Creating a livesource audio class for live555

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C++

ffmpeg

Hi
I am trying to setup live555 to stream rtsp with an ondemandserver from a microphone, but I am not sure how to do so. I have attempted to do this by overriding OnDemandServerMediaSubsession and FramedSource, but I am running into issues where I hear a bit of sound for half a second and then quiet, in VLC the Messages say "buffer arrived way too early"...

I am encoding using ffmpeg to encode the audio as mp2.

I have attached what I am doing here:

#ifndef _FRAMED_SOURCE_HH
#include "FramedSource.hh"
#include "ImageTransfer.h"
#endif

class MP2DeviceSource : public FramedSource {
public:
	static MP2DeviceSource* createNew(UsageEnvironment& env, unsigned int stream_id,
						AudioTransfer * audioTransfer);

public:
	EventTriggerId eventTriggerId;

protected:
	MP2DeviceSource(UsageEnvironment& env,
				ImageTransfer * imageTransfer );
	virtual ~MP2DeviceSource();
private:
	virtual void doGetNextFrame();

private:
	static void deliverFrame0(void* clientData);
	void deliverFrame();

private:
	
	AudioTransfer * audioTx;

};

#include "MP2DeviceSource.h"

MP2DeviceSource* 
MP2DeviceSource::createNew(UsageEnvironment& env, unsigned int stream_id,
				AudioTransfer * audioTransfer)
{
	return new MaxMP2DeviceSource(env, audioTransfer);
}

MP2DeviceSource::MP2DeviceSource(UsageEnvironment& env, AudioTransfer * audioTransfer) 
	: FramedSource(env), audioTx(audioTransfer)
{
	if (eventTriggerId == 0)
		eventTriggerId = envir().taskScheduler().createEventTrigger(deliverFrame0);
}

MP2DeviceSource::~MP2DeviceSource()
{
	envir().taskScheduler().deleteEventTrigger(eventTriggerId);
	eventTriggerId = 0;
}

void MP2DeviceSource::doGetNextFrame()
{
	deliverFrame();
}

void MP2DeviceSource::deliverFrame0(void *clientData)
{
	((MP2DeviceSource*)clientData)->deliverFrame();
}

   static const unsigned __int64 epoch = 116444736000000000;
   int
   gettimeofday(struct timeval * tp, struct timezone * tzp)
   {
       FILETIME    file_time;
       SYSTEMTIME  system_time;
       ULARGE_INTEGER ularge;
   
       GetSystemTime(&system_time);
       SystemTimeToFileTime(&system_time, &file_time);
       ularge.LowPart = file_time.dwLowDateTime;
       ularge.HighPart = file_time.dwHighDateTime;
   
       tp->tv_sec = (long) ((ularge.QuadPart - epoch) / 10000000L);
       tp->tv_usec = (long) (system_time.wMilliseconds * 1000);
   
       return 0;
  }


void MP2DeviceSource::deliverFrame()
{

gettimeofday(&fPresentationTime, NULL);

    audioTx->GetMP2Image( &fTo, &fFrameSize );
    fDurationInMicroseconds = 26000;
	
	FramedSource::afterGetting(this);
}

#ifndef _ON_DEMAND_SERVER_MEDIA_SUBSESSION_HH
#include "OnDemandServerMediaSubsession.hh"
#endif

class MP2AudioMediaSubsession: public OnDemandServerMediaSubsession {
public:
   static MP2AudioMediaSubsession* createNew(UsageEnvironment& evn, 
                                                unsigned int sid, 
                                                Boolean requestFirstSource,
 			                        AudioTransfer * audioTransfer);
protected:
   MP2AudioMediaSubsession(UsageEnvironment& env, 
                           Boolean reuseFirstSource, 
 			   AudioTransfer * audioTransfer);

   virtual ~MP2AudioMediaSubsession();

protected:
   virtual FramedSource* createNewStreamSource(unsigned clientSessionId,            
  						unsigned& estBitrate);

   virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupSock, 
    			             unsigned char rtpPayloadTypeIfDynamic, 
				     FramedSource* inputSource);

protected:
  unsigned int id;
  AudioTransfer * audioTx;
};

#include "MP2MediaSubsession.h"
#include "MP2DeviceSource.h"
#include "MPEG1or2AudioRTPSink.hh"
#include "MPEG1or2AudioStreamFramer.hh"


MP2AudioMediaSubsession* 
MP2AudioMediaSubsession::createNew(UsageEnvironment& env,
					Boolean reuseFirstSource, 
					AudioTransfer * audioTransfer)
{
	return new MP2AudioMediaSubsession(env, reuseFirstSource, imageTransfer );
}

MP2AudioMediaSubsession::MP2AudioMediaSubsession(UsageEnvironment& env, 
							Boolean reuseFirstSource,
							AudioTransfer * audioTransfer)
	: OnDemandServerMediaSubsession(env, reuseFirstSource), audioTx(audioTransfer) 
{
}
FramedSource* MP2AudioMediaSubsession::createNewStreamSource(unsigned clientSessionId,
								unsigned &estBitrate)
{
   estBitrate = 44100;
   MP2DeviceSource *source =  MP2DeviceSource::createNew(envir(), id, audioTx);
   return MPEG1or2AudioStreamFramer::createNew(envir(), source );
}

RTPSink* MP2AudioMediaSubsession::createNewRTPSink(Groupsock* rtpGroupSock,
						unsigned char rtpPayloadTypeIfDynamic,
						FramedSource* inputSource)
{
	return MPEG1or2AudioRTPSink::createNew( envir(), rtpGroupSock );
}

MP2AudioMediaSubsession::~MP2AudioMediaSubsession()
{
}