rtsp_example.c

一个简单的RTSP客户端实现源码

/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.	See the GNU Lesser General Public License for
more details.
**********/

#include "rtsp_example.h"


int main(int argc, char** argv) 
{

	char* url = argv[1];
	int socketNum = -1;
	struct timeb startTime;
	struct timeb currentTime;
	char *timeline;
	struct ResultData data;
	struct MediaAttribute Attribute;
	int Finished = -1;
	unsigned char* streamBuf;
	unsigned int video_count=0;
	unsigned int audio_count=0;
	if (argc != 2) 
	{
			fprintf(stderr,"Usage:%s <url>\n",argv[0]);
			return(0);
	}
	memset(Attribute.fConfigAsc,0,100);
	memset(Attribute.fConfigHex,0,50);
	Attribute.fVideoFrameRate	= 0;
	Attribute.fTimeIncBits = 0;
	Attribute.fVideoWidth = 0;
	Attribute.fVideoHeight = 0;
	
	
	socketNum = init_rtsp(url,&Attribute);
	if(socketNum<0) return 0;


	ftime(&startTime);
	timeline = ctime( & ( startTime.time ) );

	fprintf(stderr,"Start receive streaming....\n");
	fprintf(stderr,"Start time is: %.19s.%hu %s\n",timeline, startTime.millitm, &timeline[20]);
	streamBuf = (unsigned char *)malloc(MAX_PACKET_SIZE*sizeof(unsigned char));
	if(streamBuf == NULL)
	{
		fprintf(stderr,"alloc streamBuf failed\n");
		clearup(socketNum);
		return 0;
	}
	data.buffer = streamBuf;
	while(1)
	{
		memset(data.buffer,0,MAX_PACKET_SIZE);
		data.len = 0;
		data.fRTPPayloadFormat = 0;
		data.frtpTimestamp = 0;
	
		//Finished = tcpReadHandler(socketNum,&data);
		Finished = RTP_ReadHandler(socketNum,&data);
		if(data.fRTPPayloadFormat == 96)
		{
			video_count+=data.len;
			fprintf(stderr,"video: %d bytes,audio:%d bytes\r",video_count,audio_count); 
		}
		else if(data.fRTPPayloadFormat == 97)
		{
			audio_count+=data.len;
			fprintf(stderr,"video: %d bytes,audio:%d bytes\r",video_count,audio_count); 
		}
		if(!Finished) 
		{
			//Attribute.fVideoFrameRate = VFrameRate;
			fprintf(stderr,"MediaAttribute fVideoFrequency is %u\n",Attribute.fVideoFrequency);
			fprintf(stderr,"MediaAttribute fVideoPayloadFormat is %u\n",Attribute.fVideoPayloadFormat);
			fprintf(stderr,"MediaAttribute fConfigAsc is %s\n",Attribute.fConfigAsc);
			fprintf(stderr,"MediaAttribute fVideoFrameRate is %u\n",Attribute.fVideoFrameRate);
			fprintf(stderr,"MediaAttribute fTimeIncBits is %u\n",Attribute.fTimeIncBits);
			fprintf(stderr,"MediaAttribute fixed_vop_rate is %u\n",Attribute.fixed_vop_rate);
			fprintf(stderr,"MediaAttribute fVideoWidth is %u\n",Attribute.fVideoWidth);
			fprintf(stderr,"MediaAttribute fVideoHeight is %u\n",Attribute.fVideoHeight);
			fprintf(stderr,"MediaAttribute fAudioFrequency is %u\n",Attribute.fAudioFrequency);
			fprintf(stderr,"MediaAttribute fAudioPayloadFormat is %u\n",Attribute.fAudioPayloadFormat);
			fprintf(stderr,"MediaAttribute fTrackNum is %u\n",Attribute.fTrackNum);		
			ftime(&currentTime);
			timeline = ctime( & ( currentTime.time ) );
			fprintf(stderr,"Receive successful\n");
			fprintf(stderr,"End time is: %.19s.%hu %s\n",timeline, currentTime.millitm, &timeline[20]);
			break;
		}
	}	
	clearup(socketNum);
	return 0; 
}