[Live-devel] Capturing audio from microphone

[Ross Finlayson]

>                         sessionStateS.audioSink = 
> SimpleRTPSink::createNew(*env, &rtpGroupsock, RTPpayloadType_obj, 
> rtpTimestampFrequency_obj, rtpPayloadFormatName_obj, 
> rtpPayloadFormatName_obj, 1, false, true);

The two occurrences of "rtpPayloadFormatName_obj" are incorrect.  This call 
should instead be:
         sessionStateS.audioSink
         = SimpleRTPSink::createNew(*env, &rtpGroupsock, RTPpayloadType_obj,
                 samplingFrequency, "audio", "L16", numChannels);
where
         - RTPpayloadType_obj is a dynamic value (i.e., >= 96)

Apart from that, your code looks correct.  Note that you should really have 
been using the "testWAVAudioStreamer" test program as a starting point, as 
that code already does most of what you want (only the input PCM audio 
source is different).


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>