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Issue 1417683006: modules: more interface -> include renames (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix last incorrectly wrapped pragma message Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 11
12 #include <list> 12 #include <list>
13 13
14 #include "testing/gmock/include/gmock/gmock.h" 14 #include "testing/gmock/include/gmock/gmock.h"
15 #include "testing/gtest/include/gtest/gtest.h" 15 #include "testing/gtest/include/gtest/gtest.h"
16 #include "webrtc/base/scoped_ptr.h" 16 #include "webrtc/base/scoped_ptr.h"
17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 17 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
18 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" 18 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
19 #include "webrtc/video_engine/payload_router.h" 19 #include "webrtc/video_engine/payload_router.h"
20 20
21 using ::testing::_; 21 using ::testing::_;
22 using ::testing::AnyNumber; 22 using ::testing::AnyNumber;
23 using ::testing::NiceMock; 23 using ::testing::NiceMock;
24 using ::testing::Return; 24 using ::testing::Return;
25 25
26 namespace webrtc { 26 namespace webrtc {
27 27
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200 bitrates.resize(3); 200 bitrates.resize(3);
201 bitrates[1] = bitrate_2; 201 bitrates[1] = bitrate_2;
202 bitrates[2] = bitrate_1 + bitrate_2; 202 bitrates[2] = bitrate_1 + bitrate_2;
203 EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1)) 203 EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
204 .Times(1); 204 .Times(1);
205 EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2)) 205 EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
206 .Times(1); 206 .Times(1);
207 payload_router_->SetTargetSendBitrates(bitrates); 207 payload_router_->SetTargetSendBitrates(bitrates);
208 } 208 }
209 } // namespace webrtc 209 } // namespace webrtc
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