fix leak of RtpTransport when call is updated/paused. Fix reporting of...

fix leak of RtpTransport when call is updated/paused. Fix reporting of bandwidth, which was displayed even if the stream was inactive.

coreapi/linphonecall.c

@@ -1550,6 +1550,8 @@ void linphone_call_init_audio_stream(LinphoneCall *call){
 	AudioStream *audiostream;
 	const char *location;
 	int dscp;
+	RtpTransport *meta_rtp=NULL;
+	RtpTransport *meta_rtcp=NULL;
 
 	if (call->audiostream != NULL) return;
 	if (call->sessions[0].rtp_session==NULL){
@@ -1600,12 +1602,11 @@ void linphone_call_init_audio_stream(LinphoneCall *call){
 	}
 
 	audio_stream_set_features(audiostream,linphone_core_get_audio_features(lc));
-
-	if (lc->rtptf){
+	rtp_session_get_transports(audiostream->ms.sessions.rtp_session,&meta_rtp,&meta_rtcp);
+	if (lc->rtptf && (meta_rtp==NULL && meta_rtcp==NULL)){
+		/*the transport just need to be created once, then they are kept into the RtpSession, which is the same for the entire call duration.*/
 		RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->media_ports[0].rtp_port);
 		RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->media_ports[0].rtcp_port);
-		RtpTransport *meta_rtp;
-		RtpTransport *meta_rtcp;
 		meta_rtp_transport_new(&meta_rtp,TRUE,artp, 0);
 		meta_rtp_transport_new(&meta_rtcp,FALSE,artcp, 0);
 		rtp_session_set_transports(audiostream->ms.sessions.rtp_session,meta_rtp,meta_rtcp);
@@ -2781,14 +2782,17 @@ void linphone_call_stop_recording(LinphoneCall *call){
 **/
 
 static void report_bandwidth(LinphoneCall *call, MediaStream *as, MediaStream *vs){
-	call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth=(as!=NULL) ? (media_stream_get_down_bw(as)*1e-3) : 0;
-	call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth=(as!=NULL) ? (media_stream_get_up_bw(as)*1e-3) : 0;
-	call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth=(vs!=NULL) ? (media_stream_get_down_bw(vs)*1e-3) : 0;
-	call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth=(vs!=NULL) ? (media_stream_get_up_bw(vs)*1e-3) : 0;
-	call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_download_bandwidth=(as!=NULL) ? (media_stream_get_rtcp_down_bw(as)*1e-3) : 0;
-	call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_upload_bandwidth=(as!=NULL) ? (media_stream_get_rtcp_up_bw(as)*1e-3) : 0;
-	call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_download_bandwidth=(vs!=NULL) ? (media_stream_get_rtcp_down_bw(vs)*1e-3) : 0;
-	call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_upload_bandwidth=(vs!=NULL) ? (media_stream_get_rtcp_up_bw(vs)*1e-3) : 0;
+	bool_t as_active =  as ? (media_stream_get_state(as) == MSStreamStarted) : FALSE;
+	bool_t vs_active =  vs ? (media_stream_get_state(vs) == MSStreamStarted) : FALSE;
+	
+	call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth=(as_active) ? (media_stream_get_down_bw(as)*1e-3) : 0;
+	call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth=(as_active) ? (media_stream_get_up_bw(as)*1e-3) : 0;
+	call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth=(vs_active) ? (media_stream_get_down_bw(vs)*1e-3) : 0;
+	call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth=(vs_active) ? (media_stream_get_up_bw(vs)*1e-3) : 0;
+	call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_download_bandwidth=(as_active) ? (media_stream_get_rtcp_down_bw(as)*1e-3) : 0;
+	call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_upload_bandwidth=as_active) ? (media_stream_get_rtcp_up_bw(as)*1e-3) : 0;
+	call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_download_bandwidth=(vs_active) ? (media_stream_get_rtcp_down_bw(vs)*1e-3) : 0;
+	call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_upload_bandwidth=(vs_active) ? (media_stream_get_rtcp_up_bw(vs)*1e-3) : 0;
 
 	ms_message("Bandwidth usage for call [%p]: audio[ rtp]=[d=%.1f,u=%.1f], video[ rtp]=[d=%.1f,u=%.1f] kbit/sec",
 		call,