Fix compilation warnings.

src/voip/audiostream.c

@@ -618,8 +618,8 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 		stream->dtmfgen=ms_filter_new(MS_DTMF_GEN_ID);
 	else
 		stream->dtmfgen=NULL;
-	rtp_session_signal_connect(rtps,"telephone-event",(RtpCallback)on_dtmf_received,(unsigned long)stream);
-	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)mediastream_payload_type_changed,(unsigned long)&stream->ms);
+	rtp_session_signal_connect(rtps,"telephone-event",(RtpCallback)on_dtmf_received,stream);
+	rtp_session_signal_connect(rtps,"payload_type_changed",(RtpCallback)mediastream_payload_type_changed,&stream->ms);
 	
 	if (stream->ms.state==MSStreamPreparing){
 		/*we were using the dummy preload graph, destroy it but keep sound filters unless no soundcard is given*/

src/voip/mediastream.c

@@ -159,8 +159,8 @@ RtpSession * create_duplex_rtpsession(int loc_rtp_port, int loc_rtcp_port, bool_
 	rtp_session_enable_adaptive_jitter_compensation(rtpr, TRUE);
 	rtp_session_set_symmetric_rtp(rtpr, TRUE);
 	rtp_session_set_local_addr(rtpr, ipv6 ? "::" : "0.0.0.0", loc_rtp_port, loc_rtcp_port);
-	rtp_session_signal_connect(rtpr, "timestamp_jump", (RtpCallback)rtp_session_resync, (long)NULL);
-	rtp_session_signal_connect(rtpr, "ssrc_changed", (RtpCallback)rtp_session_resync, (long)NULL);
+	rtp_session_signal_connect(rtpr, "timestamp_jump", (RtpCallback)rtp_session_resync, NULL);
+	rtp_session_signal_connect(rtpr, "ssrc_changed", (RtpCallback)rtp_session_resync, NULL);
 	rtp_session_set_ssrc_changed_threshold(rtpr, 0);
 	rtp_session_set_rtcp_report_interval(rtpr, 2500);	/* At the beginning of the session send more reports. */
 	disable_checksums(rtp_session_get_rtp_socket(rtpr));

src/voip/videostream.c

@@ -641,7 +641,7 @@ int video_stream_start_with_source (VideoStream *stream, RtpProfile *profile, co
 	rtp_session_set_jitter_compensation(rtps,jitt_comp);
 
 	rtp_session_signal_connect(stream->ms.sessions.rtp_session,"payload_type_changed",
-			(RtpCallback)video_stream_payload_type_changed,(unsigned long)&stream->ms);
+			(RtpCallback)video_stream_payload_type_changed,&stream->ms);
 
 	rtp_session_get_jitter_buffer_params(stream->ms.sessions.rtp_session,&jbp);
 	jbp.max_packets=1000;//needed for high resolution video

tools/bench.c

@@ -90,8 +90,8 @@ RtpSession *create_duplex_rtpsession(int locport){
 	rtp_session_enable_adaptive_jitter_compensation(rtpr,FALSE);
 	rtp_session_set_symmetric_rtp(rtpr,TRUE);
 	rtp_session_set_local_addr(rtpr,"0.0.0.0",locport,locport+1);
-	rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)rtp_session_resync,(long)NULL);
-	rtp_session_signal_connect(rtpr,"ssrc_changed",(RtpCallback)rtp_session_resync,(long)NULL);
+	rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)rtp_session_resync,NULL);
+	rtp_session_signal_connect(rtpr,"ssrc_changed",(RtpCallback)rtp_session_resync,NULL);
 	return rtpr;
 }