[Ccrtp-devel] using ccrtp without internal threading or timers

[Rob Prowel]

Hi.

I've got a hardware MPEG4 encoder that generates MPEG4 video and u-law 
PCM audio.  The kernel driver returns the frames as a multiplexed stream 
of data blocks/frames where the intervals between frames are governed by 
the driver and the read() syscall is blocking until a frame arrives.  I 
need to separate the audio and video into two distinct RTP sessions and 
transmit them in as near to real-time as is possible.  Anything more 
than 200ms delay between when the event happens and when it is perceived 
by the operator on the other end of the wifi link will be unacceptable 
and the frame should then be dropped rather than presented.

I believe it would be overly complex to rely on the threading and timers 
of ccRTP to setup and manage message delivery when the frame producer 
already regulates the interval of frames to be sent.  If I have to 
buffer a few audio packets initially and put up with 50ms of a/v sync 
drift then I don't have a problem with that.

I simply don't want to have any RTP managed threading or timing going on 
since the frame producer already governs that for me.  When I send a 
timestamped frame to putData() I want it to go out immediately.  The 
available documentation isn't explicit enough to answer my questions.  
Will the following work for my task?  Please CC me in responses.


// --------------------------------
CTR1472_AVStream() stream;                                // create MPEG 
frame source or producer

RTPSession
   video(InetAddress("localhost"), 4500, 1000.0/15.0);   // 15fps, but 
not controlled by RTP
RTPSession
   audio(InetAddress("localhost"), 4510, 20.0);          // 20ms 
packets of audio

InetAddress ad("localhost");
video.SetPayloadType(StaticPayload(sptH263));             // cheating on 
MPEG4 type
video.addDestination(ad);
video.setMaxSendSegmentSize(500);                         // small size 
for benefit of wifi
audio.SetPayloadType(StaticPayload(sptPCMU));
audio.addDestination(ad);                  
audio.setMaxSendSegmentSize(500);                         // small size 
for benefit of wifi

video.start(); // is this necessary ???  I don't want to use a run() method
audio.start(); // is this necessary ???  I don't want to use a run() method

do {
   char* p;                                              // pointer to 
frame data
   int length=0;                                         // octets 
returned in current frame
   AVStreamType stype;                                   // type of 
frame returned
   float frametime=0.0f;                                 // timestamp 
of generated frame

   stream.readFrame(&p, length, stype, frametime);       // blocking 
read of next frame
   if (length)
       switch(stype) {
           MPEG4_Iframe ... MPEG4_Bframe:
           video->putData((int)(frametime*1000.0f)%1000000000, p, length);
           break;

           mulaw_PCM:
           audio->putData((int)(frametime*1000.0f)%1000000000, p, length);
           default:
           break;
      };

} while (!done);