Just saw an article talked about this about Audio streaming. The general content is that the Receiver detecting clock does not match the problem, and then compensates for why it is necessary to do this in the Receiver side, which seems to have this necessary only when the Receiver Buffer is too small. However, similar algorithm is available elsewhere, such as the Sender end and the buffer of Player. The compensation algorithm seems to be good, but in INSERT has not been compared with the General Error ConceAlment algorithm, it seems to have maintained Audio's Smoothness, which looks like
