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

Issue 221863003: Notify the track before source provider goes away. (Closed) Base URL: http://git.chromium.org/chromium/src.git@master
Patch Set: fixed the comments. Created 6 years, 8 months 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/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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