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Side by Side Diff: content/test/webrtc_audio_device_test.cc

Issue 10837131: Update webrtc to 2565. (Closed) Base URL: svn://chrome-svn/chrome/trunk/src/
Patch Set: Update webrtc to 2565. Created 8 years, 4 months ago
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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"
(...skipping 16 matching lines...) Expand all
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 {
(...skipping 318 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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