RTP (streaming & collaboration)

RTP extension contains inputs and outputs to send/receive audio to/from the network. You can use it to create collaborative VR app (spatialized voice chat), or stream audio from server to clients.

RTP extension doc can be found here.

  1. Stream and listen to microphone

Microphone stream and listen

The following code captures the microphone, sends it to localhost, reads it back and outputs final audio on the soundcard. With little modifications, it is possible to create collaborative audio application.

Please note this sample doesn’t handle errors for code clarity.

[...]

enum:int{producerIn, networkIn=1, networkOut=2, audioOut};

AudiostackContext context;
context.setLicenseKeyFromFile("LICENSE.aslc");

RtpInterface::Load(context.impl);

L16Codec codec;

context.createInput(producerIn,WindowsCoreAudioInput);	
context.setParameter("source/1/port",(unsigned int)9040);	// Input will read audio on port 9040

context.createOutput(networkOut,RtpOutput,codec.serialize());

context.connect(producerIn,networkOut);

//

context.createInput(networkIn,RtpInput,codec.serialize());

context.createOutput(audioOut,WindowsCoreAudioOutput,false);
context.setParameter("listener/2/port",(unsigned int)9040);
context.setParameter("listener/2/ip_address","127.0.0.1");	// Output will send audio to remote host 127.0.0.1:9040

context.connect(networkIn,audioOut);

//

context.play();

char c;
std::cout<<"Press q to quit."<<std::endl;
do{
	std::cin>>c;
}while(c!='q');

context.stop();

[...]