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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/command_line.h" | 5 #include "base/command_line.h" |
6 #include "base/file_util.h" | 6 #include "base/file_util.h" |
7 #include "base/files/file_path.h" | 7 #include "base/files/file_path.h" |
8 #include "base/logging.h" | 8 #include "base/logging.h" |
9 #include "base/path_service.h" | 9 #include "base/path_service.h" |
10 #include "base/time/time.h" | 10 #include "base/time/time.h" |
11 #include "content/public/common/content_switches.h" | 11 #include "content/public/common/content_switches.h" |
12 #include "content/public/common/media_stream_request.h" | 12 #include "content/public/common/media_stream_request.h" |
13 #include "content/renderer/media/media_stream_audio_processor.h" | 13 #include "content/renderer/media/media_stream_audio_processor.h" |
| 14 #include "content/renderer/media/mock_media_constraint_factory.h" |
14 #include "media/audio/audio_parameters.h" | 15 #include "media/audio/audio_parameters.h" |
15 #include "media/base/audio_bus.h" | 16 #include "media/base/audio_bus.h" |
16 #include "testing/gmock/include/gmock/gmock.h" | 17 #include "testing/gmock/include/gmock/gmock.h" |
17 #include "testing/gtest/include/gtest/gtest.h" | 18 #include "testing/gtest/include/gtest/gtest.h" |
18 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 19 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 20 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
20 | 21 |
21 using ::testing::_; | 22 using ::testing::_; |
22 using ::testing::AnyNumber; | 23 using ::testing::AnyNumber; |
23 using ::testing::AtLeast; | 24 using ::testing::AtLeast; |
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165 ProcessDataAndVerifyFormat(audio_processor, | 166 ProcessDataAndVerifyFormat(audio_processor, |
166 params_.sample_rate(), | 167 params_.sample_rate(), |
167 params_.channels(), | 168 params_.channels(), |
168 params_.sample_rate() / 100); | 169 params_.sample_rate() / 100); |
169 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives | 170 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives |
170 // |audio_processor|. | 171 // |audio_processor|. |
171 audio_processor = NULL; | 172 audio_processor = NULL; |
172 } | 173 } |
173 | 174 |
174 TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { | 175 TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { |
175 // Setup the audio processor with enabling the flag. | 176 // Setup the audio processor with the flag enabled. |
176 CommandLine::ForCurrentProcess()->AppendSwitch( | 177 CommandLine::ForCurrentProcess()->AppendSwitch( |
177 switches::kEnableAudioTrackProcessing); | 178 switches::kEnableAudioTrackProcessing); |
178 blink::WebMediaConstraints constraints; | 179 blink::WebMediaConstraints constraints; |
179 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( | 180 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( |
180 new WebRtcAudioDeviceImpl()); | 181 new WebRtcAudioDeviceImpl()); |
181 scoped_refptr<MediaStreamAudioProcessor> audio_processor( | 182 scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
182 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( | 183 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
183 constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE, | 184 constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE, |
184 webrtc_audio_device.get())); | 185 webrtc_audio_device.get())); |
185 EXPECT_TRUE(audio_processor->has_audio_processing()); | 186 EXPECT_TRUE(audio_processor->has_audio_processing()); |
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220 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( | 221 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
221 constraints, 0, MEDIA_LOOPBACK_AUDIO_CAPTURE, | 222 constraints, 0, MEDIA_LOOPBACK_AUDIO_CAPTURE, |
222 webrtc_audio_device.get()); | 223 webrtc_audio_device.get()); |
223 EXPECT_FALSE(audio_processor->has_audio_processing()); | 224 EXPECT_FALSE(audio_processor->has_audio_processing()); |
224 | 225 |
225 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives | 226 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives |
226 // |audio_processor|. | 227 // |audio_processor|. |
227 audio_processor = NULL; | 228 audio_processor = NULL; |
228 } | 229 } |
229 | 230 |
| 231 TEST_F(MediaStreamAudioProcessorTest, TurnOffDefaultConstraints) { |
| 232 // Setup the audio processor with enabling the flag. |
| 233 CommandLine::ForCurrentProcess()->AppendSwitch( |
| 234 switches::kEnableAudioTrackProcessing); |
| 235 |
| 236 // Turn off the default constraints and pass it to MediaStreamAudioProcessor. |
| 237 MockMediaConstraintFactory constraint_factory; |
| 238 constraint_factory.DisableDefaultAudioConstraints(); |
| 239 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( |
| 240 new WebRtcAudioDeviceImpl()); |
| 241 scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
| 242 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
| 243 constraint_factory.CreateWebMediaConstraints(), 0, |
| 244 MEDIA_DEVICE_AUDIO_CAPTURE, webrtc_audio_device.get())); |
| 245 EXPECT_FALSE(audio_processor->has_audio_processing()); |
| 246 audio_processor->OnCaptureFormatChanged(params_); |
| 247 |
| 248 ProcessDataAndVerifyFormat(audio_processor, |
| 249 params_.sample_rate(), |
| 250 params_.channels(), |
| 251 params_.sample_rate() / 100); |
| 252 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives |
| 253 // |audio_processor|. |
| 254 audio_processor = NULL; |
| 255 } |
| 256 |
230 } // namespace content | 257 } // namespace content |
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