Hi,
I am currently working on a project which needs a RTSP server running under
android and iOS.
I am able to play an audio file, but the audio is jerky. It's a simple mp3
file.
I didn't find a lot of documentation about live555, and if there is a way
to improve performances.
I don't think android is in fault because I built it in iOS too, and got
some lags too.
I've attached my code, I hope you will see what I am missing. It's based on
the example from live555 for RTSP streaming.
Audric
#include <jni.h>
#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include "BasicUsageEnvironment.hh"
#include <android/log.h>
#define LOG_TAG "live555_wrapper"
#define ALOG(...) __android_log_print(ANDROID_LOG_INFO,LOG_TAG,__VA_ARGS__)
// To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
#define STREAM_USING_ADUS 1
// To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
// (For more information about ADUs and interleaving,
// see <http://www.live555.com/rtp-mp3/>)
// To stream using "source-specific multicast" (SSM), uncomment the following:
//#define USE_SSM 1
#ifdef USE_SSM
Boolean const isSSM = True;
#else
Boolean const isSSM = False;
#endif
#ifdef __cplusplus
extern "C" {
#endif
// A structure to hold the state of the current session.
// It is used in the "afterPlaying()" function to clean up the session.
struct sessionState_t {
FramedSource* source;
RTPSink* sink;
RTCPInstance* rtcpInstance;
Groupsock* rtpGroupsock;
Groupsock* rtcpGroupsock;
} sessionState;
void afterPlaying(void* /*clientData*/) {
ALOG("...done streaming\n");
sessionState.sink->stopPlaying();
// End this loop by closing the current source:
Medium::close(sessionState.source);
}
JNIEXPORT jint JNICALL Java_com_athome_service_StreamerService_nativeAddMedia(
JNIEnv * jnienv, jobject obj, jlong live555env,
jlong jrtspServer, jstring streamName, jstring fileName) {
UsageEnvironment* env = (UsageEnvironment*) live555env;
RTSPServer* rtspServer = (RTSPServer*) jrtspServer;
if(rtspServer != NULL) {
const char *nativeStreamName = (jnienv)->GetStringUTFChars(streamName, 0);
const char *nativefilename = (jnienv)->GetStringUTFChars(fileName, 0);
ServerMediaSession* sms = ServerMediaSession::createNew(*env, nativeStreamName, nativefilename,
"Session streamed by \"testMP3Streamer\"", isSSM);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*sessionState.sink, sessionState.rtcpInstance));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
ALOG("Play this stream using the URL %s", url);
delete[] url;
// Open the file as a 'MP3 file source':
sessionState.source = MP3FileSource::createNew(*env, nativefilename);
if (sessionState.source == NULL) {
ALOG("Unable to open '%s' as a MP3 file source\n" , nativefilename);
return 1;
}
#ifdef STREAM_USING_ADUS
// Add a filter that converts the source MP3s to ADUs:
sessionState.source
= ADUFromMP3Source::createNew(*env, sessionState.source);
if (sessionState.source == NULL) {
ALOG("Unable to create a MP3->ADU filter for the source\n");
return 1;
}
/*#ifdef INTERLEAVE_ADUS
// Add another filter that interleaves the ADUs before packetizing them:
unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own order...
unsigned const interleaveCycleSize
= (sizeof interleaveCycle)/(sizeof (unsigned char));
Interleaving interleaving(interleaveCycleSize, interleaveCycle);
sessionState.source
= MP3ADUinterleaver::createNew(*env, interleaving, sessionState.source);
if (sessionState.source == NULL) {
ALOG("Unable to create an ADU interleaving filter for the source\n");
return 1;
}
#endif
*/#endif
// Finally, start the streaming:
ALOG("Beginning streaming...\n");
sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL);
//(jnienv)->ReleaseStringUTFChars(streamName, nativeStreamName);
//(jnienv)->ReleaseStringUTFChars(fileName, nativefilename);
return 0;
}
else {
ALOG("Cannot add media: rtspServer is NULL!...\n");
return 1;
}
}
JNIEXPORT jlong JNICALL Java_com_athome_service_StreamerService_nativeInitEnv(JNIEnv * jnienv, jobject obj) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
return (jlong) env;
}
JNIEXPORT jlong JNICALL Java_com_athome_service_StreamerService_nativeInitRtspServer(JNIEnv * jnienv, jobject obj, jlong live555env) {
UsageEnvironment* env = (UsageEnvironment*) live555env;
// Create 'groupsocks' for RTP and RTCP:
char const* destinationAddressStr
#ifdef USE_SSM
= "232.255.42.42";
#else
= "239.255.42.42";
// Note: This is a multicast address. If you wish to stream using
// unicast instead, then replace this string with the unicast address
// of the (single) destination. (You may also need to make a similar
// change to the receiver program.)
#endif
const unsigned short rtpPortNum = 6666;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 1; // low, in case routers don't admin scope
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtcpGroupsock
= new Groupsock(*env, destinationAddress, rtcpPort, ttl);
#ifdef USE_SSM
sessionState.rtpGroupsock->multicastSendOnly();
sessionState.rtcpGroupsock->multicastSendOnly();
#endif
// Create a 'MP3 RTP' sink from the RTP 'groupsock':
#ifdef STREAM_USING_ADUS
unsigned char rtpPayloadFormat = 96; // A dynamic payload format code
sessionState.sink
= MP3ADURTPSink::createNew(*env, sessionState.rtpGroupsock,
rtpPayloadFormat);
#else
sessionState.sink
= MPEG1or2AudioRTPSink::createNew(*env, sessionState.rtpGroupsock);
#endif
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 320; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
sessionState.rtcpInstance
= RTCPInstance::createNew(*env, sessionState.rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
sessionState.sink, NULL /* we're a server */,
isSSM);
RTSPServer* rtspServer = RTSPServer::createNew(*env, 55443);
if (rtspServer == NULL) {
ALOG("Failed to create RTSP server: %s", env->getResultMsg());
}
else
ALOG("RTSP Server creation: OK: %p", rtspServer);
return (jlong) rtspServer;
}
JNIEXPORT void JNICALL Java_com_athome_service_StreamerService_nativeStartServe(JNIEnv * jnienv, jobject obj, jlong live555env) {
UsageEnvironment* env = (UsageEnvironment* ) live555env;
ALOG("Starting serving rtsp....");
env->taskScheduler().doEventLoop(); // does not return
}
#ifdef __cplusplus
}
#endif_______________________________________________
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