pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +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 | */ |
| 10 | #include <assert.h> |
| 11 | |
| 12 | #include <algorithm> |
| 13 | #include <map> |
| 14 | #include <sstream> |
| 15 | #include <string> |
| 16 | |
| 17 | #include "testing/gtest/include/gtest/gtest.h" |
| 18 | |
| 19 | #include "webrtc/call.h" |
| 20 | #include "webrtc/frame_callback.h" |
| 21 | #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| 22 | #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
| 23 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| 24 | #include "webrtc/system_wrappers/interface/event_wrapper.h" |
| 25 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| 26 | #include "webrtc/system_wrappers/interface/sleep.h" |
| 27 | #include "webrtc/test/call_test.h" |
| 28 | #include "webrtc/test/direct_transport.h" |
| 29 | #include "webrtc/test/encoder_settings.h" |
| 30 | #include "webrtc/test/fake_audio_device.h" |
| 31 | #include "webrtc/test/fake_decoder.h" |
| 32 | #include "webrtc/test/fake_encoder.h" |
| 33 | #include "webrtc/test/frame_generator.h" |
| 34 | #include "webrtc/test/frame_generator_capturer.h" |
| 35 | #include "webrtc/test/null_transport.h" |
| 36 | #include "webrtc/test/rtp_rtcp_observer.h" |
| 37 | #include "webrtc/test/testsupport/fileutils.h" |
| 38 | #include "webrtc/test/testsupport/perf_test.h" |
| 39 | #include "webrtc/video/transport_adapter.h" |
| 40 | |
| 41 | namespace webrtc { |
| 42 | |
| 43 | static const int kRedPayloadType = 118; |
| 44 | static const int kUlpfecPayloadType = 119; |
| 45 | |
| 46 | class EndToEndTest : public test::CallTest { |
| 47 | public: |
| 48 | EndToEndTest() {} |
| 49 | |
| 50 | virtual ~EndToEndTest() { |
| 51 | EXPECT_EQ(NULL, send_stream_); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 52 | EXPECT_TRUE(receive_streams_.empty()); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 53 | } |
| 54 | |
| 55 | protected: |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 56 | void DecodesRetransmittedFrame(bool retransmit_over_rtx); |
| 57 | void ReceivesPliAndRecovers(int rtp_history_ms); |
| 58 | void RespectsRtcpMode(newapi::RtcpMode rtcp_mode); |
| 59 | void TestXrReceiverReferenceTimeReport(bool enable_rrtr); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 60 | void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 61 | void TestRtpStatePreservation(bool use_rtx); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 62 | }; |
| 63 | |
| 64 | TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) { |
| 65 | test::NullTransport transport; |
| 66 | CreateCalls(Call::Config(&transport), Call::Config(&transport)); |
| 67 | |
| 68 | CreateSendConfig(1); |
| 69 | CreateMatchingReceiveConfigs(); |
| 70 | |
| 71 | CreateStreams(); |
| 72 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 73 | receive_streams_[0]->Start(); |
| 74 | receive_streams_[0]->Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 75 | |
| 76 | DestroyStreams(); |
| 77 | } |
| 78 | |
| 79 | TEST_F(EndToEndTest, ReceiverCanBeStoppedTwice) { |
| 80 | test::NullTransport transport; |
| 81 | CreateCalls(Call::Config(&transport), Call::Config(&transport)); |
| 82 | |
| 83 | CreateSendConfig(1); |
| 84 | CreateMatchingReceiveConfigs(); |
| 85 | |
| 86 | CreateStreams(); |
| 87 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 88 | receive_streams_[0]->Stop(); |
| 89 | receive_streams_[0]->Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 90 | |
| 91 | DestroyStreams(); |
| 92 | } |
| 93 | |
| 94 | TEST_F(EndToEndTest, RendersSingleDelayedFrame) { |
| 95 | static const int kWidth = 320; |
| 96 | static const int kHeight = 240; |
| 97 | // This constant is chosen to be higher than the timeout in the video_render |
| 98 | // module. This makes sure that frames aren't dropped if there are no other |
| 99 | // frames in the queue. |
| 100 | static const int kDelayRenderCallbackMs = 1000; |
