Measure encoding time on encode callbacks.

webrtc/video/end_to_end_tests.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.
  */
 #include <algorithm>
 #include <map>
 #include <sstream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"

[mflodman, 2016/01/21 08:00:04]

What is this needed for?

[pbos-webrtc, 2016/01/21 14:18:33]

Local printf debugging. Gone.

 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
@@ skipping 2426 matching lines @@
     }
 
     bool CheckSendStats() {
       RTC_DCHECK(send_stream_ != nullptr);
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
       send_stats_filled_["NumStreams"] |=
           stats.substreams.size() == expected_send_ssrcs_.size();
 
       send_stats_filled_["CpuOveruseMetrics"] |=
-          stats.avg_encode_time_ms != 0 || stats.encode_usage_percent != 0;
+          stats.avg_encode_time_ms != 0 && stats.encode_usage_percent != 0;
 
       send_stats_filled_["EncoderImplementationName"] |=
           stats.encoder_implementation_name ==
           test::FakeEncoder::kImplementationName;
 
       for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
                stats.substreams.begin();
            it != stats.substreams.end(); ++it) {
         EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
                     expected_send_ssrcs_.end());
@@ skipping 847 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc