pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2013 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 | */ |
henrik.lundin@webrtc.org | d4ec1f5 | 2013-10-28 10:16:14 +0000 | [diff] [blame^] | 10 | #include <algorithm> // max |
| 11 | |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 12 | #include "testing/gtest/include/gtest/gtest.h" |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 13 | #include "webrtc/common_video/interface/i420_video_frame.h" |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 14 | #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 15 | #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 16 | #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 17 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 18 | #include "webrtc/system_wrappers/interface/event_wrapper.h" |
| 19 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
pbos@webrtc.org | 905cebd | 2013-09-11 10:14:56 +0000 | [diff] [blame] | 20 | #include "webrtc/system_wrappers/interface/sleep.h" |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 21 | #include "webrtc/system_wrappers/interface/thread_wrapper.h" |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 22 | #include "webrtc/video_engine/internal/transport_adapter.h" |
| 23 | #include "webrtc/video_engine/new_include/call.h" |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 24 | #include "webrtc/video_engine/new_include/frame_callback.h" |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 25 | #include "webrtc/video_engine/new_include/video_send_stream.h" |
| 26 | #include "webrtc/video_engine/test/common/direct_transport.h" |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 27 | #include "webrtc/video_engine/test/common/fake_encoder.h" |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 28 | #include "webrtc/video_engine/test/common/frame_generator_capturer.h" |
| 29 | #include "webrtc/video_engine/test/common/null_transport.h" |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 30 | |
| 31 | namespace webrtc { |
| 32 | |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 33 | class SendTransportObserver : public test::NullTransport { |
| 34 | public: |
| 35 | explicit SendTransportObserver(unsigned long timeout_ms) |
| 36 | : rtp_header_parser_(RtpHeaderParser::Create()), |
| 37 | send_test_complete_(EventWrapper::Create()), |
| 38 | timeout_ms_(timeout_ms) {} |
| 39 | |
pbos@webrtc.org | fdc4352 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 40 | EventTypeWrapper Wait() { return send_test_complete_->Wait(timeout_ms_); } |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 41 | |
| 42 | protected: |
| 43 | scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
| 44 | scoped_ptr<EventWrapper> send_test_complete_; |
| 45 | |
| 46 | private: |
| 47 | unsigned long timeout_ms_; |
| 48 | }; |
| 49 | |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 50 | class VideoSendStreamTest : public ::testing::Test { |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 51 | public: |
| 52 | VideoSendStreamTest() : fake_encoder_(Clock::GetRealTimeClock()) {} |
pbos@webrtc.org | fdc4352 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 53 | |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 54 | protected: |
pbos@webrtc.org | fdc4352 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 55 | void RunSendTest(Call* call, |
pbos@webrtc.org | d8e92c9 | 2013-08-23 09:19:30 +0000 | [diff] [blame] | 56 | const VideoSendStream::Config& config, |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 57 | SendTransportObserver* observer) { |
pbos@webrtc.org | d8e92c9 | 2013-08-23 09:19:30 +0000 | [diff] [blame] | 58 | VideoSendStream* send_stream = call->CreateSendStream(config); |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 59 | scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer( |
