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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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 "content/test/webrtc_audio_device_test.h" | 5 #include "content/test/webrtc_audio_device_test.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/bind_helpers.h" | 8 #include "base/bind_helpers.h" |
9 #include "base/compiler_specific.h" | 9 #include "base/compiler_specific.h" |
10 #include "base/file_util.h" | 10 #include "base/file_util.h" |
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24 #include "content/public/common/content_paths.h" | 24 #include "content/public/common/content_paths.h" |
25 #include "content/public/test/test_browser_thread.h" | 25 #include "content/public/test/test_browser_thread.h" |
26 #include "content/renderer/media/audio_input_message_filter.h" | 26 #include "content/renderer/media/audio_input_message_filter.h" |
27 #include "content/renderer/media/audio_message_filter.h" | 27 #include "content/renderer/media/audio_message_filter.h" |
28 #include "content/renderer/media/webrtc_audio_device_impl.h" | 28 #include "content/renderer/media/webrtc_audio_device_impl.h" |
29 #include "content/renderer/render_process.h" | 29 #include "content/renderer/render_process.h" |
30 #include "content/renderer/render_thread_impl.h" | 30 #include "content/renderer/render_thread_impl.h" |
31 #include "content/renderer/renderer_webkitplatformsupport_impl.h" | 31 #include "content/renderer/renderer_webkitplatformsupport_impl.h" |
32 #include "media/audio/audio_parameters.h" | 32 #include "media/audio/audio_parameters.h" |
33 #include "media/base/audio_hardware_config.h" | 33 #include "media/base/audio_hardware_config.h" |
| 34 #include "net/ssl/client_cert_store.h" |
34 #include "net/url_request/url_request_test_util.h" | 35 #include "net/url_request/url_request_test_util.h" |
35 #include "testing/gmock/include/gmock/gmock.h" | 36 #include "testing/gmock/include/gmock/gmock.h" |
36 #include "testing/gtest/include/gtest/gtest.h" | 37 #include "testing/gtest/include/gtest/gtest.h" |
37 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h" | 38 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h" |
38 #include "third_party/webrtc/voice_engine/include/voe_base.h" | 39 #include "third_party/webrtc/voice_engine/include/voe_base.h" |
39 #include "third_party/webrtc/voice_engine/include/voe_file.h" | 40 #include "third_party/webrtc/voice_engine/include/voe_file.h" |
40 #include "third_party/webrtc/voice_engine/include/voe_network.h" | 41 #include "third_party/webrtc/voice_engine/include/voe_network.h" |
41 | 42 |
42 #if defined(OS_WIN) | 43 #if defined(OS_WIN) |
43 #include "base/win/scoped_com_initializer.h" | 44 #include "base/win/scoped_com_initializer.h" |
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166 } | 167 } |
167 | 168 |
168 virtual bool AllowMicAccess(const GURL& origin) OVERRIDE { | 169 virtual bool AllowMicAccess(const GURL& origin) OVERRIDE { |
169 return false; | 170 return false; |
170 } | 171 } |
171 | 172 |
172 virtual bool AllowCameraAccess(const GURL& origin) OVERRIDE { | 173 virtual bool AllowCameraAccess(const GURL& origin) OVERRIDE { |
173 return false; | 174 return false; |
174 } | 175 } |
175 | 176 |
| 177 virtual scoped_ptr<net::ClientCertStore> GetClientCertStore() OVERRIDE { |
| 178 return scoped_ptr<net::ClientCertStore>(); |
| 179 } |
| 180 |
176 private: | 181 private: |
177 net::URLRequestContext* test_request_context_; | 182 net::URLRequestContext* test_request_context_; |
178 | 183 |
179 DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext); | 184 DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext); |
180 }; | 185 }; |
181 | 186 |
182 ACTION_P(QuitMessageLoop, loop_or_proxy) { | 187 ACTION_P(QuitMessageLoop, loop_or_proxy) { |
183 loop_or_proxy->PostTask(FROM_HERE, base::MessageLoop::QuitClosure()); | 188 loop_or_proxy->PostTask(FROM_HERE, base::MessageLoop::QuitClosure()); |
184 } | 189 } |
185 | 190 |
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438 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 443 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
439 return network_->ReceivedRTPPacket(channel, data, len); | 444 return network_->ReceivedRTPPacket(channel, data, len); |
440 } | 445 } |
441 | 446 |
442 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 447 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
443 int len) { | 448 int len) { |
444 return network_->ReceivedRTCPPacket(channel, data, len); | 449 return network_->ReceivedRTCPPacket(channel, data, len); |
445 } | 450 } |
446 | 451 |
447 } // namespace content | 452 } // namespace content |
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