC RTP communication

We present an integrated approach for flexible playout adaptation for high quality audio transmission over impaired network connections. The key concept of our framework is a continuous measurement of the transmission delay, the delay variation, and packet loss. Based on these measurements, the adaptive playout control employs audio time stretching using audio concealment and frame dropping techniques to keep the low delay requirements. In the literature, playout adaptation techniques have mainly been considered for voice over IP, using silence periods between talkspurts, or for high quality audio transmission over dedicated network links. To the best of our knowledge, our playout algorithm is the first achieving low delay high quality audio streaming over impaired network connections for both music and speech. We used a significant number of network traces to estimate the variation of the network quality in DSL, WLAN, UMTS and GPRS links and to update the parameters of our playout adaptation technique. Experimental results clearly indicate that our system provides very high accuracy for the desired accepted late loss rate and achieves a fast playout adaptation, even for rapidly changing network conditions.