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

[David Sugar]

There is a "sendImmediate" method you can use instead of putData, and
this is exactly what it is designed for :).  Bayonne uses it for fast
proxy mode.

Incidently, the latest common c++ release also has some changes to
TimerPort which should improve micro-timing and accuracy since it now
also uses the monotanic clock source.

Rob Prowel wrote:
> 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);
> 
> 
> 
> 
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