| 60 | test::FrameGeneratorCapturer::Create( |
andresp@webrtc.org | 9612f5a | 2013-09-19 12:14:03 +0000 | [diff] [blame] | 61 | send_stream->Input(), 320, 240, 30, Clock::GetRealTimeClock())); |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 62 | send_stream->StartSend(); |
| 63 | frame_generator_capturer->Start(); |
| 64 | |
| 65 | EXPECT_EQ(kEventSignaled, observer->Wait()); |
| 66 | |
| 67 | frame_generator_capturer->Stop(); |
| 68 | send_stream->StopSend(); |
| 69 | call->DestroySendStream(send_stream); |
| 70 | } |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 71 | |
pbos@webrtc.org | fdc4352 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 72 | VideoSendStream::Config GetSendTestConfig(Call* call) { |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 73 | VideoSendStream::Config config = call->GetDefaultSendConfig(); |
| 74 | config.encoder = &fake_encoder_; |
| 75 | config.internal_source = false; |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 76 | config.rtp.ssrcs.push_back(kSendSsrc); |
pbos@webrtc.org | 0ee03f9 | 2013-09-09 08:26:30 +0000 | [diff] [blame] | 77 | test::FakeEncoder::SetCodecSettings(&config.codec, 1); |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 78 | return config; |
| 79 | } |
| 80 | |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 81 | void TestNackRetransmission(uint32_t retransmit_ssrc); |
| 82 | |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 83 | static const uint32_t kSendSsrc; |
| 84 | static const uint32_t kSendRtxSsrc; |
| 85 | |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 86 | test::FakeEncoder fake_encoder_; |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 87 | }; |
| 88 | |
| 89 | const uint32_t VideoSendStreamTest::kSendSsrc = 0xC0FFEE; |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 90 | const uint32_t VideoSendStreamTest::kSendRtxSsrc = 0xBADCAFE; |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 91 | |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 92 | TEST_F(VideoSendStreamTest, SendsSetSsrc) { |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 93 | class SendSsrcObserver : public SendTransportObserver { |
| 94 | public: |
| 95 | SendSsrcObserver() : SendTransportObserver(30 * 1000) {} |
| 96 | |
| 97 | virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
| 98 | RTPHeader header; |
| 99 | EXPECT_TRUE( |
| 100 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 101 | |
| 102 | if (header.ssrc == kSendSsrc) |
| 103 | send_test_complete_->Set(); |
| 104 | |
| 105 | return true; |
| 106 | } |
| 107 | } observer; |
| 108 | |
pbos@webrtc.org | fdc4352 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 109 | Call::Config call_config(&observer); |
| 110 | scoped_ptr<Call> call(Call::Create(call_config)); |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 111 | |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 112 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
sprang@webrtc.org | 4fe8543 | 2013-10-16 11:37:54 +0000 | [diff] [blame] | 113 | send_config.rtp.max_packet_size = 128; |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 114 | |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 115 | RunSendTest(call.get(), send_config, &observer); |
| 116 | } |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 117 | |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 118 | TEST_F(VideoSendStreamTest, SupportsCName) { |
| 119 | static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo="; |
| 120 | class CNameObserver : public SendTransportObserver { |
| 121 | public: |
| 122 | CNameObserver() : SendTransportObserver(30 * 1000) {} |
