Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interfac

webrtc/call/call_perf_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 <limits>
 #include <memory>
 #include <sstream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics_default.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/drifting_clock.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
-#include "webrtc/test/histogram.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 using webrtc::test::DriftingClock;
 using webrtc::test::FakeAudioDevice;
@@ skipping 93 matching lines @@
 
 void CallPerfTest::TestAudioVideoSync(FecMode fec,
                                       CreateOrder create_first,
                                       float video_ntp_speed,
                                       float video_rtp_speed,
                                       float audio_rtp_speed) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
 
-  test::ClearHistograms();
+  metrics::Reset();
   VoiceEngine* voice_engine = VoiceEngine::Create();
   VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
   VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
   const std::string audio_filename =
       test::ResourcePath("voice_engine/audio_long16", "pcm");
   ASSERT_STRNE("", audio_filename.c_str());
   FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
                                     audio_rtp_speed);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
   Config voe_config;
@@ skipping 129 matching lines @@
 
   voe_base->DeleteChannel(send_channel_id);
   voe_base->DeleteChannel(recv_channel_id);
   voe_base->Release();
   voe_codec->Release();
 
   DestroyCalls();
 
   VoiceEngine::Delete(voice_engine);
 
-  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.AVSyncOffsetInMs"));
+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
   TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
                      DriftingClock::PercentsFaster(10.0f),
                      DriftingClock::kNoDrift, DriftingClock::kNoDrift);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
   TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
@@ skipping 379 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc