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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
91 lines
2.9 KiB
C++
91 lines
2.9 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <utility>
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#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/ptr_util.h"
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using testing::_;
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using testing::AtLeast;
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using testing::Exactly;
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using testing::Matcher;
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using testing::StrictMock;
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namespace {
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std::unique_ptr<webrtc::AudioProcessing> CreateAudioProcessing() {
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webrtc::Config config;
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std::unique_ptr<webrtc::AudioProcessing> apm(
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webrtc::AudioProcessing::Create(config));
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RTC_DCHECK(apm);
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return apm;
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}
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std::unique_ptr<webrtc::test::MockAecDump> CreateMockAecDump() {
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auto mock_aec_dump =
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rtc::MakeUnique<testing::StrictMock<webrtc::test::MockAecDump>>();
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EXPECT_CALL(*mock_aec_dump.get(), WriteConfig(_)).Times(AtLeast(1));
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EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_)).Times(AtLeast(1));
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return std::unique_ptr<webrtc::test::MockAecDump>(std::move(mock_aec_dump));
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}
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std::unique_ptr<webrtc::AudioFrame> CreateFakeFrame() {
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auto fake_frame = rtc::MakeUnique<webrtc::AudioFrame>();
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fake_frame->num_channels_ = 1;
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fake_frame->sample_rate_hz_ = 48000;
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fake_frame->samples_per_channel_ = 480;
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return fake_frame;
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}
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} // namespace
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TEST(AecDumpIntegration, ConfigurationAndInitShouldBeLogged) {
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auto apm = CreateAudioProcessing();
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apm->AttachAecDump(CreateMockAecDump());
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}
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TEST(AecDumpIntegration,
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RenderStreamShouldBeLoggedOnceEveryProcessReverseStream) {
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auto apm = CreateAudioProcessing();
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auto mock_aec_dump = CreateMockAecDump();
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auto fake_frame = CreateFakeFrame();
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EXPECT_CALL(*mock_aec_dump.get(),
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WriteRenderStreamMessage(Matcher<const webrtc::AudioFrame&>(_)))
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.Times(Exactly(1));
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apm->AttachAecDump(std::move(mock_aec_dump));
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apm->ProcessReverseStream(fake_frame.get());
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}
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TEST(AecDumpIntegration, CaptureStreamShouldBeLoggedOnceEveryProcessStream) {
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auto apm = CreateAudioProcessing();
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auto mock_aec_dump = CreateMockAecDump();
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auto fake_frame = CreateFakeFrame();
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EXPECT_CALL(*mock_aec_dump.get(),
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AddCaptureStreamInput(Matcher<const webrtc::AudioFrame&>(_)))
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.Times(AtLeast(1));
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EXPECT_CALL(*mock_aec_dump.get(),
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AddCaptureStreamOutput(Matcher<const webrtc::AudioFrame&>(_)))
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.Times(Exactly(1));
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EXPECT_CALL(*mock_aec_dump.get(), AddAudioProcessingState(_))
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.Times(Exactly(1));
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EXPECT_CALL(*mock_aec_dump.get(), WriteCaptureStreamMessage())
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.Times(Exactly(1));
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apm->AttachAecDump(std::move(mock_aec_dump));
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apm->ProcessStream(fake_frame.get());
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}
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