| 123 | |
| 124 | virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE { |
| 125 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 126 | EXPECT_TRUE(parser.IsValid()); |
| 127 | |
| 128 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 129 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 130 | if (packet_type == RTCPUtility::kRtcpSdesChunkCode) { |
| 131 | EXPECT_EQ(parser.Packet().CName.CName, kCName); |
| 132 | send_test_complete_->Set(); |
| 133 | } |
| 134 | |
| 135 | packet_type = parser.Iterate(); |
| 136 | } |
| 137 | |
| 138 | return true; |
| 139 | } |
| 140 | } observer; |
| 141 | |
pbos@webrtc.org | fdc4352 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 142 | Call::Config call_config(&observer); |
| 143 | scoped_ptr<Call> call(Call::Create(call_config)); |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 144 | |
pbos@webrtc.org | 43ec357 | 2013-09-03 09:10:37 +0000 | [diff] [blame] | 145 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
pbos@webrtc.org | debc672 | 2013-08-22 09:42:17 +0000 | [diff] [blame] | 146 | send_config.rtp.c_name = kCName; |
| 147 | |
| 148 | RunSendTest(call.get(), send_config, &observer); |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 149 | } |
| 150 | |
pbos@webrtc.org | e22b761 | 2013-09-11 19:00:39 +0000 | [diff] [blame] | 151 | TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) { |
| 152 | static const uint8_t kAbsSendTimeExtensionId = 13; |
| 153 | class AbsoluteSendTimeObserver : public SendTransportObserver { |
| 154 | public: |
| 155 | AbsoluteSendTimeObserver() : SendTransportObserver(30 * 1000) { |
| 156 | EXPECT_TRUE(rtp_header_parser_->RegisterRtpHeaderExtension( |
| 157 | kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId)); |
| 158 | } |
| 159 | |
| 160 | virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
| 161 | RTPHeader header; |
| 162 | EXPECT_TRUE( |
| 163 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 164 | |
| 165 | if (header.extension.absoluteSendTime > 0) |
| 166 | send_test_complete_->Set(); |
| 167 | |
| 168 | return true; |
| 169 | } |
| 170 | } observer; |
| 171 | |
| 172 | Call::Config call_config(&observer); |
| 173 | scoped_ptr<Call> call(Call::Create(call_config)); |
| 174 | |
| 175 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| 176 | send_config.rtp.extensions.push_back( |
| 177 | RtpExtension("abs-send-time", kAbsSendTimeExtensionId)); |
| 178 | |
| 179 | RunSendTest(call.get(), send_config, &observer); |
| 180 | } |
| 181 | |
pbos@webrtc.org | 905cebd | 2013-09-11 10:14:56 +0000 | [diff] [blame] | 182 | TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) { |
| 183 | static const uint8_t kTOffsetExtensionId = 13; |
| 184 | class DelayedEncoder : public test::FakeEncoder { |
| 185 | public: |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 186 | explicit DelayedEncoder(Clock* clock) : test::FakeEncoder(clock) {} |
pbos@webrtc.org | 905cebd | 2013-09-11 10:14:56 +0000 | [diff] [blame] | 187 | virtual int32_t Encode( |
| 188 | const I420VideoFrame& input_image, |
| 189 | const CodecSpecificInfo* codec_specific_info, |
| 190 | const std::vector<VideoFrameType>* frame_types) OVERRIDE { |
| 191 | // A delay needs to be introduced to assure that we get a timestamp |
| 192 | // offset. |
| 193 | SleepMs(5); |
| 194 | return FakeEncoder::Encode(input_image, codec_specific_info, frame_types); |
| 195 | } |
| 196 | } encoder(Clock::GetRealTimeClock()); |
| 197 | |
| 198 | class TransmissionTimeOffsetObserver : public SendTransportObserver { |
| 199 | public: |
| 200 | TransmissionTimeOffsetObserver() : SendTransportObserver(30 * 1000) { |
| 201 | EXPECT_TRUE(rtp_header_parser_->RegisterRtpHeaderExtension( |
