MockAecDump and integration tests between AecDump and AudioProcessing

webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc

Index: webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
diff --git a/webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc b/webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
new file mode 100644
index 0000000000000000000000000000000000000000..b1c30ef006221767e1b5dcc17c06f93faade7a82
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2017 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 <utility>
+
+#include "webrtc/base/ptr_util.h"
+#include "webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+using testing::_;
+using testing::AtLeast;
+using testing::Exactly;
+using testing::Matcher;
+using testing::StrictMock;
+
+namespace {
+std::unique_ptr<webrtc::AudioProcessing> CreateAudioProcessing() {
+  webrtc::Config config;
+  std::unique_ptr<webrtc::AudioProcessing> apm(
+      webrtc::AudioProcessing::Create(config));
+  RTC_DCHECK(apm);
+  return apm;
+}
+
+std::unique_ptr<webrtc::test::MockAecDump> CreateMockAecDump() {
+  auto mock_aec_dump =
+      rtc::MakeUnique<testing::StrictMock<webrtc::test::MockAecDump>>();
+  EXPECT_CALL(*mock_aec_dump.get(), WriteConfig(_)).Times(AtLeast(1));
+  EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_)).Times(AtLeast(1));
+  return std::unique_ptr<webrtc::test::MockAecDump>(std::move(mock_aec_dump));
+}
+
+std::unique_ptr<webrtc::AudioFrame> CreateFakeFrame() {
+  auto fake_frame = rtc::MakeUnique<webrtc::AudioFrame>();
+  fake_frame->num_channels_ = 1;
+  fake_frame->sample_rate_hz_ = 48000;
+  fake_frame->samples_per_channel_ = 480;
+  return fake_frame;
+}
+
+}  // namespace
+
+TEST(AecDumpIntegration, ConfigurationAndInitShouldBeLogged) {
+  auto apm = CreateAudioProcessing();
+
+  apm->AttachAecDump(CreateMockAecDump());
+}
+
+TEST(AecDumpIntegration,
+     RenderStreamShouldBeLoggedOnceEveryProcessReverseStream) {
+  auto apm = CreateAudioProcessing();
+  auto mock_aec_dump = CreateMockAecDump();
+  auto fake_frame = CreateFakeFrame();
+
+  EXPECT_CALL(*mock_aec_dump.get(),
+              WriteRenderStreamMessage(Matcher<const webrtc::AudioFrame&>(_)))
+      .Times(Exactly(1));
+
+  apm->AttachAecDump(std::move(mock_aec_dump));
+  apm->ProcessReverseStream(fake_frame.get());
+}
+
+TEST(AecDumpIntegration, CaptureStreamShouldBeLoggedOnceEveryProcessStream) {
+  auto apm = CreateAudioProcessing();
+  auto mock_aec_dump = CreateMockAecDump();
+  auto fake_frame = CreateFakeFrame();
+
+  EXPECT_CALL(*mock_aec_dump.get(),
+              AddCaptureStreamInput(Matcher<const webrtc::AudioFrame&>(_)))
+      .Times(AtLeast(1));
+
+  EXPECT_CALL(*mock_aec_dump.get(),
+              AddCaptureStreamOutput(Matcher<const webrtc::AudioFrame&>(_)))
+      .Times(Exactly(1));
+
+  EXPECT_CALL(*mock_aec_dump.get(), AddAudioProcessingState(_))
+      .Times(Exactly(1));
+
+  EXPECT_CALL(*mock_aec_dump.get(), WriteCaptureStreamMessage())
+      .Times(Exactly(1));
+
+  apm->AttachAecDump(std::move(mock_aec_dump));
+  apm->ProcessStream(fake_frame.get());
+}