1 /* 2 * Copyright (c) 2015 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 <string> 12 13 #include "testing/gtest/include/gtest/gtest.h" 14 15 #include "webrtc/audio/audio_receive_stream.h" 16 #include "webrtc/audio/conversion.h" 17 #include "webrtc/call/mock/mock_congestion_controller.h" 18 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h" 19 #include "webrtc/modules/pacing/packet_router.h" 20 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h" 21 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 22 #include "webrtc/modules/utility/include/mock/mock_process_thread.h" 23 #include "webrtc/system_wrappers/include/clock.h" 24 #include "webrtc/test/mock_voe_channel_proxy.h" 25 #include "webrtc/test/mock_voice_engine.h" 26 #include "webrtc/video/call_stats.h" 27 28 namespace webrtc { 29 namespace test { 30 namespace { 31 32 using testing::_; 33 using testing::Return; 34 35 AudioDecodingCallStats MakeAudioDecodeStatsForTest() { 36 AudioDecodingCallStats audio_decode_stats; 37 audio_decode_stats.calls_to_silence_generator = 234; 38 audio_decode_stats.calls_to_neteq = 567; 39 audio_decode_stats.decoded_normal = 890; 40 audio_decode_stats.decoded_plc = 123; 41 audio_decode_stats.decoded_cng = 456; 42 audio_decode_stats.decoded_plc_cng = 789; 43 return audio_decode_stats; 44 } 45 46 const int kChannelId = 2; 47 const uint32_t kRemoteSsrc = 1234; 48 const uint32_t kLocalSsrc = 5678; 49 const size_t kOneByteExtensionHeaderLength = 4; 50 const size_t kOneByteExtensionLength = 4; 51 const int kAbsSendTimeId = 2; 52 const int kAudioLevelId = 3; 53 const int kTransportSequenceNumberId = 4; 54 const int kJitterBufferDelay = -7; 55 const int kPlayoutBufferDelay = 302; 56 const unsigned int kSpeechOutputLevel = 99; 57 const CallStatistics kCallStats = { 58 345, 678, 901, 234, -12, 3456, 7890, 567, 890, 123}; 59 const CodecInst kCodecInst = { 60 123, "codec_name_recv", 96000, -187, 0, -103}; 61 const NetworkStatistics kNetworkStats = { 62 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0}; 63 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest(); 64 65 struct ConfigHelper { 66 ConfigHelper() 67 : simulated_clock_(123456), 68 call_stats_(&simulated_clock_), 69 congestion_controller_(&process_thread_, 70 &call_stats_, 71 &bitrate_observer_) { 72 using testing::Invoke; 73 74 EXPECT_CALL(voice_engine_, 75 RegisterVoiceEngineObserver(_)).WillOnce(Return(0)); 76 EXPECT_CALL(voice_engine_, 77 DeRegisterVoiceEngineObserver()).WillOnce(Return(0)); 78 AudioState::Config config; 79 config.voice_engine = &voice_engine_; 80 audio_state_ = AudioState::Create(config); 81 82 EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) 83 .WillOnce(Invoke([this](int channel_id) { 84 EXPECT_FALSE(channel_proxy_); 85 channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); 86 EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1); 87 EXPECT_CALL(*channel_proxy_, 88 SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId)) 89 .Times(1); 90 EXPECT_CALL(*channel_proxy_, 91 SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId)) 92 .Times(1); 93 EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects( 94 nullptr, nullptr, &packet_router_)) 95 .Times(1); 96 EXPECT_CALL(congestion_controller_, packet_router()) 97 .WillOnce(Return(&packet_router_)); 98 EXPECT_CALL(*channel_proxy_, 99 SetCongestionControlObjects(nullptr, nullptr, nullptr)) 100 .Times(1); 101 return channel_proxy_; 102 })); 103 stream_config_.voe_channel_id = kChannelId; 104 stream_config_.rtp.local_ssrc = kLocalSsrc; 105 stream_config_.rtp.remote_ssrc = kRemoteSsrc; 106 stream_config_.rtp.extensions.push_back( 107 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); 108 stream_config_.rtp.extensions.push_back( 109 RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId)); 110 } 111 112 MockCongestionController* congestion_controller() { 113 return &congestion_controller_; 114 } 115 MockRemoteBitrateEstimator* remote_bitrate_estimator() { 116 return &remote_bitrate_estimator_; 117 } 118 AudioReceiveStream::Config& config() { return stream_config_; } 119 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } 120 MockVoiceEngine& voice_engine() { return voice_engine_; } 121 122 void SetupMockForBweFeedback(bool send_side_bwe) { 123 EXPECT_CALL(congestion_controller_, 124 GetRemoteBitrateEstimator(send_side_bwe)) 125 .WillOnce(Return(&remote_bitrate_estimator_)); 126 EXPECT_CALL(remote_bitrate_estimator_, 127 RemoveStream(stream_config_.rtp.remote_ssrc)); 128 } 129 130 void SetupMockForGetStats() { 131 using testing::DoAll; 132 using testing::SetArgReferee; 133 134 ASSERT_TRUE(channel_proxy_); 135 EXPECT_CALL(*channel_proxy_, GetRTCPStatistics()) 136 .WillOnce(Return(kCallStats)); 137 EXPECT_CALL(*channel_proxy_, GetDelayEstimate()) 138 .