| 202 | kRtpExtensionTransmissionTimeOffset, kTOffsetExtensionId)); |
| 203 | } |
| 204 | |
| 205 | virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
| 206 | RTPHeader header; |
| 207 | EXPECT_TRUE( |
| 208 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 209 | |
| 210 | EXPECT_GT(header.extension.transmissionTimeOffset, 0); |
| 211 | send_test_complete_->Set(); |
| 212 | |
| 213 | return true; |
| 214 | } |
| 215 | } observer; |
| 216 | |
| 217 | Call::Config call_config(&observer); |
| 218 | scoped_ptr<Call> call(Call::Create(call_config)); |
| 219 | |
| 220 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| 221 | send_config.encoder = &encoder; |
| 222 | send_config.rtp.extensions.push_back( |
| 223 | RtpExtension("toffset", kTOffsetExtensionId)); |
| 224 | |
| 225 | RunSendTest(call.get(), send_config, &observer); |
| 226 | } |
| 227 | |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 228 | class FakeReceiveStatistics : public NullReceiveStatistics { |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 229 | public: |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 230 | FakeReceiveStatistics(uint32_t send_ssrc, |
| 231 | uint32_t last_sequence_number, |
| 232 | uint32_t cumulative_lost, |
| 233 | uint8_t fraction_lost) |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 234 | : lossy_stats_(new LossyStatistician(last_sequence_number, |
| 235 | cumulative_lost, |
| 236 | fraction_lost)) { |
| 237 | stats_map_[send_ssrc] = lossy_stats_.get(); |
| 238 | } |
| 239 | |
| 240 | virtual StatisticianMap GetActiveStatisticians() const OVERRIDE { |
| 241 | return stats_map_; |
| 242 | } |
| 243 | |
| 244 | virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE { |
| 245 | return lossy_stats_.get(); |
| 246 | } |
| 247 | |
| 248 | private: |
| 249 | class LossyStatistician : public StreamStatistician { |
| 250 | public: |
| 251 | LossyStatistician(uint32_t extended_max_sequence_number, |
| 252 | uint32_t cumulative_lost, |
| 253 | uint8_t fraction_lost) { |
| 254 | stats_.fraction_lost = fraction_lost; |
| 255 | stats_.cumulative_lost = cumulative_lost; |
| 256 | stats_.extended_max_sequence_number = extended_max_sequence_number; |
| 257 | } |
| 258 | virtual bool GetStatistics(Statistics* statistics, bool reset) OVERRIDE { |
| 259 | *statistics = stats_; |
| 260 | return true; |
| 261 | } |
| 262 | virtual void GetDataCounters(uint32_t* bytes_received, |
| 263 | uint32_t* packets_received) const OVERRIDE { |
| 264 | *bytes_received = 0; |
| 265 | *packets_received = 0; |
| 266 | } |
| 267 | virtual uint32_t BitrateReceived() const OVERRIDE { return 0; } |
| 268 | virtual void ResetStatistics() OVERRIDE {} |
| 269 | virtual bool IsRetransmitOfOldPacket(const RTPHeader& header, |
| 270 | int min_rtt) const OVERRIDE { |
| 271 | return false; |
| 272 | } |
| 273 | |
| 274 | virtual bool IsPacketInOrder(uint16_t sequence_number) const OVERRIDE { |
| 275 | return true; |
| 276 | } |
| 277 | Statistics stats_; |
| 278 | }; |
| 279 | |
| 280 | scoped_ptr<LossyStatistician> lossy_stats_; |
| 281 | StatisticianMap stats_map_; |
| 282 | }; |
| 283 | |
| 284 | TEST_F(VideoSendStreamTest, SupportsFec) { |
| 285 | static const int kRedPayloadType = 118; |
| 286 | static const int kUlpfecPayloadType = 119; |
| 287 | class FecObserver : public SendTransportObserver { |
| 288 | public: |
| 289 | FecObserver() |
| 290 | : SendTransportObserver(30 * 1000), |
| 291 | transport_adapter_(&transport_), |
| 292 | send_count_(0), |
| 293 | received_media_(false), |
| 294 | received_fec_(false) {} |
| 295 | |
| 296 | void SetReceiver(PacketReceiver* receiver) { |
| 297 | transport_.SetReceiver(receiver); |
| 298 | } |
| 299 | |
| 300 | virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
