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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 |
| 11 #include <algorithm> | 11 #include <algorithm> |
| 12 #include <iterator> | 12 #include <iterator> |
| 13 #include <list> | 13 #include <list> |
| 14 #include <memory> | 14 #include <memory> |
| 15 #include <set> | 15 #include <set> |
| 16 | 16 |
| 17 #include "testing/gtest/include/gtest/gtest.h" | 17 #include "testing/gtest/include/gtest/gtest.h" |
| 18 #include "webrtc/base/rate_limiter.h" | |
| 19 #include "webrtc/common_types.h" | 18 #include "webrtc/common_types.h" |
| 20 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | 19 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
| 21 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 20 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| 22 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" | 21 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
| 23 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | 22 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
| 24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| 26 #include "webrtc/transport.h" | 25 #include "webrtc/transport.h" |
| 27 | 26 |
| 28 namespace webrtc { | 27 namespace webrtc { |
| 29 | 28 |
| 30 const int kVideoNackListSize = 30; | 29 const int kVideoNackListSize = 30; |
| 31 const uint32_t kTestSsrc = 3456; | 30 const uint32_t kTestSsrc = 3456; |
| 32 const uint16_t kTestSequenceNumber = 2345; | 31 const uint16_t kTestSequenceNumber = 2345; |
| 33 const uint32_t kTestNumberOfPackets = 1350; | 32 const uint32_t kTestNumberOfPackets = 1350; |
| 34 const int kTestNumberOfRtxPackets = 149; | 33 const int kTestNumberOfRtxPackets = 149; |
| 35 const int kNumFrames = 30; | 34 const int kNumFrames = 30; |
| 36 const int kPayloadType = 123; | 35 const int kPayloadType = 123; |
| 37 const int kRtxPayloadType = 98; | 36 const int kRtxPayloadType = 98; |
| 38 const int64_t kMaxRttMs = 1000; | |
| 39 | 37 |
| 40 class VerifyingRtxReceiver : public NullRtpData { | 38 class VerifyingRtxReceiver : public NullRtpData { |
| 41 public: | 39 public: |
| 42 VerifyingRtxReceiver() {} | 40 VerifyingRtxReceiver() {} |
| 43 | 41 |
| 44 int32_t OnReceivedPayloadData( | 42 int32_t OnReceivedPayloadData( |
| 45 const uint8_t* data, | 43 const uint8_t* data, |
| 46 size_t size, | 44 size_t size, |
| 47 const webrtc::WebRtcRTPHeader* rtp_header) override { | 45 const webrtc::WebRtcRTPHeader* rtp_header) override { |
| 48 if (!sequence_numbers_.empty()) | 46 if (!sequence_numbers_.empty()) |
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| 163 RTPPayloadRegistry* rtp_payload_registry_; | 161 RTPPayloadRegistry* rtp_payload_registry_; |
| 164 RtpReceiver* rtp_receiver_; | 162 RtpReceiver* rtp_receiver_; |
| 165 RtpRtcp* module_; | 163 RtpRtcp* module_; |
| 166 std::set<uint16_t> expected_sequence_numbers_; | 164 std::set<uint16_t> expected_sequence_numbers_; |
| 167 }; | 165 }; |
| 168 | 166 |
| 169 class RtpRtcpRtxNackTest : public ::testing::Test { | 167 class RtpRtcpRtxNackTest : public ::testing::Test { |
| 170 protected: | 168 protected: |
| 171 RtpRtcpRtxNackTest() | 169 RtpRtcpRtxNackTest() |
| 172 : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)), | 170 : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)), |
| 173 rtp_rtcp_module_(nullptr), | 171 rtp_rtcp_module_(NULL), |
| 174 transport_(kTestSsrc + 1), | 172 transport_(kTestSsrc + 1), |
| 175 receiver_(), | 173 receiver_(), |
| 176 payload_data_length(sizeof(payload_data)), | 174 payload_data_length(sizeof(payload_data)), |
| 177 fake_clock(123456), | 175 fake_clock(123456) {} |
| 178 retranmission_rate_limiter_(&fake_clock, kMaxRttMs) {} | |
| 179 ~RtpRtcpRtxNackTest() {} | 176 ~RtpRtcpRtxNackTest() {} |
| 180 | 177 |
| 181 void SetUp() override { | 178 void SetUp() override { |
| 182 RtpRtcp::Configuration configuration; | 179 RtpRtcp::Configuration configuration; |
| 183 configuration.audio = false; | 180 configuration.audio = false; |
| 184 configuration.clock = &fake_clock; | 181 configuration.clock = &fake_clock; |
