Make use of new APM statistics interface.

webrtc/audio/audio_send_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 <string>
 #include <vector>
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Return;
 
 const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
 const char* kCName = "foo_name";
 const int kAudioLevelId = 2;
 const int kTransportSequenceNumberId = 4;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
-const float kResidualEchoLikelihood = 0.0f;
+const float kResidualEchoLikelihood = -1.0f;
 const unsigned int kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 const int kTelephoneEventPayloadType = 123;
 const int kTelephoneEventCode = 45;
 const int kTelephoneEventDuration = 6789;
 const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000};
 
 class MockLimitObserver : public BitrateAllocator::LimitObserver {
@@ skipping 120 matching lines @@
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
         .WillRepeatedly(Return(kCallStats));
     EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
         .WillRepeatedly(Return(report_blocks));
 
     EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
         .WillRepeatedly(DoAll(SetArgReferee<1>(kIsacCodec), Return(0)));
     EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
-    EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_))
-        .WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0)));
-    EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _))
-        .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss),
-                        SetArgReferee<1>(kEchoReturnLossEnhancement),
-                        Return(0)));
-    EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
-        .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
-                        SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
+    EXPECT_CALL(voice_engine_, audio_processing())
+        .WillRepeatedly(Return(&audio_processing_));
+
+    // We have to set the instantaneous value, the average, min and max. We only
+    // care about the instantaneous value, so we set all to the same value.
+    audio_processing_stats_.echo_return_loss.Set(
+        kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss);
+    audio_processing_stats_.echo_return_loss_enhancement.Set(
+        kEchoReturnLossEnhancement, kEchoReturnLossEnhancement,
+        kEchoReturnLossEnhancement, kEchoReturnLossEnhancement);
+    audio_processing_stats_.delay_median = kEchoDelayMedian;
+    audio_processing_stats_.delay_standard_deviation = kEchoDelayStdDev;
+
+    EXPECT_CALL(audio_processing_, GetStatistics())
+        .WillRepeatedly(Return(audio_processing_stats_));
   }
 
  private:
   SimulatedClock simulated_clock_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
+  MockAudioProcessing audio_processing_;
+  AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
   CongestionController congestion_controller_;
   MockRtcEventLog event_log_;
   testing::NiceMock<MockLimitObserver> limit_observer_;
   BitrateAllocator bitrate_allocator_;
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 };
 }  // namespace
 
@@ skipping 163 matching lines @@
   EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
       .WillOnce(Return(0));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
 }
 
 }  // namespace test
 }  // namespace webrtc