| 301 | RTPHeader header; |
| 302 | EXPECT_TRUE( |
| 303 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 304 | |
| 305 | // Send lossy receive reports to trigger FEC enabling. |
| 306 | if (send_count_++ % 2 != 0) { |
| 307 | // Receive statistics reporting having lost 50% of the packets. |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 308 | FakeReceiveStatistics lossy_receive_stats( |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 309 | kSendSsrc, header.sequenceNumber, send_count_ / 2, 127); |
| 310 | RTCPSender rtcp_sender( |
| 311 | 0, false, Clock::GetRealTimeClock(), &lossy_receive_stats); |
| 312 | EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_)); |
| 313 | |
| 314 | rtcp_sender.SetRTCPStatus(kRtcpNonCompound); |
| 315 | rtcp_sender.SetRemoteSSRC(kSendSsrc); |
| 316 | |
| 317 | RTCPSender::FeedbackState feedback_state; |
| 318 | |
| 319 | EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr)); |
| 320 | } |
| 321 | |
| 322 | EXPECT_EQ(kRedPayloadType, header.payloadType); |
| 323 | |
| 324 | uint8_t encapsulated_payload_type = packet[header.headerLength]; |
| 325 | |
| 326 | if (encapsulated_payload_type == kUlpfecPayloadType) { |
| 327 | received_fec_ = true; |
| 328 | } else { |
| 329 | received_media_ = true; |
| 330 | } |
| 331 | |
| 332 | if (received_media_ && received_fec_) |
| 333 | send_test_complete_->Set(); |
| 334 | |
| 335 | return true; |
| 336 | } |
| 337 | |
| 338 | private: |
| 339 | internal::TransportAdapter transport_adapter_; |
| 340 | test::DirectTransport transport_; |
| 341 | int send_count_; |
| 342 | bool received_media_; |
| 343 | bool received_fec_; |
| 344 | } observer; |
| 345 | |
| 346 | Call::Config call_config(&observer); |
| 347 | scoped_ptr<Call> call(Call::Create(call_config)); |
| 348 | |
| 349 | observer.SetReceiver(call->Receiver()); |
| 350 | |
| 351 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| 352 | send_config.rtp.fec.red_payload_type = kRedPayloadType; |
| 353 | send_config.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| 354 | |
| 355 | RunSendTest(call.get(), send_config, &observer); |
| 356 | } |
| 357 | |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 358 | void VideoSendStreamTest::TestNackRetransmission(uint32_t retransmit_ssrc) { |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 359 | class NackObserver : public SendTransportObserver { |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 360 | public: |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 361 | explicit NackObserver(uint32_t retransmit_ssrc) |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 362 | : SendTransportObserver(30 * 1000), |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 363 | transport_adapter_(&transport_), |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 364 | send_count_(0), |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 365 | retransmit_ssrc_(retransmit_ssrc), |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 366 | nacked_sequence_number_(0) {} |
| 367 | |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 368 | void SetReceiver(PacketReceiver* receiver) { |
| 369 | transport_.SetReceiver(receiver); |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 370 | } |
| 371 | |
| 372 | virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 373 | RTPHeader header; |
| 374 | EXPECT_TRUE( |
| 375 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 376 | |
| 377 | // Nack second packet after receiving the third one. |
| 378 | if (++send_count_ == 3) { |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 379 | nacked_sequence_number_ = header.sequenceNumber - 1; |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 380 | NullReceiveStatistics null_stats; |
