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-rw-r--r--third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc93
1 files changed, 93 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
new file mode 100644
index 0000000000..503135d87f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
@@ -0,0 +1,93 @@
+/*
+ * 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 <array>
+#include <memory>
+#include <utility>
+
+#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Exactly;
+using ::testing::StrictMock;
+
+namespace {
+rtc::scoped_refptr<webrtc::AudioProcessing> CreateAudioProcessing() {
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm(
+ webrtc::AudioProcessingBuilderForTesting().Create());
+ RTC_DCHECK(apm);
+ return apm;
+}
+
+std::unique_ptr<webrtc::test::MockAecDump> CreateMockAecDump() {
+ auto mock_aec_dump =
+ std::make_unique<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));
+}
+
+} // namespace
+
+TEST(AecDumpIntegration, ConfigurationAndInitShouldBeLogged) {
+ auto apm = CreateAudioProcessing();
+
+ apm->AttachAecDump(CreateMockAecDump());
+}
+
+TEST(AecDumpIntegration,
+ RenderStreamShouldBeLoggedOnceEveryProcessReverseStream) {
+ auto apm = CreateAudioProcessing();
+ auto mock_aec_dump = CreateMockAecDump();
+ constexpr int kNumChannels = 1;
+ constexpr int kNumSampleRateHz = 16000;
+ constexpr int kNumSamplesPerChannel = kNumSampleRateHz / 100;
+ std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
+ frame.fill(0.f);
+ webrtc::StreamConfig stream_config(kNumSampleRateHz, kNumChannels);
+
+ EXPECT_CALL(*mock_aec_dump.get(), WriteRenderStreamMessage(_, _, _))
+ .Times(Exactly(1));
+
+ apm->AttachAecDump(std::move(mock_aec_dump));
+ apm->ProcessReverseStream(frame.data(), stream_config, stream_config,
+ frame.data());
+}
+
+TEST(AecDumpIntegration, CaptureStreamShouldBeLoggedOnceEveryProcessStream) {
+ auto apm = CreateAudioProcessing();
+ auto mock_aec_dump = CreateMockAecDump();
+ constexpr int kNumChannels = 1;
+ constexpr int kNumSampleRateHz = 16000;
+ constexpr int kNumSamplesPerChannel = kNumSampleRateHz / 100;
+ std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
+ frame.fill(0.f);
+
+ webrtc::StreamConfig stream_config(kNumSampleRateHz, kNumChannels);
+
+ EXPECT_CALL(*mock_aec_dump.get(), AddCaptureStreamInput(_, _, _))
+ .Times(AtLeast(1));
+
+ EXPECT_CALL(*mock_aec_dump.get(), AddCaptureStreamOutput(_, _, _))
+ .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(frame.data(), stream_config, stream_config, frame.data());
+}