我使用的是定制的德州仪器OMAP-L138基板，基本上由基于ARM9的SoC和DSP处理器组成。它连接到相机镜头。我想要做的是捕获发送到dsp处理器的实时视频流，用于H264编码，该编码通过uPP以8192字节的数据包发送。我想使用Live555提供的testH264VideoStreamer在RTSP上直播H264编码视频。我修改过的代码如下所示：
#include <liveMedia.hh>
#include <BasicUsageEnvironment.hh>
#include <GroupsockHelper.hh>
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <fcntl.h>
#include <string.h>
#include <errno.h>
#include <string.h>
#include <unistd.h> //to allow read() function
UsageEnvironment* env;
H264VideoStreamFramer* videoSource;
RTPSink* videoSink;
//-------------------------------------------------------------------------------
/* Open File Descriptor*/
int stream = open("/dev/upp", O_RDONLY);
/* Declaring a static 8 bit unsigned integer of size 8192 bytes that keeps its value between invocations */
static uint8_t buf[8192];
//------------------------------------------------------------------------------
//------------------------------------------------------------------------------
// Execute play function as a forwarding mechanism
//------------------------------------------------------------------------------
void play(); // forward
//------------------------------------------------------------------------------
// MAIN FUNCTION / ENTRY POINT
//------------------------------------------------------------------------------
int main(int argc, char** argv)
{
// Begin by setting up our live555 usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 255;
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
rtpGroupsock.multicastSendOnly(); // we're a SSM source
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
rtcpGroupsock.multicastSendOnly(); // we're a SSM source
// Create a 'H264 Video RTP' sink from the RTP 'groupsock':
OutPacketBuffer::maxSize = 1000000;
videoSink = H264VideoRTPSink::createNew(*env, &rtpGroupsock, 96);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 500; // 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
RTCPInstance* rtcp
= RTCPInstance::createNew(*env, &rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
videoSink, NULL /* we're a server */,
True /* we're a SSM source */);
// Note: This starts RTCP running automatically
/*Create RTSP SERVER*/
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
if (rtspServer == NULL)
{
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, "IPCAM @ TeReSol","UPP Buffer" ,
"Session streamed by \"testH264VideoStreamer\"",
True /*SSM*/);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
// Start the streaming:
*env << "Beginning streaming...\n";
play();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
//----------------------------------------------------------------------------------
// afterPlaying() -> Defines what to do once a buffer is streamed
//----------------------------------------------------------------------------------
void afterPlaying(void* /*clientData*/)
{
*env << "...done reading from upp buffer\n";
//videoSink->stopPlaying();
//Medium::close(videoSource);
// Note that this also closes the input file that this source read from.
// Start playing once again to get the next stream
play();
/* We don't need to close the dev as long as we're reading from it. But if we do, use: close( "/dev/upp", O_RDWR);*/
}
//----------------------------------------------------------------------------------------------
// play() Method -> Defines how to read and what to make of the input stream
//----------------------------------------------------------------------------------------------
void play()
{
/* Read nbytes of buffer (sizeof buf ) from the filedescriptor stream and assign them to address where buf is located */
read(stream, &buf, sizeof buf);
printf("Reading from UPP in to Buffer");
/*Open the input file as a 'byte-stream file source': */
ByteStreamMemoryBufferSource* buffSource
= ByteStreamMemoryBufferSource::createNew(*env, buf, sizeof buf,False/*Empty Buffer After Reading*/);
/*By passing False in the above creatNew() method means that the buffer would be read at once */
if (buffSource == NULL)
{
*env << "Unable to read from\"" << "Buffer"
<< "\" as a byte-stream source\n";
exit(1);
}
FramedSource* videoES = buffSource;
// Create a framer for the Video Elementary Stream:
videoSource = H264VideoStreamFramer::createNew(*env, videoES,False);
// Finally, start playing:
*env << "Beginning to read from UPP...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}
问题是代码虽然编译成功但我无法获得所需的输出。 VLC播放器上的RTSP流处于播放模式，但我看不到任何视频。我对此事的任何帮助表示感谢。在我的描述中，我可能会有点模糊，但我很乐意进一步解释所需的任何部分。
答案 0 :(得分：9)
