[simple_rtsp_server] Some audio buffers are possibly lost along the way

[Noarkhh]

In Boombox we use this server for tests. When running a "mp4 -> hls" test, the Compare utility unpacks 432 audio buffers from the test output for bun10s.mp4 fixture file. In case of "rtsp -> hls" test Compare unpacks 431 audio buffers (from the same fixture). When an assertion, that the subject and the reference must produce the same amount of buffers, is removed, the "rtsp -> hls" test passes with the same reference as the "mp4 -> hls" test. That makes it seem, that the last buffer in case of streaming with rtsp is somehow lost along the way. The first place I'd look would be RTP AAC payloader and depayloader.