view src/mediastreamer/mediastream.c @ 12024:e67993da8a22

[gaim-migrate @ 14317] I strongly suspect CruiseControl is going to yell at me for this. A voice chat API, GUI + mediastreamer. This is what I'm using for Google Talk. This doesn't actually do anything at all. There's no code in the Jabber plugin yet to use this API (although it Works For Me). All it will do is compile and link. If you're lucky. To build this, you should install oRTP from Linphone, Speex and iLBC (also from linphone, I believe). To not build this, ./configure --disable-vv. Most of the configure.ac and Makefile.am hackery was lifted right out of Linphone with a few modifications. It seems to work if you have everything installed or if you --disable-vv. I haven't really tested not having everything installed and not --disabling-vv. It's kinda funky to include all of mediastreamer in the source tree like this, but linphone doesn't build it as a separate library. I'll probably wind up writing them a patch to build it as a .so so we can link it dynamically instead. This code certainly isn't finished. It'll adapt as I progress on the Google code, but it's certainly of more use here in CVS than in my personal tree. committer: Tailor Script <[e-mail]>

/*
  The mediastreamer library aims at providing modular media processing and I/O
	for linphone, but also for any telephony application.
  Copyright (C) 2001  Simon MORLAT [e-mail]
  										
  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.

  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.

  You should have received a copy of the GNU Lesser General Public
  License along with this library; if not, write to the Free Software
  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
*/

#include "mediastream.h"

#include <signal.h>
#include <sys/types.h>
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>

static int cond=1;

void stop_handler(int signum)
{
	cond--;
	if (cond<0) exit(-1);
}

void parse_addr(gchar *addr, char **ip, int *port)
{
	char *semicolon;
	gint iplen;
	
	*ip=NULL;
	*port=0;
	semicolon=strchr(addr,':');
	if (semicolon==NULL) return;
	iplen=semicolon-addr;
	*ip=g_malloc(iplen+1);
	strncpy(*ip,addr,iplen);
	(*ip)[iplen]='\0';
	*port=atoi(semicolon+1);
}

char *usage="mediastream --local <port> --remote <ip:port> --payload <payload type>\n";
void run_media_streams(gint localport, gchar *remote_ip, gint remoteport, gint payload);

int main(int argc, char * argv[])
{
	gint i;
	gint localport=0,remoteport=0,payload=0;
	gchar *ip;
	gchar *tmp;
	
	/*create the rtp session */
	ortp_init();
	ortp_set_debug_file("oRTP",NULL);
	rtp_profile_set_payload(&av_profile,115,&lpc1015);
	rtp_profile_set_payload(&av_profile,110,&speex_nb);
	
	if (argc<4) {
		printf(usage);
		return -1;
	}
	for (i=1;i<argc;i++){
		if (strcmp(argv[i],"--local")==0){
			i++;
			localport=atoi(argv[i]);
		}else if (strcmp(argv[i],"--remote")==0){
			i++;
			parse_addr(argv[i],&ip,&remoteport);
			if (ip==NULL) {
				printf(usage);
				return -1;
			}
			printf("Remote addr: ip=%s port=%i\n",ip,remoteport);
		}else if (strcmp(argv[i],"--payload")==0){
			i++;
			payload=atoi(argv[i]);
		}
	}
	tmp=getenv("defcard");
	if (tmp!=NULL) audio_stream_set_default_card(atoi(tmp));
	run_media_streams(localport,ip,remoteport,payload);
	return 0;
}

void run_media_streams(gint localport, gchar *remote_ip, gint remoteport, gint payload)
{
	AudioStream *audio;
	ms_init();
	ms_speex_codec_init();
	signal(SIGINT,stop_handler);
	
	audio=audio_stream_start(&av_profile,localport,remote_ip,remoteport,payload,250);
	while(cond)
	{
		/* sleep until we receive SIGINT */
		sleep(1);
	}
	
	printf("stoping all...\n");
	
	audio_stream_stop(audio);
}