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay)); 139 EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange()) 140 .WillOnce(Return(kSpeechOutputLevel)); 141 EXPECT_CALL(*channel_proxy_, GetNetworkStatistics()) 142 .WillOnce(Return(kNetworkStats)); 143 EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics()) 144 .WillOnce(Return(kAudioDecodeStats)); 145 146 EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _)) 147 .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0))); 148 } 149 150 private: 151 SimulatedClock simulated_clock_; 152 CallStats call_stats_; 153 PacketRouter packet_router_; 154 testing::NiceMock<MockBitrateObserver> bitrate_observer_; 155 testing::NiceMock<MockProcessThread> process_thread_; 156 MockCongestionController congestion_controller_; 157 MockRemoteBitrateEstimator remote_bitrate_estimator_; 158 testing::StrictMock<MockVoiceEngine> voice_engine_; 159 rtc::scoped_refptr<AudioState> audio_state_; 160 AudioReceiveStream::Config stream_config_; 161 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; 162 }; 163 164 void BuildOneByteExtension(std::vector<uint8_t>::iterator it, 165 int id, 166 uint32_t extension_value, 167 size_t value_length) { 168 const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE; 169 ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId); 170 it += 2; 171 172 ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4); 173 it += 2; 174 const size_t kExtensionDataLength = kOneByteExtensionLength - 1; 175 uint32_t shifted_value = extension_value 176 << (8 * (kExtensionDataLength - value_length)); 177 *it = (id << 4) + (value_length - 1); 178 ++it; 179 ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it), 180 shifted_value); 181 } 182 183 std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension( 184 int extension_id, 185 uint32_t extension_value, 186 size_t value_length) { 187 std::vector<uint8_t> header; 188 header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength + 189 kOneByteExtensionLength); 190 header[0] = 0x80; // Version 2. 191 header[0] |= 0x10; // Set extension bit. 192 header[1] = 100; // Payload type. 193 header[1] |= 0x80; // Marker bit is set. 194 ByteWriter<uint16_t>::WriteBigEndian(&header[2], 0x1234); // Sequence number. 195 ByteWriter<uint32_t>::WriteBigEndian(&header[4], 0x5678); // Timestamp. 196 ByteWriter<uint32_t>::WriteBigEndian(&header[8], 0x4321); // SSRC. 197 198 BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id, 199 extension_value, value_length); 200 return header; 201 } 202 } // namespace 203 204 TEST(AudioReceiveStreamTest, ConfigToString) { 205 AudioReceiveStream::Config config; 206 config.rtp.remote_ssrc = kRemoteSsrc; 207 config.rtp.local_ssrc = kLocalSsrc; 208 config.rtp.extensions.push_back( 209 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); 210 config.voe_channel_id = kChannelId; 211 config.combined_audio_video_bwe = true; 212 EXPECT_EQ( 213 "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: " 214 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, " 215 "receive_transport: nullptr, rtcp_send_transport: nullptr, " 216 "voe_channel_id: 2, combined_audio_video_bwe: true}", 217 config.ToString()); 218 } 219 220 TEST(AudioReceiveStreamTest, ConstructDestruct) { 221 ConfigHelper helper; 222 internal::AudioReceiveStream recv_stream( 223 helper.congestion_controller(), helper.config(), helper.audio_state()); 224 } 225 226 MATCHER_P(VerifyHeaderExtension, expected_extension, "") { 227 return arg.extension.hasAbsoluteSendTime == 228 expected_extension.hasAbsoluteSendTime && 229 arg.extension.absoluteSendTime == 230 expected_extension.absoluteSendTime && 231 arg.extension.hasTransportSequenceNumber == 232 expected_extension.hasTransportSequenceNumber && 233 arg.extension.transportSequenceNumber == 234 expected_extension.transportSequenceNumber; 235 } 236 237 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) { 