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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 |
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| 184 configuration.outgoing_transport = &transport_; | 184 configuration.outgoing_transport = &transport_; |
| 185 rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration); | 185 rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration); |
| 186 | 186 |
| 187 rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_)); | 187 rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_)); |
| 188 | 188 |
| 189 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( | 189 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( |
| 190 &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_)); | 190 &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_)); |
| 191 | 191 |
| 192 rtp_rtcp_module_->SetSSRC(kTestSsrc); | 192 rtp_rtcp_module_->SetSSRC(kTestSsrc); |
| 193 rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound); | 193 rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound); |
| 194 rtp_receiver_->SetNACKStatus(kNackRtcp); | |
| 195 rtp_rtcp_module_->SetStorePacketsStatus(true, 600); | 194 rtp_rtcp_module_->SetStorePacketsStatus(true, 600); |
| 196 EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true)); | 195 EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true)); |
| 197 rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber); | 196 rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber); |
| 198 rtp_rtcp_module_->SetStartTimestamp(111111); | 197 rtp_rtcp_module_->SetStartTimestamp(111111); |
| 199 | 198 |
| 200 transport_.SetSendModule(rtp_rtcp_module_, &rtp_payload_registry_, | 199 transport_.SetSendModule(rtp_rtcp_module_, &rtp_payload_registry_, |
| 201 rtp_receiver_.get()); | 200 rtp_receiver_.get()); |
| 202 | 201 |
| 203 VideoCodec video_codec; | 202 VideoCodec video_codec; |
| 204 memset(&video_codec, 0, sizeof(video_codec)); | 203 memset(&video_codec, 0, sizeof(video_codec)); |
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| 333 RunRtxTest(kRtxRetransmitted, 10); | 332 RunRtxTest(kRtxRetransmitted, 10); |
| 334 EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin())); | 333 EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin())); |
| 335 EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1, | 334 EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1, |
| 336 *(receiver_.sequence_numbers_.rbegin())); | 335 *(receiver_.sequence_numbers_.rbegin())); |
| 337 EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size()); | 336 EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size()); |
| 338 EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_); | 337 EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_); |
| 339 EXPECT_TRUE(ExpectedPacketsReceived()); | 338 EXPECT_TRUE(ExpectedPacketsReceived()); |
| 340 } | 339 } |
| 341 | 340 |
| 342 } // namespace webrtc | 341 } // namespace webrtc |
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