| 101 | |
| 102 | class Renderer : public VideoRenderer { |
| 103 | public: |
| 104 | Renderer() : event_(EventWrapper::Create()) {} |
| 105 | |
| 106 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 107 | int /*time_to_render_ms*/) OVERRIDE { |
| 108 | event_->Set(); |
| 109 | } |
| 110 | |
| 111 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 112 | |
| 113 | scoped_ptr<EventWrapper> event_; |
| 114 | } renderer; |
| 115 | |
| 116 | class TestFrameCallback : public I420FrameCallback { |
| 117 | public: |
| 118 | TestFrameCallback() : event_(EventWrapper::Create()) {} |
| 119 | |
| 120 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 121 | |
| 122 | private: |
| 123 | virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE { |
| 124 | SleepMs(kDelayRenderCallbackMs); |
| 125 | event_->Set(); |
| 126 | } |
| 127 | |
| 128 | scoped_ptr<EventWrapper> event_; |
| 129 | }; |
| 130 | |
| 131 | test::DirectTransport sender_transport, receiver_transport; |
| 132 | |
| 133 | CreateCalls(Call::Config(&sender_transport), |
| 134 | Call::Config(&receiver_transport)); |
| 135 | |
| 136 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 137 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 138 | |
| 139 | CreateSendConfig(1); |
| 140 | CreateMatchingReceiveConfigs(); |
| 141 | |
| 142 | TestFrameCallback pre_render_callback; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 143 | receive_configs_[0].pre_render_callback = &pre_render_callback; |
| 144 | receive_configs_[0].renderer = &renderer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 145 | |
| 146 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 147 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 148 | |
| 149 | // Create frames that are smaller than the send width/height, this is done to |
| 150 | // check that the callbacks are done after processing video. |
| 151 | scoped_ptr<test::FrameGenerator> frame_generator( |
| 152 | test::FrameGenerator::Create(kWidth, kHeight)); |
| 153 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 154 | EXPECT_EQ(kEventSignaled, pre_render_callback.Wait()) |
| 155 | << "Timed out while waiting for pre-render callback."; |
| 156 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 157 | << "Timed out while waiting for the frame to render."; |
| 158 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 159 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 160 | |
| 161 | sender_transport.StopSending(); |
| 162 | receiver_transport.StopSending(); |
| 163 | |
| 164 | DestroyStreams(); |
| 165 | } |
| 166 | |
| 167 | TEST_F(EndToEndTest, TransmitsFirstFrame) { |
| 168 | class Renderer : public VideoRenderer { |
| 169 | public: |
| 170 | Renderer() : event_(EventWrapper::Create()) {} |
| 171 | |
| 172 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 173 | int /*time_to_render_ms*/) OVERRIDE { |
| 174 | event_->Set(); |
| 175 | } |
| 176 | |
| 177 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 178 | |
| 179 | scoped_ptr<EventWrapper> event_; |
| 180 | } renderer; |
| 181 | |
| 182 | test::DirectTransport sender_transport, receiver_transport; |
| 183 | |
| 184 | CreateCalls(Call::Config(&sender_transport), |
| 185 | Call::Config(&receiver_transport)); |
| 186 | |
| 187 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 188 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 189 | |
| 190 | CreateSendConfig(1); |
| 191 | CreateMatchingReceiveConfigs(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 192 | receive_configs_[0].renderer = &renderer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 193 | |
| 194 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 195 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 196 | |
| 197 | scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create( |
| 198 | video_streams_[0].width, video_streams_[0].height)); |
| 199 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 200 | |
