Hi,
I'd like to use pjsip for our current project; I am trying to set up a SIP server that receives phone calls, forwards the decoded raw audio data to a pre-existing software module written in c++, and sends back to the caller some audio generated by the same c++ module.
I have successfully compiled the example applications, and I see that the functionality I need it's all there. The sipecho.c example showed me that it's possible to quickly build the answering part of the application, the playsine shows that it's possible to attach a traditional audio callback to the output channel, and simple_ua shows that I can get some useful information from the media stream of the caller.
Still, I am a bit struggling to understand how to put all the pieces together.
The goal is to have an application that whenever there is a call can instantiate and keep in the memory a structure with this interface (this is a fa?ade C interface)
struct OurAudioProcessingLibrary {
void process(float** samplesIn, float** samplesOut, int numSamples);
};
but I haven't found a way until today. This signature can be thought as an audio processing black box, or an audio fx. I ask if somebody can put me in the right direction, or if someone is available to work for us during the development of this software.
Thank you
Alessandro Saccoia
www.dinahmoe.com
