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Side by Side Diff: content/renderer/media/webrtc_local_audio_track_unittest.cc

Issue 686523002: Standardize usage of virtual/override/final specifiers. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Rebase Created 6 years, 1 month ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "base/synchronization/waitable_event.h" 5 #include "base/synchronization/waitable_event.h"
6 #include "base/test/test_timeouts.h" 6 #include "base/test/test_timeouts.h"
7 #include "content/renderer/media/media_stream_audio_source.h" 7 #include "content/renderer/media/media_stream_audio_source.h"
8 #include "content/renderer/media/mock_media_constraint_factory.h" 8 #include "content/renderer/media/mock_media_constraint_factory.h"
9 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 9 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
10 #include "content/renderer/media/webrtc_audio_capturer.h" 10 #include "content/renderer/media/webrtc_audio_capturer.h"
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161 const media::AudioParameters& audio_params() const { return params_; } 161 const media::AudioParameters& audio_params() const { return params_; }
162 162
163 private: 163 private:
164 media::AudioParameters params_; 164 media::AudioParameters params_;
165 }; 165 };
166 166
167 } // namespace 167 } // namespace
168 168
169 class WebRtcLocalAudioTrackTest : public ::testing::Test { 169 class WebRtcLocalAudioTrackTest : public ::testing::Test {
170 protected: 170 protected:
171 virtual void SetUp() override { 171 void SetUp() override {
172 params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, 172 params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
173 media::CHANNEL_LAYOUT_STEREO, 2, 48000, 16, 480); 173 media::CHANNEL_LAYOUT_STEREO, 2, 48000, 16, 480);
174 MockMediaConstraintFactory constraint_factory; 174 MockMediaConstraintFactory constraint_factory;
175 blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio, 175 blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
176 "dummy"); 176 "dummy");
177 MediaStreamAudioSource* audio_source = new MediaStreamAudioSource(); 177 MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
178 blink_source_.setExtraData(audio_source); 178 blink_source_.setExtraData(audio_source);
179 179
180 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, 180 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
181 std::string(), std::string()); 181 std::string(), std::string());
182 capturer_ = WebRtcAudioCapturer::CreateCapturer( 182 capturer_ = WebRtcAudioCapturer::CreateCapturer(
183 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, 183 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
184 audio_source); 184 audio_source);
185 audio_source->SetAudioCapturer(capturer_.get()); 185 audio_source->SetAudioCapturer(capturer_.get());
186 capturer_source_ = new MockCapturerSource(capturer_.get()); 186 capturer_source_ = new MockCapturerSource(capturer_.get());
187 EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1)) 187 EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
188 .WillOnce(Return()); 188 .WillOnce(Return());
189 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); 189 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
190 EXPECT_CALL(*capturer_source_.get(), OnStart()); 190 EXPECT_CALL(*capturer_source_.get(), OnStart());
191 capturer_->SetCapturerSourceForTesting(capturer_source_, params_); 191 capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
192 } 192 }
193 193
194 virtual void TearDown() override { 194 void TearDown() override {
195 blink_source_.reset(); 195 blink_source_.reset();
196 blink::WebHeap::collectAllGarbageForTesting(); 196 blink::WebHeap::collectAllGarbageForTesting();
197 } 197 }
198 198
199 media::AudioParameters params_; 199 media::AudioParameters params_;
200 blink::WebMediaStreamSource blink_source_; 200 blink::WebMediaStreamSource blink_source_;
201 scoped_refptr<MockCapturerSource> capturer_source_; 201 scoped_refptr<MockCapturerSource> capturer_source_;
202 scoped_refptr<WebRtcAudioCapturer> capturer_; 202 scoped_refptr<WebRtcAudioCapturer> capturer_;
203 }; 203 };
204 204
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534 // Stopping the new source will stop the second track. 534 // Stopping the new source will stop the second track.
535 EXPECT_CALL(*source.get(), OnStop()).Times(1); 535 EXPECT_CALL(*source.get(), OnStop()).Times(1);
536 capturer->Stop(); 536 capturer->Stop();
537 537
538 // Even though this test don't use |capturer_source_| it will be stopped 538 // Even though this test don't use |capturer_source_| it will be stopped
539 // during teardown of the test harness. 539 // during teardown of the test harness.
540 EXPECT_CALL(*capturer_source_.get(), OnStop()); 540 EXPECT_CALL(*capturer_source_.get(), OnStop());
541 } 541 }
542 542
543 } // namespace content 543 } // namespace content
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