Only use payload size within the receive/send interval for bitrate probing.

webrtc/modules/congestion_controller/probe_bitrate_estimator.h

Index: webrtc/modules/congestion_controller/probe_bitrate_estimator.h
diff --git a/webrtc/modules/congestion_controller/probe_bitrate_estimator.h b/webrtc/modules/congestion_controller/probe_bitrate_estimator.h
index 7f0c4569c97f039976e822d94887987d66e8e994..6c04020b4310db5c31603cf7e7aa34f1f8ac460d 100644
--- a/webrtc/modules/congestion_controller/probe_bitrate_estimator.h
+++ b/webrtc/modules/congestion_controller/probe_bitrate_estimator.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
 #define WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
 
-#include <map>
 #include <limits>
+#include <map>
+#include <set>
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
@@ -40,7 +41,11 @@ class ProbeBitrateEstimator {
 
  private:
   struct AggregatedCluster {
-    int num_probes = 0;
+    // Since sending padding can result in multiple packets being sent (all

[danilchap, 2016/08/19 14:08:31]

same send time is common, but not guaranteed (it is calculated per packet and
just happen to be close enough to look same).

+    // with the same send time), we keep track of how many unique send times
+    // we have seen to make sure we don't count the same request to send padding
+    // as separate packets in the probing cluster.
+    std::set<int64_t> send_times_ms;
     int64_t first_send_ms = std::numeric_limits<int64_t>::max();
     int64_t last_send_ms = 0;
     int64_t first_receive_ms = std::numeric_limits<int64_t>::max();