| 201 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 202 | << "Timed out while waiting for the frame to render."; |
| 203 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 204 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 205 | |
| 206 | sender_transport.StopSending(); |
| 207 | receiver_transport.StopSending(); |
| 208 | |
| 209 | DestroyStreams(); |
| 210 | } |
| 211 | |
| 212 | TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { |
| 213 | class SyncRtcpObserver : public test::EndToEndTest { |
| 214 | public: |
| 215 | SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
| 216 | |
| 217 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 218 | size_t length) OVERRIDE { |
| 219 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 220 | EXPECT_TRUE(parser.IsValid()); |
| 221 | uint32_t ssrc = 0; |
| 222 | ssrc |= static_cast<uint32_t>(packet[4]) << 24; |
| 223 | ssrc |= static_cast<uint32_t>(packet[5]) << 16; |
| 224 | ssrc |= static_cast<uint32_t>(packet[6]) << 8; |
| 225 | ssrc |= static_cast<uint32_t>(packet[7]) << 0; |
| 226 | EXPECT_EQ(kReceiverLocalSsrc, ssrc); |
| 227 | observation_complete_->Set(); |
| 228 | |
| 229 | return SEND_PACKET; |
| 230 | } |
| 231 | |
| 232 | virtual void PerformTest() OVERRIDE { |
| 233 | EXPECT_EQ(kEventSignaled, Wait()) |
| 234 | << "Timed out while waiting for a receiver RTCP packet to be sent."; |
| 235 | } |
| 236 | } test; |
| 237 | |
| 238 | RunBaseTest(&test); |
| 239 | } |
| 240 | |
| 241 | TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) { |
| 242 | static const int kNumberOfNacksToObserve = 2; |
| 243 | static const int kLossBurstSize = 2; |
| 244 | static const int kPacketsBetweenLossBursts = 9; |
| 245 | class NackObserver : public test::EndToEndTest { |
| 246 | public: |
| 247 | NackObserver() |
| 248 | : EndToEndTest(kLongTimeoutMs), |
| 249 | rtp_parser_(RtpHeaderParser::Create()), |
| 250 | sent_rtp_packets_(0), |
| 251 | packets_left_to_drop_(0), |
| 252 | nacks_left_(kNumberOfNacksToObserve) {} |
| 253 | |
| 254 | private: |
| 255 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 256 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 257 | EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 258 | |
| 259 | // Never drop retransmitted packets. |
| 260 | if (dropped_packets_.find(header.sequenceNumber) != |
| 261 | dropped_packets_.end()) { |
| 262 | retransmitted_packets_.insert(header.sequenceNumber); |
| 263 | if (nacks_left_ == 0 && |
| 264 | retransmitted_packets_.size() == dropped_packets_.size()) { |
| 265 | observation_complete_->Set(); |
| 266 | } |
| 267 | return SEND_PACKET; |
| 268 | } |
| 269 | |
| 270 | ++sent_rtp_packets_; |
| 271 | |
| 272 | // Enough NACKs received, stop dropping packets. |
| 273 | if (nacks_left_ == 0) |
| 274 | return SEND_PACKET; |
| 275 | |
| 276 | // Check if it's time for a new loss burst. |
| 277 | if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0) |
| 278 | packets_left_to_drop_ = kLossBurstSize; |
| 279 | |
| 280 | if (packets_left_to_drop_ > 0) { |
| 281 | --packets_left_to_drop_; |
| 282 | dropped_packets_.insert(header.sequenceNumber); |
| 283 | return DROP_PACKET; |
| 284 | } |
| 285 | |
| 286 | return SEND_PACKET; |
| 287 | } |
| 288 | |
| 289 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 290 | size_t length) OVERRIDE { |
| 291 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 292 | EXPECT_TRUE(parser.IsValid()); |
| 293 | |
| 294 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 295 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 296 | if (packet_type == RTCPUtility::kRtcpRtpfbNackCode) { |
| 297 | --nacks_left_; |
| 298 | break; |
| 299 | } |
| 300 | packet_type = parser.Iterate(); |
| 301 | } |
| 302 | return SEND_PACKET; |
| 303 | } |
| 304 | |
| 305 | virtual void ModifyConfigs( |
| 306 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 307 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 308 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 309 | send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 310 | (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 311 | } |
| 312 | |
| 313 | virtual void PerformTest() OVERRIDE { |
| 314 | EXPECT_EQ(kEventSignaled, Wait()) |
| 315 | << "Timed out waiting for packets to be NACKed, retransmitted and " |
| 316 | "rendered."; |
| 317 | } |
| 318 | |
| 319 | scoped_ptr<RtpHeaderParser> rtp_parser_; |
| 320 | std::set<uint16_t> dropped_packets_; |
| 321 | std::set<uint16_t> retransmitted_packets_; |
| 322 | uint64_t sent_rtp_packets_; |
| 323 | int packets_left_to_drop_; |
| 324 | int nacks_left_; |
| 325 | } test; |
| 326 | |
| 327 | RunBaseTest(&test); |
| 328 | } |
| 329 | |
| 330 | // TODO(pbos): Flaky, webrtc:3269 |
| 331 | TEST_F(EndToEndTest, DISABLED_CanReceiveFec) { |
| 332 | class FecRenderObserver : public test::EndToEndTest, public VideoRenderer { |
| 333 | public: |
| 334 | FecRenderObserver() |
| 335 | : EndToEndTest(kDefaultTimeoutMs), |
| 336 | state_(kFirstPacket), |
| 337 | protected_sequence_number_(0), |
| 338 | protected_frame_timestamp_(0) {} |
| 339 | |
| 340 | private: |
| 341 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE |
| 342 | EXCLUSIVE_LOCKS_REQUIRED(crit_) { |
| 343 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 344 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 345 | |
| 346 | EXPECT_EQ(kRedPayloadType, header.payloadType); |
| 347 | int encapsulated_payload_type = |
| 348 | static_cast<int>(packet[header.headerLength]); |
| 349 | if (encapsulated_payload_type != kFakeSendPayloadType) |
| 350 | EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type); |
| 351 | |
| 352 | switch (state_) { |
| 353 | case kFirstPacket: |
| 354 | state_ = kDropEveryOtherPacketUntilFec; |
| 355 | break; |
| 356 | case kDropEveryOtherPacketUntilFec: |
| 357 | if (encapsulated_payload_type == kUlpfecPayloadType) { |
| 358 | state_ = kDropNextMediaPacket; |
| 359 | return SEND_PACKET; |
| 360 | } |
| 361 | if (header.sequenceNumber % 2 == 0) |
| 362 | return DROP_PACKET; |
| 363 | break; |
| 364 | case kDropNextMediaPacket: |
| 365 | if (encapsulated_payload_type == kFakeSendPayloadType) { |
| 366 | protected_sequence_number_ = header.sequenceNumber; |
| 367 | protected_frame_timestamp_ = header.timestamp; |
| 368 | state_ = kProtectedPacketDropped; |
| 369 | return DROP_PACKET; |
| 370 | } |
| 371 | break; |
| 372 | case kProtectedPacketDropped: |
| 373 | EXPECT_NE(header.sequenceNumber, protected_sequence_number_) |
| 374 | << "Protected packet retransmitted. Should not happen with FEC."; |
| 375 | break; |
| 376 | } |
| 377 | |
| 378 | return SEND_PACKET; |
| 379 | } |
| 380 | |
| 381 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 382 | int time_to_render_ms) OVERRIDE { |
| 383 | CriticalSectionScoped lock(crit_.get()); |
| 384 | // Rendering frame with timestamp associated with dropped packet -> FEC |
| 385 | // protection worked. |
| 386 | if (state_ == kProtectedPacketDropped && |
| 387 | video_frame.timestamp() == protected_frame_timestamp_) { |
| 388 | observation_complete_->Set(); |
| 389 | } |
| 390 | } |
| 391 | |
| 392 | enum { |
| 393 | kFirstPacket, |
| 394 | kDropEveryOtherPacketUntilFec, |
| 395 | kDropNextMediaPacket, |
| 396 | kProtectedPacketDropped, |
| 397 | } state_; |
| 398 | |
| 399 | virtual void ModifyConfigs( |
| 400 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 401 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 402 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 403 | // TODO(pbos): Run this test with combined NACK/FEC enabled as well. |
| 404 | // int rtp_history_ms = 1000; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 405 | // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 406 | // send_config->rtp.nack.rtp_history_ms = rtp_history_ms; |
| 407 | send_config->rtp.fec.red_payload_type = kRedPayloadType; |
