Hello,
I'm trying to send a RTP stream.
I have negotiated and sdp then created an stream and retrieved de port with pjmedia_stream_create
My internal audio is originaly defined as: baudrate = 44100 samples per frame = 554 bits per sample 16
So I've create a resample por with:
pjmedia_resample_port_create( pool, port,44100, PJMEDIA_RESAMPLE_USE_LINEARPJMEDIA_RESAMPLE_USE_SMALL_FILTER|,&resample_port));
and sent the audio using
frame.size=554*2; frame.buf=bufPt; (bufPt is my frame with 554 samples) rc=pjmedia_port_put_frame(resample_port,&frame);
Everything lucks ok but in the other side the audio is received distorted.
I've started to make different test and I have detect that I can't send my frame with 554 samples, that I have to send 20ms. (that is 882 samples)
What I'm doing wrong? I couldn't send the frame size that I want? If actually I have to send a predefined frame size (or time size), how I get that size/time?
Thanks!!!
Best regards.
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