Add logs when no RTCP RR has been received for three regular RTCP intervals.
BUG=1267
TEST=Unittest added.
Review URL: https://webrtc-codereview.appspot.com/1019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 920a4fd..60ad98d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -8,20 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "rtp_rtcp_impl.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+
+#include <cassert>
+#include <string.h>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
+#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/trace.h"
#ifdef MATLAB
-#include "../test/BWEStandAlone/MatlabPlot.h"
+#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
extern MatlabEngine eng; // global variable defined elsewhere
#endif
-#include <string.h> //memcpy
-#include <cassert> //assert
-
-#include "common_types.h"
-#include "rtp_receiver_audio.h"
-#include "rtp_receiver_video.h"
-#include "trace.h"
// local for this file
namespace {
@@ -228,6 +230,19 @@
// No own rtt calculation or set rtt, use default value.
max_rtt = kDefaultRtt;
}
+
+ // Verify receiver reports are delivered and the reported sequence number is
+ // increasing.
+ if (_rtcpSender.Sending()) {
+ int64_t rtcp_interval = RtcpReportInterval();
+ if (_rtcpReceiver.RtcpRrTimeout(rtcp_interval)) {
+ LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
+ } else if (_rtcpReceiver.RtcpRrSequenceNumberTimeout(rtcp_interval)) {
+ LOG_F(LS_WARNING) <<
+ "Timeout: No increase in RTCP RR extended highest sequence number.";
+ }
+ }
+
if (remote_bitrate_) {
// TODO(mflodman) Remove this and let this be propagated by CallStats.
remote_bitrate_->SetRtt(max_rtt);
@@ -2037,4 +2052,12 @@
TMMBRSet*& boundingSet) {
return _rtcpReceiver.BoundingSet(tmmbrOwner, boundingSet);
}
+
+int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
+ if (_audio)
+ return RTCP_INTERVAL_AUDIO_MS;
+ else
+ return RTCP_INTERVAL_VIDEO_MS;
+}
+
} // namespace webrtc
