1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #include "webrtc/modules/interface/module_common_types.h" 12 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 13 #include "webrtc/system_wrappers/interface/atomic32.h" 14 #include "webrtc/system_wrappers/interface/sleep.h" 15 #include "webrtc/video_engine/include/vie_network.h" 16 #include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h" 17 18 using ::testing::_; 19 using ::testing::AtLeast; 20 using ::testing::Eq; 21 using ::testing::Field; 22 23 class ExtensionVerifyTransport : public webrtc::Transport { 24 public: 25 ExtensionVerifyTransport() 26 : parser_(webrtc::RtpHeaderParser::Create()), 27 received_packets_(0), 28 bad_packets_(0), 29 audio_level_id_(-1), 30 absolute_sender_time_id_(-1) {} 31 32 virtual int SendPacket(int channel, const void* data, int len) { 33 webrtc::RTPHeader header; 34 if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), 35 static_cast<size_t>(len), 36 &header)) { 37 bool ok = true; 38 if (audio_level_id_ >= 0 && 39 !header.extension.hasAudioLevel) { 40 ok = false; 41 } 42 if (absolute_sender_time_id_ >= 0 && 43 !header.extension.hasAbsoluteSendTime) { 44 ok = false; 45 } 46 if (!ok) { 47 // bad_packets_ count packets we expected to have an extension but 48 // didn't have one. 49 ++bad_packets_; 50 } 51 } 52 // received_packets_ count all packets we receive. 53 ++received_packets_; 54 return len; 55 } 56 57 virtual int SendRTCPPacket(int channel, const void* data, int len) { 58 return len; 59 } 60 61 void SetAudioLevelId(int id) { 62 audio_level_id_ = id; 63 parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, id); 64 } 65 66 void SetAbsoluteSenderTimeId(int id) { 67 absolute_sender_time_id_ = id; 68 parser_->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAbsoluteSendTime, 69 id); 70 } 71 72 bool Wait() { 73 // Wait until we've received to specified number of packets. 74 while (received_packets_.Value() < kPacketsExpected) { 75 webrtc::SleepMs(kSleepIntervalMs); 76 } 77 // Check whether any where 'bad' (didn't contain an extension when they 78 // where supposed to). 79 return bad_packets_.Value() == 0; 80 } 81 82 private: 83 enum { 84 kPacketsExpected = 10, 85 kSleepIntervalMs = 10 86 }; 87 webrtc::scoped_ptr<webrtc::RtpHeaderParser> parser_; 88 webrtc::Atomic32 received_packets_; 89 webrtc::Atomic32 bad_packets_; 90 int audio_level_id_; 91 int absolute_sender_time_id_; 92 }; 93 94 class SendRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture { 95 protected: 96 virtual void SetUp() { 97 EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_)); 98 EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, 99 verifying_transport_)); 100 } 101 virtual void TearDown() { 102 PausePlaying(); 103 } 104 105 ExtensionVerifyTransport verifying_transport_; 106 }; 107 108 TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeNoAudioLevel) { 109 verifying_transport_.SetAudioLevelId(0); 110 ResumePlaying(); 111 EXPECT_FALSE(verifying_transport_.Wait()); 112 } 113 114 TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAudioLevel) { 115 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true, 116 9)); 117 verifying_transport_.SetAudioLevelId(9); 118 ResumePlaying(); 119 EXPECT_TRUE(verifying_transport_.Wait()); 120 } 121 122 TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeNoAbsoluteSenderTime) 123 { 124 verifying_transport_.SetAbsoluteSenderTimeId(0); 125 ResumePlaying(); 126 EXPECT_FALSE(verifying_transport_.Wait()); 127 } 128 129 TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAbsoluteSenderTime) { 130 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true, 131 11)); 132 verifying_transport_.SetAbsoluteSenderTimeId(11); 133 ResumePlaying(); 134 EXPECT_TRUE(verifying_transport_.Wait()); 135 } 136 137 TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAllExtensions1) { 138 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true, 139 9)); 140 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true, 141 11)); 142 verifying_transport_.SetAudioLevelId(9); 143 verifying_transport_.SetAbsoluteSenderTimeId(11); 144 ResumePlaying(); 145 EXPECT_TRUE(verifying_transport_.Wait()); 146 } 147 148 TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAllExtensions2) { 149 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true, 150 3)); 151 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true, 152 9)); 153 verifying_transport_.SetAbsoluteSenderTimeId(3); 154 // Don't register audio level with header parser - unknown extensions should 155 // be ignored when parsing. 