好的，所以我想出了需要做什么，并且为了所有可能面临类似问题的人的利益而写作。我需要做的是修改我的testH264VideoStreamer.cpp和DeviceSource.cpp文件，使其直接从设备读取数据（在我的情况下，它是自定义的am1808板），将其存储在缓冲区并流式传输。我所做的改变是：
<强> testH264VideoStreamer.cpp
#include <liveMedia.hh>
#include <BasicUsageEnvironment.hh>
#include <GroupsockHelper.hh>
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <fcntl.h>
#include <string.h>
#include <errno.h>
#include <string.h>
#include <unistd.h> //to allow read() function
UsageEnvironment* env;
H264VideoStreamFramer* videoSource;
RTPSink* videoSink;
void play(); // forward
//-------------------------------------------------------------------------
//Entry Point -> Main FUNCTION
//-------------------------------------------------------------------------
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 255;
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
rtpGroupsock.multicastSendOnly(); // we're a SSM source
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
rtcpGroupsock.multicastSendOnly(); // we're a SSM source
// Create a 'H264 Video RTP' sink from the RTP 'groupsock':
OutPacketBuffer::maxSize = 600000;
videoSink = H264VideoRTPSink::createNew(*env, &rtpGroupsock, 96);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 1024; // 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
RTCPInstance* rtcp
= RTCPInstance::createNew(*env, &rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
videoSink, NULL /* we're a server */,
True /* we're a SSM source */);
// Note: This starts RTCP running automatically
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, "ipcamera","UPP Buffer" ,
"Session streamed by \"testH264VideoStreamer\"",
True /*SSM*/);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
// Start the streaming:
*env << "Beginning streaming...\n";
play();
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
//----------------------------------------------------------------------
//AFTER PLAY FUNCTION CALLED HERE
//----------------------------------------------------------------------
void afterPlaying(void* /*clientData*/)
{
play();
}
//------------------------------------------------------------------------
//PLAY FUNCTION ()
//------------------------------------------------------------------------
void play()
{
// Open the input file as with Device as the source:
DeviceSource* devSource
= DeviceSource::createNew(*env);
if (devSource == NULL)
{
*env << "Unable to read from\"" << "Buffer"
<< "\" as a byte-stream source\n";
exit(1);
}
FramedSource* videoES = devSource;
// Create a framer for the Video Elementary Stream:
videoSource = H264VideoStreamFramer::createNew(*env, videoES,False);
// Finally, start playing:
*env << "Beginning to read from UPP...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}
<强> DeviceSource.cpp
#include "DeviceSource.hh"
#include <GroupsockHelper.hh> // for "gettimeofday()"
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <fcntl.h>
#include <string.h>
#include <errno.h>
#include <string.h>
#include <unistd.h>
//static uint8_t *buf = (uint8_t*)malloc(102400);
static uint8_t buf[8192];
int upp_stream;
//static uint8_t *bufPtr = buf;
DeviceSource*
DeviceSource::createNew(UsageEnvironment& env)
{
return new DeviceSource(env);
}
EventTriggerId DeviceSource::eventTriggerId = 0;
unsigned DeviceSource::referenceCount = 0;
DeviceSource::DeviceSource(UsageEnvironment& env):FramedSource(env)
{
if (referenceCount == 0)
{
upp_stream = open("/dev/upp",O_RDWR);
}
++referenceCount;
if (eventTriggerId == 0)
{
eventTriggerId = envir().taskScheduler().createEventTrigger(deliverFrame0);
}
}
DeviceSource::~DeviceSource(void) {
--referenceCount;
envir().taskScheduler().deleteEventTrigger(eventTriggerId);
eventTriggerId = 0;
if (referenceCount == 0)
{
}
}
int loop_count;
void DeviceSource::doGetNextFrame()
{
//for (loop_count=0; loop_count < 13; loop_count++)
//{
read(upp_stream,buf, 8192);
//bufPtr+=8192;
//}
deliverFrame();
}
void DeviceSource::deliverFrame0(void* clientData)
{
((DeviceSource*)clientData)->deliverFrame();
}
void DeviceSource::deliverFrame()
{
if (!isCurrentlyAwaitingData()) return; // we're not ready for the data yet
u_int8_t* newFrameDataStart = (u_int8_t*) buf; //(u_int8_t*) buf; //%%% TO BE WRITTEN %%%
unsigned newFrameSize = sizeof(buf); //%%% TO BE WRITTEN %%%
// Deliver the data here:
if (newFrameSize > fMaxSize) {
fFrameSize = fMaxSize;
fNumTruncatedBytes = newFrameSize - fMaxSize;
} else {
fFrameSize = newFrameSize;
}
gettimeofday(&fPresentationTime, NULL);
memmove(fTo, newFrameDataStart, fFrameSize);
FramedSource::afterGetting(this);
}
使用这些修改编译代码后，我能够在vlc播放器上接收视频流。