audio device input is now handle in its own package which resides in the new input directory
a list of codecs was added to codecutil package to help with multiple packages using the same codecs
Since adding the extra bufSize arg to Lex functions, the test functions using them needed to be updated.
NewAudioDevice was changed to accept a logger to log to instead of creating a new one.
It seems unnecessary to have the rtcp.Client.Done() func, considering that we could use
the rtcp.Client.err channel itself to determine if the RTCP client has been stopped.
We simple wait on a chan receive in revid in the error handling routine, and we check the
'ok' return - if it is false, then the err chan has been closed and we can get out of the
error handling loop. This should also reduce CPU usage significantly.
Removed the command line flags that were used to specifiy local and remote addresses for RTP and RTCP. These are now
derived from the initial RTSP connection and also from the RTSP SETUP method reply.
Now adopting an RTCP client so that the RTP stream from the RTSP server can be maintained past 1 minute.
This change involved some refactor.
The rtcp.NewClient signature has been simplified. There is now a default send interval and name for use
in the source description in the receiver reports. These can be customised if required with the new
SetSendInterval and SetName funcs. The rtcp.NewClient signature now takes an rtp.Client, so that it
can get information from the RTP stream, like most recent sequence number. As a result of this requirement
the rtp package parse file has been extended with some functions for parsing out the sequence number and
ssrc from RTP packets and the RTP client provides getters for these things.
Added H264ID and H265ID consts and added logic to select this const for use in encoder based on mediaType param in NewEncoder. Also now
declaring PMT in NewEncoder so that we can set streamID correctly based on mediaType.
Currently just connecting to RTSP server, requesting OPTIONS, DESCRIBE, SETUP and PLAY. Also creating RTP client and giving
this to process from for the lexer.
The audio lexers need to know how much data they will be receiving unlike video which has a fixed buffer size.
This means that all the lex function will need to be given a buffer size since they are used as a function pointer with the same signature.
Alan Noble