238 ConfigHelper helper; 239 helper.config().combined_audio_video_bwe = true; 240 helper.SetupMockForBweFeedback(false); 241 internal::AudioReceiveStream recv_stream( 242 helper.congestion_controller(), helper.config(), helper.audio_state()); 243 const int kAbsSendTimeValue = 1234; 244 std::vector<uint8_t> rtp_packet = 245 CreateRtpHeaderWithOneByteExtension(kAbsSendTimeId, kAbsSendTimeValue, 3); 246 PacketTime packet_time(5678000, 0); 247 const size_t kExpectedHeaderLength = 20; 248 RTPHeaderExtension expected_extension; 249 expected_extension.hasAbsoluteSendTime = true; 250 expected_extension.absoluteSendTime = kAbsSendTimeValue; 251 EXPECT_CALL(*helper.remote_bitrate_estimator(), 252 IncomingPacket(packet_time.timestamp / 1000, 253 rtp_packet.size() - kExpectedHeaderLength, 254 VerifyHeaderExtension(expected_extension), false)) 255 .Times(1); 256 EXPECT_TRUE( 257 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); 258 } 259 260 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) { 261 ConfigHelper helper; 262 helper.config().combined_audio_video_bwe = true; 263 helper.config().rtp.transport_cc = true; 264 helper.config().rtp.extensions.push_back(RtpExtension( 265 RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId)); 266 helper.SetupMockForBweFeedback(true); 267 internal::AudioReceiveStream recv_stream( 268 helper.congestion_controller(), helper.config(), helper.audio_state()); 269 const int kTransportSequenceNumberValue = 1234; 270 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( 271 kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); 272 PacketTime packet_time(5678000, 0); 273 const size_t kExpectedHeaderLength = 20; 274 RTPHeaderExtension expected_extension; 275 expected_extension.hasTransportSequenceNumber = true; 276 expected_extension.transportSequenceNumber = kTransportSequenceNumberValue; 277 EXPECT_CALL(*helper.remote_bitrate_estimator(), 278 IncomingPacket(packet_time.timestamp / 1000, 279 rtp_packet.size() - kExpectedHeaderLength, 280 VerifyHeaderExtension(expected_extension), false)) 281 .Times(1); 282 EXPECT_TRUE( 283 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); 284 } 285 286 TEST(AudioReceiveStreamTest, GetStats) { 287 ConfigHelper helper; 288 internal::AudioReceiveStream recv_stream( 289 helper.congestion_controller(), helper.config(), helper.audio_state()); 290 helper.SetupMockForGetStats(); 291 AudioReceiveStream::Stats stats = recv_stream.GetStats(); 292 EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); 293 EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd); 294 EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived), 295 stats.packets_rcvd); 296 EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost); 297 EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost); 298 EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name); 299 EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum); 300 EXPECT_EQ(kCallStats.jitterSamples / (kCodecInst.plfreq / 1000), 301 stats.jitter_ms); 302 EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms); 303 EXPECT_EQ(kNetworkStats.preferredBufferSize, 304 stats.jitter_buffer_preferred_ms); 305 EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay), 306 stats.delay_estimate_ms); 307 EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level); 308 EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate); 309 EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate), 310 stats.speech_expand_rate); 311 EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate), 312 stats.secondary_decoded_rate); 313 EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate), 314 stats.accelerate_rate); 315 EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate), 316 stats.preemptive_expand_rate); 317 EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator, 318 stats.decoding_calls_to_silence_generator); 319 EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq); 320 EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal); 321 EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc); 322 EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng); 323 EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng); 324 EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, 325 stats.capture_start_ntp_time_ms); 326 } 327 } // namespace test 328 } // namespace webrtc 329
