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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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27 #include "content/renderer/media/audio_hardware.h" | 27 #include "content/renderer/media/audio_hardware.h" |
28 #include "content/renderer/media/audio_input_message_filter.h" | 28 #include "content/renderer/media/audio_input_message_filter.h" |
29 #include "content/renderer/media/audio_message_filter.h" | 29 #include "content/renderer/media/audio_message_filter.h" |
30 #include "content/renderer/media/webrtc_audio_device_impl.h" | 30 #include "content/renderer/media/webrtc_audio_device_impl.h" |
31 #include "content/renderer/render_process.h" | 31 #include "content/renderer/render_process.h" |
32 #include "content/renderer/render_thread_impl.h" | 32 #include "content/renderer/render_thread_impl.h" |
33 #include "content/renderer/renderer_webkitplatformsupport_impl.h" | 33 #include "content/renderer/renderer_webkitplatformsupport_impl.h" |
34 #include "net/url_request/url_request_test_util.h" | 34 #include "net/url_request/url_request_test_util.h" |
35 #include "testing/gmock/include/gmock/gmock.h" | 35 #include "testing/gmock/include/gmock/gmock.h" |
36 #include "testing/gtest/include/gtest/gtest.h" | 36 #include "testing/gtest/include/gtest/gtest.h" |
37 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h" | 37 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h" |
38 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h" | 38 #include "third_party/webrtc/voice_engine/include/voe_base.h" |
39 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h" | 39 #include "third_party/webrtc/voice_engine/include/voe_file.h" |
40 #include "third_party/webrtc/voice_engine/main/interface/voe_network.h" | 40 #include "third_party/webrtc/voice_engine/include/voe_network.h" |
41 | 41 |
42 using base::win::ScopedCOMInitializer; | 42 using base::win::ScopedCOMInitializer; |
43 using testing::_; | 43 using testing::_; |
44 using testing::InvokeWithoutArgs; | 44 using testing::InvokeWithoutArgs; |
45 using testing::Return; | 45 using testing::Return; |
46 using testing::StrEq; | 46 using testing::StrEq; |
47 | 47 |
48 // This class is a mock of the child process singleton which is needed | 48 // This class is a mock of the child process singleton which is needed |
49 // to be able to create a RenderThread object. | 49 // to be able to create a RenderThread object. |
50 class WebRTCMockRenderProcess : public RenderProcess { | 50 class WebRTCMockRenderProcess : public RenderProcess { |
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369 WebRTCTransportImpl::~WebRTCTransportImpl() {} | 369 WebRTCTransportImpl::~WebRTCTransportImpl() {} |
370 | 370 |
371 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 371 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
372 return network_->ReceivedRTPPacket(channel, data, len); | 372 return network_->ReceivedRTPPacket(channel, data, len); |
373 } | 373 } |
374 | 374 |
375 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 375 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
376 int len) { | 376 int len) { |
377 return network_->ReceivedRTCPPacket(channel, data, len); | 377 return network_->ReceivedRTCPPacket(channel, data, len); |
378 } | 378 } |
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