Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead.

webrtc/video/send_statistics_proxy_unittest.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 270 matching lines @@
     int avg_delay_ms = ssrc;
     int max_delay_ms = ssrc + 1;
     observer->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
     expected_.substreams[ssrc].avg_delay_ms = avg_delay_ms;
     expected_.substreams[ssrc].max_delay_ms = max_delay_ms;
   }
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
   ExpectEqual(expected_, stats);
 }
 
+TEST_F(SendStatisticsProxyTest, OnEncodedFrame) {
+  const int kEncodeTimeMs = 11;
+  statistics_proxy_->OnEncodedFrame(kEncodeTimeMs);
+
+  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(kEncodeTimeMs, stats.avg_encode_time_ms);

[pbos-webrtc, 2015/12/07 10:55:02]

If you wanna, add two samples and make sure the avg is somewhere in between.

+}
+
 TEST_F(SendStatisticsProxyTest, NoSubstreams) {
   uint32_t excluded_ssrc =
       std::max(
           *std::max_element(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end()),
           *std::max_element(config_.rtp.rtx.ssrcs.begin(),
                             config_.rtp.rtx.ssrcs.end())) +
       1;
   // From RtcpStatisticsCallback.
   RtcpStatistics rtcp_stats;
   RtcpStatisticsCallback* rtcp_callback = statistics_proxy_.get();
@@ skipping 96 matching lines @@
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
   EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps),
             stats.substreams[config_.rtp.ssrcs[0]].total_bitrate_bps);
   EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps),
             stats.substreams[config_.rtp.ssrcs[0]].retransmit_bitrate_bps);
   EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].total_bitrate_bps);
   EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].retransmit_bitrate_bps);
 }
 
 }  // namespace webrtc