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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/renderer/media/media_stream_audio_processor.h" | 13 #include "content/renderer/media/media_stream_audio_processor.h" |
13 #include "media/audio/audio_parameters.h" | 14 #include "media/audio/audio_parameters.h" |
14 #include "media/base/audio_bus.h" | 15 #include "media/base/audio_bus.h" |
15 #include "testing/gmock/include/gmock/gmock.h" | 16 #include "testing/gmock/include/gmock/gmock.h" |
16 #include "testing/gtest/include/gtest/gtest.h" | 17 #include "testing/gtest/include/gtest/gtest.h" |
17 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 18 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
18 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
19 | 20 |
20 using ::testing::_; | 21 using ::testing::_; |
21 using ::testing::AnyNumber; | 22 using ::testing::AnyNumber; |
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149 media::AudioParameters params_; | 150 media::AudioParameters params_; |
150 }; | 151 }; |
151 | 152 |
152 TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) { | 153 TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) { |
153 // Setup the audio processor without enabling the flag. | 154 // Setup the audio processor without enabling the flag. |
154 blink::WebMediaConstraints constraints; | 155 blink::WebMediaConstraints constraints; |
155 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( | 156 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( |
156 new WebRtcAudioDeviceImpl()); | 157 new WebRtcAudioDeviceImpl()); |
157 scoped_refptr<MediaStreamAudioProcessor> audio_processor( | 158 scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
158 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( | 159 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
159 params_, constraints, 0, webrtc_audio_device.get())); | 160 params_, constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE, |
161 webrtc_audio_device.get())); | |
160 EXPECT_FALSE(audio_processor->has_audio_processing()); | 162 EXPECT_FALSE(audio_processor->has_audio_processing()); |
161 | 163 |
162 ProcessDataAndVerifyFormat(audio_processor, | 164 ProcessDataAndVerifyFormat(audio_processor, |
163 params_.sample_rate(), | 165 params_.sample_rate(), |
164 params_.channels(), | 166 params_.channels(), |
165 params_.sample_rate() / 100); | 167 params_.sample_rate() / 100); |
166 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives | 168 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives |
167 // |audio_processor|. | 169 // |audio_processor|. |
168 audio_processor = NULL; | 170 audio_processor = NULL; |
169 } | 171 } |
170 | 172 |
171 TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { | 173 TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) { |
172 // Setup the audio processor with enabling the flag. | 174 // Setup the audio processor with enabling the flag. |
173 CommandLine::ForCurrentProcess()->AppendSwitch( | 175 CommandLine::ForCurrentProcess()->AppendSwitch( |
174 switches::kEnableAudioTrackProcessing); | 176 switches::kEnableAudioTrackProcessing); |
175 blink::WebMediaConstraints constraints; | 177 blink::WebMediaConstraints constraints; |
176 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( | 178 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( |
177 new WebRtcAudioDeviceImpl()); | 179 new WebRtcAudioDeviceImpl()); |
178 scoped_refptr<MediaStreamAudioProcessor> audio_processor( | 180 scoped_refptr<MediaStreamAudioProcessor> audio_processor( |
179 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( | 181 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( |
180 params_, constraints, 0, webrtc_audio_device.get())); | 182 params_, constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE, |
183 webrtc_audio_device.get())); | |
181 EXPECT_TRUE(audio_processor->has_audio_processing()); | 184 EXPECT_TRUE(audio_processor->has_audio_processing()); |
182 VerifyDefaultComponents(audio_processor); | 185 VerifyDefaultComponents(audio_processor); |
183 | 186 |
184 ProcessDataAndVerifyFormat(audio_processor, | 187 ProcessDataAndVerifyFormat(audio_processor, |
185 kAudioProcessingSampleRate, | 188 kAudioProcessingSampleRate, |
186 kAudioProcessingNumberOfChannel, | 189 kAudioProcessingNumberOfChannel, |
187 kAudioProcessingSampleRate / 100); | 190 kAudioProcessingSampleRate / 100); |
188 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives | 191 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives |
189 // |audio_processor|. | 192 // |audio_processor|. |
190 audio_processor = NULL; | 193 audio_processor = NULL; |
191 } | 194 } |
192 | 195 |
196 TEST_F(MediaStreamAudioProcessorTest, VerifyCastWithoutAudioProcessing) { | |
Alpha Left Google
2014/03/07 21:27:29
Should be VerifyTabCapture...
no longer working on chromium
2014/03/07 21:56:52
But we only verified MEDIA_LOOPBACK_AUDIO_CAPTURE
Alpha Left Google
2014/03/07 22:09:42
Cast is really using tab capture. This enum is for
no longer working on chromium
2014/03/09 15:12:57
Done.
| |
197 // Setup the audio processor with enabling the flag. | |
198 CommandLine::ForCurrentProcess()->AppendSwitch( | |
199 switches::kEnableAudioTrackProcessing); | |
200 blink::WebMediaConstraints constraints; | |
201 scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( | |
202 new WebRtcAudioDeviceImpl()); | |
203 // Create MediaStreamAudioProcessor instance for MEDIA_TAB_AUDIO_CAPTURE type. | |
204 scoped_refptr<MediaStreamAudioProcessor> audio_processor( | |
205 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( | |
206 params_, constraints, 0, MEDIA_TAB_AUDIO_CAPTURE, | |
207 webrtc_audio_device.get())); | |
208 EXPECT_FALSE(audio_processor->has_audio_processing()); | |
209 | |
210 ProcessDataAndVerifyFormat(audio_processor, | |
211 params_.sample_rate(), | |
212 params_.channels(), | |
213 params_.sample_rate() / 100); | |
214 | |
215 // Create MediaStreamAudioProcessor instance for MEDIA_LOOPBACK_AUDIO_CAPTURE. | |
216 audio_processor = | |
217 new talk_base::RefCountedObject<MediaStreamAudioProcessor>( | |
218 params_, constraints, 0, MEDIA_LOOPBACK_AUDIO_CAPTURE, | |
219 webrtc_audio_device.get()); | |
220 EXPECT_FALSE(audio_processor->has_audio_processing()); | |
221 | |
222 // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives | |
223 // |audio_processor|. | |
224 audio_processor = NULL; | |
225 } | |
226 | |
193 } // namespace content | 227 } // namespace content |
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