| 408 | send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| 409 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 410 | (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType; |
| 411 | (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| 412 | (*receive_configs)[0].renderer = this; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 413 | } |
| 414 | |
| 415 | virtual void PerformTest() OVERRIDE { |
| 416 | EXPECT_EQ(kEventSignaled, Wait()) |
| 417 | << "Timed out while waiting for retransmitted NACKed frames to be " |
| 418 | "rendered again."; |
| 419 | } |
| 420 | |
| 421 | uint32_t protected_sequence_number_ GUARDED_BY(crit_); |
| 422 | uint32_t protected_frame_timestamp_ GUARDED_BY(crit_); |
| 423 | } test; |
| 424 | |
| 425 | RunBaseTest(&test); |
| 426 | } |
| 427 | |
| 428 | // This test drops second RTP packet with a marker bit set, makes sure it's |
| 429 | // retransmitted and renders. Retransmission SSRCs are also checked. |
| 430 | void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) { |
| 431 | static const int kDroppedFrameNumber = 2; |
| 432 | class RetransmissionObserver : public test::EndToEndTest, |
| 433 | public I420FrameCallback { |
| 434 | public: |
| 435 | explicit RetransmissionObserver(bool expect_rtx) |
| 436 | : EndToEndTest(kDefaultTimeoutMs), |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 437 | retransmission_ssrc_(expect_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]), |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 438 | retransmission_payload_type_(expect_rtx ? kSendRtxPayloadType |
| 439 | : kFakeSendPayloadType), |
| 440 | marker_bits_observed_(0), |
| 441 | retransmitted_timestamp_(0), |
| 442 | frame_retransmitted_(false) {} |
| 443 | |
| 444 | private: |
| 445 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 446 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 447 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 448 | |
| 449 | if (header.timestamp == retransmitted_timestamp_) { |
| 450 | EXPECT_EQ(retransmission_ssrc_, header.ssrc); |
| 451 | EXPECT_EQ(retransmission_payload_type_, header.payloadType); |
| 452 | frame_retransmitted_ = true; |
| 453 | return SEND_PACKET; |
| 454 | } |
| 455 | |
| 456 | EXPECT_EQ(kSendSsrcs[0], header.ssrc); |
| 457 | EXPECT_EQ(kFakeSendPayloadType, header.payloadType); |
| 458 | |
| 459 | // Found the second frame's final packet, drop this and expect a |
| 460 | // retransmission. |
| 461 | if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) { |
| 462 | retransmitted_timestamp_ = header.timestamp; |
| 463 | return DROP_PACKET; |
| 464 | } |
| 465 | |
| 466 | return SEND_PACKET; |
| 467 | } |
| 468 | |
| 469 | virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE { |
| 470 | CriticalSectionScoped lock(crit_.get()); |
| 471 | if (frame->timestamp() == retransmitted_timestamp_) { |
| 472 | EXPECT_TRUE(frame_retransmitted_); |
| 473 | observation_complete_->Set(); |
| 474 | } |
| 475 | } |
| 476 | |
| 477 | virtual void ModifyConfigs( |
| 478 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 479 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 480 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 481 | send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 482 | (*receive_configs)[0].pre_render_callback = this; |
| 483 | (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 484 | if (retransmission_ssrc_ == kSendRtxSsrcs[0]) { |
| 485 | send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 486 | send_config->rtp.rtx.payload_type = kSendRtxPayloadType; |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 487 | (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc = |
| 488 | kSendRtxSsrcs[0]; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 489 | (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].payload_type = |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 490 | kSendRtxPayloadType; |
| 491 | } |
| 492 | } |
| 493 | |
| 494 | virtual void PerformTest() OVERRIDE { |
