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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/logging.h" | 6 #include "base/logging.h" |
| 7 #include "content/public/common/content_switches.h" | 7 #include "content/public/common/content_switches.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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| 99 constraints, NULL); | 99 constraints, NULL); |
| 100 capturer_source_ = new MockCapturerSource(); | 100 capturer_source_ = new MockCapturerSource(); |
| 101 EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1)); | 101 EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1)); |
| 102 capturer_->SetCapturerSourceForTesting(capturer_source_, params_); | 102 capturer_->SetCapturerSourceForTesting(capturer_source_, params_); |
| 103 | 103 |
| 104 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); | 104 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| 105 EXPECT_CALL(*capturer_source_.get(), Start()); | 105 EXPECT_CALL(*capturer_source_.get(), Start()); |
| 106 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 106 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
| 107 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 107 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| 108 track_.reset(new WebRtcLocalAudioTrack(adapter, capturer_, NULL)); | 108 track_.reset(new WebRtcLocalAudioTrack(adapter, capturer_, NULL)); |
| 109 static_cast<WebRtcLocalAudioSourceProvider*>( | |
| 110 track_->audio_source_provider())->SetSinkParamsForTesting(params_); | |
| 111 track_->Start(); | 109 track_->Start(); |
| 112 | 110 |
| 113 // Connect a mock sink to the track. | 111 // Connect a mock sink to the track. |
| 114 scoped_ptr<MockPeerConnectionAudioSink> sink( | 112 scoped_ptr<MockPeerConnectionAudioSink> sink( |
| 115 new MockPeerConnectionAudioSink()); | 113 new MockPeerConnectionAudioSink()); |
| 116 track_->AddSink(sink.get()); | 114 track_->AddSink(sink.get()); |
| 117 | 115 |
| 118 int delay_ms = 65; | 116 int delay_ms = 65; |
| 119 bool key_pressed = true; | 117 bool key_pressed = true; |
| 120 double volume = 0.9; | 118 double volume = 0.9; |
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| 167 TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) { | 165 TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) { |
| 168 EnableAudioTrackProcessing(); | 166 EnableAudioTrackProcessing(); |
| 169 // Turn off the default constraints to verify that the sink will get packets | 167 // Turn off the default constraints to verify that the sink will get packets |
| 170 // with a buffer size smaller than 10ms. | 168 // with a buffer size smaller than 10ms. |
| 171 MockMediaConstraintFactory constraint_factory; | 169 MockMediaConstraintFactory constraint_factory; |
| 172 constraint_factory.DisableDefaultAudioConstraints(); | 170 constraint_factory.DisableDefaultAudioConstraints(); |
| 173 VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false); | 171 VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false); |
| 174 } | 172 } |
| 175 | 173 |
| 176 } // namespace content | 174 } // namespace content |
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