Merge branch 'master' of git.linphone.org:mediastreamer2 into dev_avpf

include/mediastreamer2/mediastream.h

@@ -137,7 +137,6 @@ struct _MediaStream {
 	time_t start_time;
 	time_t last_iterate_time;
 	bool_t use_rc;
-	bool_t is_beginning;
 	bool_t owns_sessions;
 	bool_t pad[1];
 	/**

src/voip/audiostream.c

@@ -192,6 +192,7 @@ void audio_stream_prepare_sound(AudioStream *stream, MSSndCard *playcard, MSSndC
 	audio_stream_unprepare_sound(stream);
 	stream->dummy=ms_filter_new(MS_RTP_RECV_ID);
 	rtp_session_set_payload_type(stream->ms.sessions.rtp_session,0);
+	rtp_session_enable_rtcp(stream->ms.sessions.rtp_session, FALSE);
 	ms_filter_call_method(stream->dummy,MS_RTP_RECV_SET_SESSION,stream->ms.sessions.rtp_session);
 
 	if (captcard && playcard){
@@ -337,8 +338,10 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 
 	rtp_session_set_profile(rtps,profile);
 	if (rem_rtp_port>0) rtp_session_set_remote_addr_full(rtps,rem_rtp_ip,rem_rtp_port,rem_rtcp_ip,rem_rtcp_port);
-	if (rem_rtcp_port<=0){
-		rtp_session_enable_rtcp(rtps,FALSE);
+	if (rem_rtcp_port > 0) {
+		rtp_session_enable_rtcp(rtps, TRUE);
+	} else {
+		rtp_session_enable_rtcp(rtps, FALSE);
 	}
 	rtp_session_set_payload_type(rtps,payload);
 	rtp_session_set_jitter_compensation(rtps,jitt_comp);
@@ -521,7 +524,7 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 		ms_message("Setting audio encoder network bitrate to [%i] on stream [%p]",stream->ms.target_bitrate,stream);
 		ms_filter_call_method(stream->ms.encoder,MS_FILTER_SET_BITRATE,&stream->ms.target_bitrate);
 	}
-	rtp_session_set_target_upload_bandwidth(stream->ms.sessions.rtp_session, stream->ms.target_bitrate);
+	rtp_session_set_target_upload_bandwidth(rtps, stream->ms.target_bitrate);
 	ms_filter_call_method(stream->ms.encoder,MS_FILTER_SET_NCHANNELS,&nchannels);
 	ms_filter_call_method(stream->ms.decoder,MS_FILTER_SET_SAMPLE_RATE,&sample_rate);
 	ms_filter_call_method(stream->ms.decoder,MS_FILTER_SET_NCHANNELS,&nchannels);
@@ -652,7 +655,6 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 				,NULL);
 
 	stream->ms.start_time=stream->last_packet_time=ms_time(NULL);
-	stream->ms.is_beginning=TRUE;
 	stream->ms.state=MSStreamStarted;
 
 	return 0;

src/voip/mediastream.c

@@ -158,7 +158,6 @@ RtpSession * create_duplex_rtpsession(int loc_rtp_port, int loc_rtcp_port, bool_
 	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_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));
 	return rtpr;
 }
@@ -472,10 +471,6 @@ void media_stream_iterate(MediaStream *stream){
 	if (stream->ice_check_list) ice_check_list_process(stream->ice_check_list,stream->sessions.rtp_session);
 	/*we choose to update the quality indicator as much as possible, since local statistics can be computed realtime. */
 	if (stream->state==MSStreamStarted){
-		if (stream->is_beginning && (curtime-stream->start_time>15)){
-			rtp_session_set_rtcp_report_interval(stream->sessions.rtp_session,5000);
-			stream->is_beginning=FALSE;
-		}
 		if (stream->qi && curtime>stream->last_iterate_time) ms_quality_indicator_update_local(stream->qi);
 	}
 	stream->last_iterate_time=curtime;

src/voip/videostream.c

@@ -411,6 +411,11 @@ int video_stream_start (VideoStream *stream, RtpProfile *profile, const char *re
 	rtp_session_set_profile(rtps,profile);
 	if (rem_rtp_port>0)
 		rtp_session_set_remote_addr_full(rtps,rem_rtp_ip,rem_rtp_port,rem_rtcp_ip,rem_rtcp_port);
+	if (rem_rtcp_port > 0) {
+		rtp_session_enable_rtcp(rtps, TRUE);
+	} else {
+		rtp_session_enable_rtcp(rtps, FALSE);
+	}
 	rtp_session_set_payload_type(rtps,payload);
 	rtp_session_set_jitter_compensation(rtps,jitt_comp);
 
@@ -595,7 +600,6 @@ int video_stream_start (VideoStream *stream, RtpProfile *profile, const char *re
 	if (stream->ms.sessions.ticker==NULL) media_stream_start_ticker(&stream->ms);
 
 	stream->ms.start_time=ms_time(NULL);
-	stream->ms.is_beginning=TRUE;
 
 	/* attach the graphs */
 	if (stream->source)
@@ -611,6 +615,7 @@ void video_stream_prepare_video(VideoStream *stream){
 	video_stream_unprepare_video(stream);
 	stream->ms.rtprecv=ms_filter_new(MS_RTP_RECV_ID);
 	rtp_session_set_payload_type(stream->ms.sessions.rtp_session,0);
+	rtp_session_enable_rtcp(stream->ms.sessions.rtp_session, FALSE);
 	ms_filter_call_method(stream->ms.rtprecv,MS_RTP_RECV_SET_SESSION,stream->ms.sessions.rtp_session);
 	stream->ms.voidsink=ms_filter_new(MS_VOID_SINK_ID);
 	ms_filter_link(stream->ms.rtprecv,0,stream->ms.voidsink,0);