| 495 | EXPECT_EQ(kEventSignaled, Wait()) |
| 496 | << "Timed out while waiting for retransmission to render."; |
| 497 | } |
| 498 | |
| 499 | const uint32_t retransmission_ssrc_; |
| 500 | const int retransmission_payload_type_; |
| 501 | int marker_bits_observed_; |
| 502 | uint32_t retransmitted_timestamp_; |
| 503 | bool frame_retransmitted_; |
| 504 | } test(retransmit_over_rtx); |
| 505 | |
| 506 | RunBaseTest(&test); |
| 507 | } |
| 508 | |
| 509 | TEST_F(EndToEndTest, DecodesRetransmittedFrame) { |
| 510 | DecodesRetransmittedFrame(false); |
| 511 | } |
| 512 | |
| 513 | TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) { |
| 514 | DecodesRetransmittedFrame(true); |
| 515 | } |
| 516 | |
| 517 | TEST_F(EndToEndTest, UsesFrameCallbacks) { |
| 518 | static const int kWidth = 320; |
| 519 | static const int kHeight = 240; |
| 520 | |
| 521 | class Renderer : public VideoRenderer { |
| 522 | public: |
| 523 | Renderer() : event_(EventWrapper::Create()) {} |
| 524 | |
| 525 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 526 | int /*time_to_render_ms*/) OVERRIDE { |
| 527 | EXPECT_EQ(0, *video_frame.buffer(kYPlane)) |
| 528 | << "Rendered frame should have zero luma which is applied by the " |
| 529 | "pre-render callback."; |
| 530 | event_->Set(); |
| 531 | } |
| 532 | |
| 533 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 534 | scoped_ptr<EventWrapper> event_; |
| 535 | } renderer; |
| 536 | |
| 537 | class TestFrameCallback : public I420FrameCallback { |
| 538 | public: |
| 539 | TestFrameCallback(int expected_luma_byte, int next_luma_byte) |
| 540 | : event_(EventWrapper::Create()), |
| 541 | expected_luma_byte_(expected_luma_byte), |
| 542 | next_luma_byte_(next_luma_byte) {} |
| 543 | |
| 544 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 545 | |
| 546 | private: |
| 547 | virtual void FrameCallback(I420VideoFrame* frame) { |
| 548 | EXPECT_EQ(kWidth, frame->width()) |
| 549 | << "Width not as expected, callback done before resize?"; |
| 550 | EXPECT_EQ(kHeight, frame->height()) |
| 551 | << "Height not as expected, callback done before resize?"; |
| 552 | |
| 553 | // Previous luma specified, observed luma should be fairly close. |
| 554 | if (expected_luma_byte_ != -1) { |
| 555 | EXPECT_NEAR(expected_luma_byte_, *frame->buffer(kYPlane), 10); |
| 556 | } |
| 557 | |
| 558 | memset(frame->buffer(kYPlane), |
| 559 | next_luma_byte_, |
| 560 | frame->allocated_size(kYPlane)); |
| 561 | |
| 562 | event_->Set(); |
| 563 | } |
| 564 | |
| 565 | scoped_ptr<EventWrapper> event_; |
| 566 | int expected_luma_byte_; |
| 567 | int next_luma_byte_; |
| 568 | }; |
| 569 | |
| 570 | TestFrameCallback pre_encode_callback(-1, 255); // Changes luma to 255. |
| 571 | TestFrameCallback pre_render_callback(255, 0); // Changes luma from 255 to 0. |
| 572 | |
| 573 | test::DirectTransport sender_transport, receiver_transport; |
| 574 | |
| 575 | CreateCalls(Call::Config(&sender_transport), |
| 576 | Call::Config(&receiver_transport)); |
| 577 | |
| 578 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 579 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 580 | |
| 581 | CreateSendConfig(1); |
| 582 | scoped_ptr<VP8Encoder> encoder(VP8Encoder::Create()); |
| 583 | send_config_.encoder_settings.encoder = encoder.get(); |
| 584 | send_config_.encoder_settings.payload_name = "VP8"; |
| 585 | ASSERT_EQ(1u, video_streams_.size()) << "Test setup error."; |
| 586 | video_streams_[0].width = kWidth; |
| 587 | video_streams_[0].height = kHeight; |
| 588 | send_config_.pre_encode_callback = &pre_encode_callback; |
| 589 | |
| 590 | CreateMatchingReceiveConfigs(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 591 | receive_configs_[0].pre_render_callback = &pre_render_callback; |
| 592 | receive_configs_[0].renderer = &renderer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 593 | |
| 594 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 595 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 596 | |
