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Side by Side Diff: webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

Issue 1946183002: Removing some old code which looked like it had to do with NACK handling but in reality did nothing. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 7 months ago
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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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