Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(111)

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

Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed nit Created 4 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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"
18 #include "webrtc/common_types.h" 19 #include "webrtc/common_types.h"
19 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" 20 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
20 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 21 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
21 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 23 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
25 #include "webrtc/transport.h" 26 #include "webrtc/transport.h"
26 27
27 namespace webrtc { 28 namespace webrtc {
28 29
29 const int kVideoNackListSize = 30; 30 const int kVideoNackListSize = 30;
30 const uint32_t kTestSsrc = 3456; 31 const uint32_t kTestSsrc = 3456;
31 const uint16_t kTestSequenceNumber = 2345; 32 const uint16_t kTestSequenceNumber = 2345;
32 const uint32_t kTestNumberOfPackets = 1350; 33 const uint32_t kTestNumberOfPackets = 1350;
33 const int kTestNumberOfRtxPackets = 149; 34 const int kTestNumberOfRtxPackets = 149;
34 const int kNumFrames = 30; 35 const int kNumFrames = 30;
35 const int kPayloadType = 123; 36 const int kPayloadType = 123;
36 const int kRtxPayloadType = 98; 37 const int kRtxPayloadType = 98;
38 const int64_t kMaxRttMs = 1000;
37 39
38 class VerifyingRtxReceiver : public NullRtpData { 40 class VerifyingRtxReceiver : public NullRtpData {
39 public: 41 public:
40 VerifyingRtxReceiver() {} 42 VerifyingRtxReceiver() {}
41 43
42 int32_t OnReceivedPayloadData( 44 int32_t OnReceivedPayloadData(
43 const uint8_t* data, 45 const uint8_t* data,
44 size_t size, 46 size_t size,
45 const webrtc::WebRtcRTPHeader* rtp_header) override { 47 const webrtc::WebRtcRTPHeader* rtp_header) override {
46 if (!sequence_numbers_.empty()) 48 if (!sequence_numbers_.empty())
(...skipping 114 matching lines...) Expand 10 before | Expand all | Expand 10 after
161 RTPPayloadRegistry* rtp_payload_registry_; 163 RTPPayloadRegistry* rtp_payload_registry_;
162 RtpReceiver* rtp_receiver_; 164 RtpReceiver* rtp_receiver_;
163 RtpRtcp* module_; 165 RtpRtcp* module_;
164 std::set<uint16_t> expected_sequence_numbers_; 166 std::set<uint16_t> expected_sequence_numbers_;
165 }; 167 };
166 168
167 class RtpRtcpRtxNackTest : public ::testing::Test { 169 class RtpRtcpRtxNackTest : public ::testing::Test {
168 protected: 170 protected:
169 RtpRtcpRtxNackTest() 171 RtpRtcpRtxNackTest()
170 : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)), 172 : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
171 rtp_rtcp_module_(NULL), 173 rtp_rtcp_module_(nullptr),
172 transport_(kTestSsrc + 1), 174 transport_(kTestSsrc + 1),
173 receiver_(), 175 receiver_(),
174 payload_data_length(sizeof(payload_data)), 176 payload_data_length(sizeof(payload_data)),
175 fake_clock(123456) {} 177 fake_clock(123456),
178 retranmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
176 ~RtpRtcpRtxNackTest() {} 179 ~RtpRtcpRtxNackTest() {}
177 180
178 void SetUp() override { 181 void SetUp() override {
179 RtpRtcp::Configuration configuration; 182 RtpRtcp::Configuration configuration;
180 configuration.audio = false; 183 configuration.audio = false;
181 configuration.clock = &fake_clock; 184 configuration.clock = &fake_clock;
182 receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock)); 185 receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
183 configuration.receive_statistics = receive_statistics_.get(); 186 configuration.receive_statistics = receive_statistics_.get();
184 configuration.outgoing_transport = &transport_; 187 configuration.outgoing_transport = &transport_;
188 configuration.retransmission_rate_limiter = &retranmission_rate_limiter_;
185 rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration); 189 rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
186 190
187 rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_)); 191 rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
188 192
189 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( 193 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
190 &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_)); 194 &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_));
191 195
192 rtp_rtcp_module_->SetSSRC(kTestSsrc); 196 rtp_rtcp_module_->SetSSRC(kTestSsrc);
193 rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound); 197 rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
194 rtp_rtcp_module_->SetStorePacketsStatus(true, 600); 198 rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
(...skipping 86 matching lines...) Expand 10 before | Expand all | Expand 10 after
281 std::unique_ptr<ReceiveStatistics> receive_statistics_; 285 std::unique_ptr<ReceiveStatistics> receive_statistics_;
282 RTPPayloadRegistry rtp_payload_registry_; 286 RTPPayloadRegistry rtp_payload_registry_;
283 std::unique_ptr<RtpReceiver> rtp_receiver_; 287 std::unique_ptr<RtpReceiver> rtp_receiver_;
284 RtpRtcp* rtp_rtcp_module_; 288 RtpRtcp* rtp_rtcp_module_;
285 std::unique_ptr<TestRtpFeedback> rtp_feedback_; 289 std::unique_ptr<TestRtpFeedback> rtp_feedback_;
286 RtxLoopBackTransport transport_; 290 RtxLoopBackTransport transport_;
287 VerifyingRtxReceiver receiver_; 291 VerifyingRtxReceiver receiver_;
288 uint8_t payload_data[65000]; 292 uint8_t payload_data[65000];
289 size_t payload_data_length; 293 size_t payload_data_length;
290 SimulatedClock fake_clock; 294 SimulatedClock fake_clock;
295 RateLimiter retranmission_rate_limiter_;
291 }; 296 };
292 297
293 TEST_F(RtpRtcpRtxNackTest, LongNackList) { 298 TEST_F(RtpRtcpRtxNackTest, LongNackList) {
294 const int kNumPacketsToDrop = 900; 299 const int kNumPacketsToDrop = 900;
295 const int kNumRequiredRtcp = 4; 300 const int kNumRequiredRtcp = 4;
296 uint32_t timestamp = 3000; 301 uint32_t timestamp = 3000;
297 uint16_t nack_list[kNumPacketsToDrop]; 302 uint16_t nack_list[kNumPacketsToDrop];
298 // Disable StorePackets to be able to set a larger packet history. 303 // Disable StorePackets to be able to set a larger packet history.
299 rtp_rtcp_module_->SetStorePacketsStatus(false, 0); 304 rtp_rtcp_module_->SetStorePacketsStatus(false, 0);
300 // Enable StorePackets with a packet history of 2000 packets. 305 // Enable StorePackets with a packet history of 2000 packets.
(...skipping 31 matching lines...) Expand 10 before | Expand all | Expand 10 after
332 RunRtxTest(kRtxRetransmitted, 10); 337 RunRtxTest(kRtxRetransmitted, 10);
333 EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin())); 338 EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
334 EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1, 339 EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
335 *(receiver_.sequence_numbers_.rbegin())); 340 *(receiver_.sequence_numbers_.rbegin()));
336 EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size()); 341 EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
337 EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_); 342 EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
338 EXPECT_TRUE(ExpectedPacketsReceived()); 343 EXPECT_TRUE(ExpectedPacketsReceived());
339 } 344 }
340 345
341 } // namespace webrtc 346 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/bitrate.cc ('k') | webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698