| 381 | RTCPSender rtcp_sender( |
| 382 | 0, false, Clock::GetRealTimeClock(), &null_stats); |
| 383 | EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_)); |
| 384 | |
| 385 | rtcp_sender.SetRTCPStatus(kRtcpNonCompound); |
| 386 | rtcp_sender.SetRemoteSSRC(kSendSsrc); |
| 387 | |
| 388 | RTCPSender::FeedbackState feedback_state; |
| 389 | |
| 390 | EXPECT_EQ(0, |
| 391 | rtcp_sender.SendRTCP( |
| 392 | feedback_state, kRtcpNack, 1, &nacked_sequence_number_)); |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 393 | } |
| 394 | |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 395 | uint16_t sequence_number = header.sequenceNumber; |
| 396 | |
| 397 | if (header.ssrc == retransmit_ssrc_ && retransmit_ssrc_ != kSendSsrc) { |
| 398 | // Not kSendSsrc, assume correct RTX packet. Extract sequence number. |
| 399 | const uint8_t* rtx_header = packet + header.headerLength; |
| 400 | sequence_number = (rtx_header[0] << 8) + rtx_header[1]; |
| 401 | } |
| 402 | |
| 403 | if (sequence_number == nacked_sequence_number_) { |
| 404 | EXPECT_EQ(retransmit_ssrc_, header.ssrc); |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 405 | send_test_complete_->Set(); |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 406 | } |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 407 | |
| 408 | return true; |
| 409 | } |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 410 | |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 411 | private: |
pbos@webrtc.org | aa693dd | 2013-09-20 11:56:26 +0000 | [diff] [blame] | 412 | internal::TransportAdapter transport_adapter_; |
| 413 | test::DirectTransport transport_; |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 414 | int send_count_; |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 415 | uint32_t retransmit_ssrc_; |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 416 | uint16_t nacked_sequence_number_; |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 417 | } observer(retransmit_ssrc); |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 418 | |
| 419 | Call::Config call_config(&observer); |
| 420 | scoped_ptr<Call> call(Call::Create(call_config)); |
| 421 | observer.SetReceiver(call->Receiver()); |
| 422 | |
| 423 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| 424 | send_config.rtp.nack.rtp_history_ms = 1000; |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 425 | if (retransmit_ssrc != kSendSsrc) |
| 426 | send_config.rtp.rtx.ssrcs.push_back(retransmit_ssrc); |
pbos@webrtc.org | af73083 | 2013-09-10 14:56:33 +0000 | [diff] [blame] | 427 | |
| 428 | RunSendTest(call.get(), send_config, &observer); |
| 429 | } |
| 430 | |
pbos@webrtc.org | f952fce | 2013-09-16 13:01:47 +0000 | [diff] [blame] | 431 | TEST_F(VideoSendStreamTest, RetransmitsNack) { |
| 432 | // Normal NACKs should use the send SSRC. |
| 433 | TestNackRetransmission(kSendSsrc); |
| 434 | } |
| 435 | |
| 436 | TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) { |
| 437 | // NACKs over RTX should use a separate SSRC. |
| 438 | TestNackRetransmission(kSendRtxSsrc); |
| 439 | } |
| 440 | |
sprang@webrtc.org | 4fe8543 | 2013-10-16 11:37:54 +0000 | [diff] [blame] | 441 | TEST_F(VideoSendStreamTest, MaxPacketSize) { |
| 442 | class PacketSizeObserver : public SendTransportObserver { |
| 443 | public: |
| 444 | PacketSizeObserver(size_t max_length) : SendTransportObserver(30 * 1000), |
| 445 | max_length_(max_length), accumulated_size_(0) {} |
| 446 | |
| 447 | virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
| 448 | RTPHeader header; |
| 449 | EXPECT_TRUE( |
| 450 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 451 | |
sprang@webrtc.org | 6133dd5 | 2013-10-16 13:29:14 +0000 | [diff] [blame] | 452 | EXPECT_LE(length, max_length_); |
