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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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