Use http instead of https in gvctrl getLogin request. Gave RTP client a close method so that the conn can
be closed after we're done with the client. Put timeout on reading from the RTP client PacketConn so that
we don't hang on the ReadFrom call if the conn is closed. Closing the RTSP and RTP clients when
startRTSPCamera is returned.
Added required new params to config structure, and therefore also added new defaults and validation checking.
Also updated revid's update func so that we can control new parameters using vars. Now using gvctrl in
startRTSPCamera function and interpreting revid config to configure camera.
Two fixes were involved, firstly, access unit delimeters were used to denote access units and the Write timeout on rtmpSender's ringBuffer was
increased to stop the 'unable to dump old write' errors. Also made some small changes elsewhere that should fix future issues, particular with
MTS output.
Changes included adding support for variable InputCodec that may be set via netreceiver/vidgrind to set to H264/MJPEG. Also setting revid's lexTo
function to mjpeg.Lex in the case of an MJPEG InputCodec. Added options parameter to mts.NewEncoder function so that we can change options of the
encoder, namely whether it bases PSI interval on NAL type, or number of frames - in the case of MJPEG we based PSI interval on number of frames.
This change included a rename of IntraRefreshPeriod to MinPeriod, and the addition of the ClipDuration param. PSI are now written before IDR. Clips are no longer outputed
based on PSI but rather a time ClipDuration, where ClipDuration >= MinPeriod, however, PSI must still be at the beginning of each clip. Also created functionality to update
meta time even if we don't have a response to update.
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.
audio.go will be used for recording sound from the sound card and mic
it is like audio-netsender but it is a package instead of a command
and without the netsender.
Now that we want buffered senders (as required), the ringBuffer that was after the
lexer has been removed. Instead, we now have an ioext.multiWriterCloser to which the
lexer writes to. This then writes to the encoders, and then encoders write to each of
their own multiWriteClosers, which write to the appropriate senders. We now call
close on the first multiWriteCloser to close down the entired pipeline, as this close
call propogates through each level.
We have removed the outputClips routine as it's not required anymore to get data
from the revid ringBuffer, and have removed other things that were used by this, like
the IsRunning function.
We have also updated tests to work with these changes - they are passing.
The mtsSender now has a ringBuffer and tests have been updated accordingly. The mtsSender now uses an output routine to get data from it's ringBuffer to send.
Revid now uses ioext.multiWriteClosers for encoders to write to so that senders can be closed and therefore any output routines.
We are now using an io.multiWriter rather than the multiSender. Code has been updated inside revid.go to account for this change, and tests have also been updated accordingly. Tests for
the multiSender have been removed. A dummyMultiWriter has been written to inject our own multiWriter during testing.