sprang@webrtc.org | 4fe8543 | 2013-10-16 11:37:54 +0000 | [diff] [blame] | 453 | |
| 454 | accumulated_size_ += length; |
| 455 | |
| 456 | // Marker bit set indicates last fragment of a packet |
| 457 | if (header.markerBit) { |
| 458 | if (accumulated_size_ + length > max_length_) { |
| 459 | // The packet was fragmented, total size was larger than max size, |
| 460 | // but size of individual fragments were within size limit => pass! |
| 461 | send_test_complete_->Set(); |
| 462 | } |
| 463 | accumulated_size_ = 0; // Last fragment, reset packet size |
| 464 | } |
| 465 | |
| 466 | return true; |
| 467 | } |
| 468 | |
| 469 | private: |
| 470 | size_t max_length_; |
| 471 | size_t accumulated_size_; |
| 472 | }; |
| 473 | |
| 474 | static const uint32_t kMaxPacketSize = 128; |
| 475 | |
| 476 | PacketSizeObserver observer(kMaxPacketSize); |
| 477 | Call::Config call_config(&observer); |
| 478 | scoped_ptr<Call> call(Call::Create(call_config)); |
| 479 | |
| 480 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| 481 | send_config.rtp.max_packet_size = kMaxPacketSize; |
| 482 | |
| 483 | RunSendTest(call.get(), send_config, &observer); |
| 484 | } |
| 485 | |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 486 | // The test will go through a number of phases. |
| 487 | // 1. Start sending packets. |
| 488 | // 2. As soon as the RTP stream has been detected, signal a low REMB value to |
| 489 | // activate the auto muter. |
| 490 | // 3. Wait until |kMuteTimeFrames| have been captured without seeing any RTP |
| 491 | // packets. |
| 492 | // 4. Signal a high REMB and the wait for the RTP stream to start again. |
| 493 | // When the stream is detected again, the test ends. |
| 494 | TEST_F(VideoSendStreamTest, AutoMute) { |
| 495 | static const int kMuteTimeFrames = 60; // Mute for 2 seconds @ 30 fps. |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 496 | |
| 497 | class RembObserver : public SendTransportObserver, public I420FrameCallback { |
| 498 | public: |
| 499 | RembObserver() |
| 500 | : SendTransportObserver(30 * 1000), // Timeout after 30 seconds. |
| 501 | transport_adapter_(&transport_), |
| 502 | clock_(Clock::GetRealTimeClock()), |
| 503 | test_state_(kBeforeMute), |
| 504 | rtp_count_(0), |
| 505 | last_sequence_number_(0), |
| 506 | mute_frame_count_(0), |
henrik.lundin@webrtc.org | d4ec1f5 | 2013-10-28 10:16:14 +0000 | [diff] [blame^] | 507 | low_remb_bps_(0), |
| 508 | high_remb_bps_(0), |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 509 | crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {} |
| 510 | |
| 511 | void SetReceiver(PacketReceiver* receiver) { |
| 512 | transport_.SetReceiver(receiver); |
| 513 | } |
| 514 | |
| 515 | virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE { |
| 516 | // Receive statistics reporting having lost 0% of the packets. |
| 517 | // This is needed for the send-side bitrate controller to work properly. |
| 518 | CriticalSectionScoped lock(crit_sect_.get()); |
| 519 | SendRtcpFeedback(0); // REMB is only sent if value is > 0. |
| 520 | return true; |
| 521 | } |
| 522 | |
| 523 | virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
| 524 | CriticalSectionScoped lock(crit_sect_.get()); |
| 525 | ++rtp_count_; |
| 526 | RTPHeader header; |
| 527 | EXPECT_TRUE( |
| 528 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 529 | last_sequence_number_ = header.sequenceNumber; |
| 530 | |
| 531 | if (test_state_ == kBeforeMute) { |
| 532 | // The stream has started. Try to mute it. |
henrik.lundin@webrtc.org | d4ec1f5 | 2013-10-28 10:16:14 +0000 | [diff] [blame^] | 533 | SendRtcpFeedback(low_remb_bps_); |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 534 | test_state_ = kDuringMute; |
