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);
>
>
>
>
> _______________________________________________
> Ccrtp-devel mailing list
> [email protected]
> http://lists.gnu.org/mailman/listinfo/ccrtp-devel
begin:vcard
fn:David Sugar
n:Sugar;David
org:GNU Telephony
email;internet:[EMAIL PROTECTED]
tel;work:+1 201 215 2609
url:http://www.gnutelephony.org
version:2.1
end:vcard
_______________________________________________
Ccrtp-devel mailing list
[email protected]
http://lists.gnu.org/mailman/listinfo/ccrtp-devel
