I am using transcoder with module that switches between pre-recorded file played in a loop and live stream. It uses Stream and Playlist classes for that. Output stream is transcoded into 3 bitrates and sent to Akamai.
Unfortunately if pre-recorded file has audio with different sample rate than live stream (for example one has 48kHz and other 44.1kHz) then after switch there is no audio in transcoded stream when played on iPad or OSMF player (RTMP). It started working ok after I reencoded the file so it has the same audio as live stream.
Is there a way to configure transcoder to force exactly the same audio output no matter what comes on input?
Another problems is that this setup does not work with sanjosestreamingpacketizer reliably. When tested on Strobe Player 1.6 it often goes into endless buffering state on switch from pre-recorded to live. I am not sure if this is Wowza's or Strobe's fault.
I also tested it with DVR - on switch it complains about negative or too long packet. I believe it is because transcoder just passes original timestamps along instead of regenerating them.