| 535 | } else if (test_state_ == kDuringMute) { |
| 536 | mute_frame_count_ = 0; |
| 537 | } else if (test_state_ == kWaitingForPacket) { |
| 538 | send_test_complete_->Set(); |
| 539 | } |
| 540 | |
| 541 | return true; |
| 542 | } |
| 543 | |
| 544 | // This method implements the I420FrameCallback. |
| 545 | void FrameCallback(I420VideoFrame* video_frame) OVERRIDE { |
| 546 | CriticalSectionScoped lock(crit_sect_.get()); |
| 547 | if (test_state_ == kDuringMute && ++mute_frame_count_ > kMuteTimeFrames) { |
henrik.lundin@webrtc.org | d4ec1f5 | 2013-10-28 10:16:14 +0000 | [diff] [blame^] | 548 | SendRtcpFeedback(high_remb_bps_); |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 549 | test_state_ = kWaitingForPacket; |
| 550 | } |
| 551 | } |
| 552 | |
henrik.lundin@webrtc.org | d4ec1f5 | 2013-10-28 10:16:14 +0000 | [diff] [blame^] | 553 | void set_low_remb_bps(int value) { low_remb_bps_ = value; } |
| 554 | |
| 555 | void set_high_remb_bps(int value) { high_remb_bps_ = value; } |
| 556 | |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 557 | private: |
| 558 | enum TestState { |
| 559 | kBeforeMute, |
| 560 | kDuringMute, |
| 561 | kWaitingForPacket, |
| 562 | kAfterMute |
| 563 | }; |
| 564 | |
| 565 | virtual void SendRtcpFeedback(int remb_value) { |
| 566 | FakeReceiveStatistics receive_stats( |
| 567 | kSendSsrc, last_sequence_number_, rtp_count_, 0); |
| 568 | RTCPSender rtcp_sender(0, false, clock_, &receive_stats); |
| 569 | EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_)); |
| 570 | |
| 571 | rtcp_sender.SetRTCPStatus(kRtcpNonCompound); |
| 572 | rtcp_sender.SetRemoteSSRC(kSendSsrc); |
| 573 | if (remb_value > 0) { |
| 574 | rtcp_sender.SetREMBStatus(true); |
| 575 | rtcp_sender.SetREMBData(remb_value, 0, NULL); |
| 576 | } |
| 577 | RTCPSender::FeedbackState feedback_state; |
| 578 | EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr)); |
| 579 | } |
| 580 | |
| 581 | internal::TransportAdapter transport_adapter_; |
| 582 | test::DirectTransport transport_; |
| 583 | Clock* clock_; |
| 584 | TestState test_state_; |
| 585 | int rtp_count_; |
| 586 | int last_sequence_number_; |
| 587 | int mute_frame_count_; |
henrik.lundin@webrtc.org | d4ec1f5 | 2013-10-28 10:16:14 +0000 | [diff] [blame^] | 588 | int low_remb_bps_; |
| 589 | int high_remb_bps_; |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 590 | scoped_ptr<CriticalSectionWrapper> crit_sect_; |
| 591 | } observer; |
| 592 | |
| 593 | Call::Config call_config(&observer); |
| 594 | scoped_ptr<Call> call(Call::Create(call_config)); |
| 595 | observer.SetReceiver(call->Receiver()); |
| 596 | |
| 597 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| 598 | send_config.rtp.nack.rtp_history_ms = 1000; |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 599 | send_config.pre_encode_callback = &observer; |
henrik.lundin@webrtc.org | d4ec1f5 | 2013-10-28 10:16:14 +0000 | [diff] [blame^] | 600 | send_config.auto_mute = true; |
| 601 | unsigned int min_bitrate_bps = |
| 602 | send_config.codec.simulcastStream[0].minBitrate * 1000; |
| 603 | observer.set_low_remb_bps(min_bitrate_bps - 10000); |
| 604 | unsigned int threshold_window = std::max(min_bitrate_bps / 10, 10000u); |
| 605 | ASSERT_GT(send_config.codec.simulcastStream[0].maxBitrate * 1000, |
| 606 | min_bitrate_bps + threshold_window + 5000); |
| 607 | observer.set_high_remb_bps(min_bitrate_bps + threshold_window + 5000); |
henrik.lundin@webrtc.org | ce21c82 | 2013-10-23 11:04:57 +0000 | [diff] [blame] | 608 | |
| 609 | RunSendTest(call.get(), send_config, &observer); |
| 610 | } |
| 611 | |
pbos@webrtc.org | 8a9739d | 2013-08-20 13:14:07 +0000 | [diff] [blame] | 612 | } // namespace webrtc |