156 ResumePlaying(); 157 EXPECT_TRUE(verifying_transport_.Wait()); 158 } 159 160 class MockViENetwork : public webrtc::ViENetwork { 161 public: 162 MockViENetwork() {} 163 virtual ~MockViENetwork() {} 164 165 MOCK_METHOD0(Release, int()); 166 MOCK_METHOD2(SetNetworkTransmissionState, void(const int, const bool)); 167 MOCK_METHOD2(RegisterSendTransport, int(const int, webrtc::Transport&)); 168 MOCK_METHOD1(DeregisterSendTransport, int(const int)); 169 MOCK_METHOD4(ReceivedRTPPacket, int(const int, const void*, const int, 170 const webrtc::PacketTime&)); 171 MOCK_METHOD3(ReceivedRTCPPacket, int(const int, const void*, const int)); 172 MOCK_METHOD2(SetMTU, int(int, unsigned int)); 173 MOCK_METHOD4(ReceivedBWEPacket, int(const int, int64_t, int, 174 const webrtc::RTPHeader&)); 175 }; 176 177 class ReceiveRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture { 178 protected: 179 virtual void SetUp() { 180 EXPECT_EQ(0, 181 voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true, 11)); 182 EXPECT_EQ(0, 183 voe_rtp_rtcp_->SetReceiveAbsoluteSenderTimeStatus(channel_, true, 11)); 184 } 185 186 enum { 187 kVideoChannelId1 = 667, 188 kVideoChannelId2 = 668 189 }; 190 MockViENetwork mock_network_; 191 }; 192 193 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTDisabled) { 194 ResumePlaying(); 195 Sleep(500); 196 } 197 198 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTFailSetTarget) { 199 EXPECT_CALL(mock_network_, Release()).Times(1); 200 EXPECT_EQ(-1, voe_rtp_rtcp_->SetVideoEngineBWETarget(-1, &mock_network_, 201 kVideoChannelId1)); 202 ResumePlaying(); 203 } 204 205 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabled) { 206 EXPECT_CALL(mock_network_, Release()).Times(1); 207 EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _, 208 Field(&webrtc::RTPHeader::extension, 209 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true))))) 210 .Times(AtLeast(1)); 211 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_, 212 kVideoChannelId1)); 213 ResumePlaying(); 214 Sleep(500); 215 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1)); 216 } 217 218 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledBadExtensionId) { 219 EXPECT_CALL(mock_network_, Release()).Times(1); 220 EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _, 221 Field(&webrtc::RTPHeader::extension, 222 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(false))))) 223 .Times(AtLeast(1)); 224 EXPECT_EQ(0, voe_rtp_rtcp_->SetReceiveAbsoluteSenderTimeStatus(channel_, true, 225 1)); 226 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_, 227 kVideoChannelId1)); 228 ResumePlaying(); 229 Sleep(500); 230 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1)); 231 } 232 233 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledNotSending) { 234 EXPECT_CALL(mock_network_, Release()).Times(1); 235 EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _, 236 Field(&webrtc::RTPHeader::extension, 237 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(false))))) 238 .Times(AtLeast(1)); 239 EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, false, 240 11)); 241 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_, 242 kVideoChannelId1)); 243 ResumePlaying(); 244 Sleep(500); 245 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1)); 246 } 247 248 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledNotReceiving) { 249 EXPECT_CALL(mock_network_, Release()).Times(1); 250 EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _, 251 Field(&webrtc::RTPHeader::extension, 252 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(false))))) 253 .Times(AtLeast(1)); 254 EXPECT_EQ(0, voe_rtp_rtcp_->SetReceiveAbsoluteSenderTimeStatus(channel_, 255 false, 11)); 256 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_, 257 kVideoChannelId1)); 258 ResumePlaying(); 259 Sleep(500); 260 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1)); 261 } 262 263 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTSwitchViENetwork) { 264 MockViENetwork mock_network_2; 265 EXPECT_CALL(mock_network_2, Release()).Times(1); 266 EXPECT_CALL(mock_network_2, ReceivedBWEPacket(kVideoChannelId1, _, _, 267 Field(&webrtc::RTPHeader::extension, 268 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true))))) 269 .Times(AtLeast(1)); 270 EXPECT_CALL(mock_network_, Release()).Times(1); 271 EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _, 272 Field(&webrtc::RTPHeader::extension, 273 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true))))) 274 .Times(AtLeast(1)); 275 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_2, 276 kVideoChannelId1)); 277 ResumePlaying(); 278 Sleep(500); 279 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_, 280 kVideoChannelId1)); 281 Sleep(500); 282 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1)); 283 } 284 285 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTSwitchVideoChannel) { 286 EXPECT_CALL(mock_network_, Release()).Times(2); 287 EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _, 288 Field(&webrtc::RTPHeader::extension, 289 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true))))) 290 .Times(AtLeast(1)); 291 EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId2, _, _, 292 Field(&webrtc::RTPHeader::extension, 293 Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true))))) 294 .Times(AtLeast(1)); 295 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_, 296 kVideoChannelId1)); 297 ResumePlaying(); 298 Sleep(500); 299 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_, 300 kVideoChannelId2)); 301 Sleep(500); 302 EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1)); 303 } 304
