Added log messages for important call setup events: first packet sent/received

webrtc/modules/video_coding/jitter_buffer.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/modules/video_coding/jitter_buffer.h"
@@ skipping 1225 matching lines @@
   } else {
     TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", frame.TimeStamp(),
                             "DeltaComplete");
   }
 
   // Update receive statistics. We count all layers, thus when you use layers
   // adding all key and delta frames might differ from frame count.
   if (frame.IsSessionComplete()) {
     if (frame.FrameType() == kVideoFrameKey) {
       ++receive_statistics_.key_frames;
+      if (receive_statistics_.key_frames == 1) {
+        LOG(LS_INFO) << "Received first complete key frame";
+      }
     } else {
       ++receive_statistics_.delta_frames;
     }
+
     if (stats_callback_ != NULL)
       stats_callback_->OnFrameCountsUpdated(receive_statistics_);
   }
 }
 
 void VCMJitterBuffer::UpdateAveragePacketsPerFrame(int current_number_packets) {
   if (frame_counter_ > kFastConvergeThreshold) {
     average_packets_per_frame_ =
         average_packets_per_frame_ * (1 - kNormalConvergeMultiplier) +
         current_number_packets * kNormalConvergeMultiplier;
@@ skipping 78 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc