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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 #include <memory> | 11 #include <memory> |
| 12 | 12 |
| 13 #include "testing/gmock/include/gmock/gmock.h" | 13 #include "webrtc/test/gmock.h" |
| 14 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "webrtc/test/gtest.h" |
| 15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| 16 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" | 16 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
| 17 #include "webrtc/modules/video_coding/include/video_codec_interface.h" | 17 #include "webrtc/modules/video_coding/include/video_codec_interface.h" |
| 18 #include "webrtc/video/payload_router.h" | 18 #include "webrtc/video/payload_router.h" |
| 19 | 19 |
| 20 using ::testing::_; | 20 using ::testing::_; |
| 21 using ::testing::AnyNumber; | 21 using ::testing::AnyNumber; |
| 22 using ::testing::NiceMock; | 22 using ::testing::NiceMock; |
| 23 using ::testing::Return; | 23 using ::testing::Return; |
| 24 | 24 |
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| 156 const size_t kTestMinPayloadLength = 1001; | 156 const size_t kTestMinPayloadLength = 1001; |
| 157 EXPECT_CALL(rtp_1, MaxDataPayloadLength()) | 157 EXPECT_CALL(rtp_1, MaxDataPayloadLength()) |
| 158 .Times(1) | 158 .Times(1) |
| 159 .WillOnce(Return(kTestMinPayloadLength + 10)); | 159 .WillOnce(Return(kTestMinPayloadLength + 10)); |
| 160 EXPECT_CALL(rtp_2, MaxDataPayloadLength()) | 160 EXPECT_CALL(rtp_2, MaxDataPayloadLength()) |
| 161 .Times(1) | 161 .Times(1) |
| 162 .WillOnce(Return(kTestMinPayloadLength)); | 162 .WillOnce(Return(kTestMinPayloadLength)); |
| 163 EXPECT_EQ(kTestMinPayloadLength, payload_router.MaxPayloadLength()); | 163 EXPECT_EQ(kTestMinPayloadLength, payload_router.MaxPayloadLength()); |
| 164 } | 164 } |
| 165 } // namespace webrtc | 165 } // namespace webrtc |
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