| 185 receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock)); | 182 receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock)); |
| 186 configuration.receive_statistics = receive_statistics_.get(); | 183 configuration.receive_statistics = receive_statistics_.get(); |
| 187 configuration.outgoing_transport = &transport_; | 184 configuration.outgoing_transport = &transport_; |
| 188 configuration.retransmission_rate_limiter = &retranmission_rate_limiter_; | |
| 189 rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration); | 185 rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration); |
| 190 | 186 |
| 191 rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_)); | 187 rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_)); |
| 192 | 188 |
| 193 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( | 189 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( |
| 194 &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_)); | 190 &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_)); |
| 195 | 191 |
| 196 rtp_rtcp_module_->SetSSRC(kTestSsrc); | 192 rtp_rtcp_module_->SetSSRC(kTestSsrc); |
| 197 rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound); | 193 rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound); |
| 198 rtp_rtcp_module_->SetStorePacketsStatus(true, 600); | 194 rtp_rtcp_module_->SetStorePacketsStatus(true, 600); |
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| 285 std::unique_ptr<ReceiveStatistics> receive_statistics_; | 281 std::unique_ptr<ReceiveStatistics> receive_statistics_; |
| 286 RTPPayloadRegistry rtp_payload_registry_; | 282 RTPPayloadRegistry rtp_payload_registry_; |
| 287 std::unique_ptr<RtpReceiver> rtp_receiver_; | 283 std::unique_ptr<RtpReceiver> rtp_receiver_; |
| 288 RtpRtcp* rtp_rtcp_module_; | 284 RtpRtcp* rtp_rtcp_module_; |
| 289 std::unique_ptr<TestRtpFeedback> rtp_feedback_; | 285 std::unique_ptr<TestRtpFeedback> rtp_feedback_; |
| 290 RtxLoopBackTransport transport_; | 286 RtxLoopBackTransport transport_; |
| 291 VerifyingRtxReceiver receiver_; | 287 VerifyingRtxReceiver receiver_; |
| 292 uint8_t payload_data[65000]; | 288 uint8_t payload_data[65000]; |
| 293 size_t payload_data_length; | 289 size_t payload_data_length; |
| 294 SimulatedClock fake_clock; | 290 SimulatedClock fake_clock; |
| 295 RateLimiter retranmission_rate_limiter_; | |
| 296 }; | 291 }; |
| 297 | 292 |
| 298 TEST_F(RtpRtcpRtxNackTest, LongNackList) { | 293 TEST_F(RtpRtcpRtxNackTest, LongNackList) { |
| 299 const int kNumPacketsToDrop = 900; | 294 const int kNumPacketsToDrop = 900; |
| 300 const int kNumRequiredRtcp = 4; | 295 const int kNumRequiredRtcp = 4; |
| 301 uint32_t timestamp = 3000; | 296 uint32_t timestamp = 3000; |
| 302 uint16_t nack_list[kNumPacketsToDrop]; | 297 uint16_t nack_list[kNumPacketsToDrop]; |
| 303 // Disable StorePackets to be able to set a larger packet history. | 298 // Disable StorePackets to be able to set a larger packet history. |
| 304 rtp_rtcp_module_->SetStorePacketsStatus(false, 0); | 299 rtp_rtcp_module_->SetStorePacketsStatus(false, 0); |
| 305 // Enable StorePackets with a packet history of 2000 packets. | 300 // Enable StorePackets with a packet history of 2000 packets. |
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| 337 RunRtxTest(kRtxRetransmitted, 10); | 332 RunRtxTest(kRtxRetransmitted, 10); |
| 338 EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin())); | 333 EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin())); |
| 339 EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1, | 334 EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1, |
| 340 *(receiver_.sequence_numbers_.rbegin())); | 335 *(receiver_.sequence_numbers_.rbegin())); |
| 341 EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size()); | 336 EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size()); |
| 342 EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_); | 337 EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_); |
| 343 EXPECT_TRUE(ExpectedPacketsReceived()); | 338 EXPECT_TRUE(ExpectedPacketsReceived()); |
| 344 } | 339 } |
| 345 | 340 |
| 346 } // namespace webrtc | 341 } // namespace webrtc |
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