| 597 | // Create frames that are smaller than the send width/height, this is done to |
| 598 | // check that the callbacks are done after processing video. |
| 599 | scoped_ptr<test::FrameGenerator> frame_generator( |
| 600 | test::FrameGenerator::Create(kWidth / 2, kHeight / 2)); |
| 601 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 602 | |
| 603 | EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait()) |
| 604 | << "Timed out while waiting for pre-encode callback."; |
| 605 | EXPECT_EQ(kEventSignaled, pre_render_callback.Wait()) |
| 606 | << "Timed out while waiting for pre-render callback."; |
| 607 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 608 | << "Timed out while waiting for the frame to render."; |
| 609 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 610 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 611 | |
| 612 | sender_transport.StopSending(); |
| 613 | receiver_transport.StopSending(); |
| 614 | |
| 615 | DestroyStreams(); |
| 616 | } |
| 617 | |
| 618 | void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) { |
| 619 | static const int kPacketsToDrop = 1; |
| 620 | |
| 621 | class PliObserver : public test::EndToEndTest, public VideoRenderer { |
| 622 | public: |
| 623 | explicit PliObserver(int rtp_history_ms) |
| 624 | : EndToEndTest(kLongTimeoutMs), |
| 625 | rtp_history_ms_(rtp_history_ms), |
| 626 | nack_enabled_(rtp_history_ms > 0), |
| 627 | highest_dropped_timestamp_(0), |
| 628 | frames_to_drop_(0), |
| 629 | received_pli_(false) {} |
| 630 | |
| 631 | private: |
| 632 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 633 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 634 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 635 | |
| 636 | // Drop all retransmitted packets to force a PLI. |
| 637 | if (header.timestamp <= highest_dropped_timestamp_) |
| 638 | return DROP_PACKET; |
| 639 | |
| 640 | if (frames_to_drop_ > 0) { |
| 641 | highest_dropped_timestamp_ = header.timestamp; |
| 642 | --frames_to_drop_; |
| 643 | return DROP_PACKET; |
| 644 | } |
| 645 | |
| 646 | return SEND_PACKET; |
| 647 | } |
| 648 | |
| 649 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 650 | size_t length) OVERRIDE { |
| 651 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 652 | EXPECT_TRUE(parser.IsValid()); |
| 653 | |
| 654 | for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 655 | packet_type != RTCPUtility::kRtcpNotValidCode; |
| 656 | packet_type = parser.Iterate()) { |
| 657 | if (!nack_enabled_) |
| 658 | EXPECT_NE(packet_type, RTCPUtility::kRtcpRtpfbNackCode); |
| 659 | |
| 660 | if (packet_type == RTCPUtility::kRtcpPsfbPliCode) { |
| 661 | received_pli_ = true; |
| 662 | break; |
| 663 | } |
| 664 | } |
| 665 | return SEND_PACKET; |
| 666 | } |
| 667 | |
| 668 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 669 | int time_to_render_ms) OVERRIDE { |
| 670 | CriticalSectionScoped lock(crit_.get()); |
| 671 | if (received_pli_ && |
| 672 | video_frame.timestamp() > highest_dropped_timestamp_) { |
| 673 | observation_complete_->Set(); |
| 674 | } |
| 675 | if (!received_pli_) |
| 676 | frames_to_drop_ = kPacketsToDrop; |
| 677 | } |
| 678 | |
| 679 | virtual void ModifyConfigs( |
| 680 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 681 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 682 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 683 | send_config->rtp.nack.rtp_history_ms = rtp_history_ms_; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 684 | (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_; |
| 685 | (*receive_configs)[0].renderer = this; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 686 | } |
| 687 | |
| 688 | virtual void PerformTest() OVERRIDE { |
| 689 | EXPECT_EQ(kEventSignaled, Wait()) << "Timed out waiting for PLI to be " |
| 690 | "received and a frame to be " |
| 691 | "rendered afterwards."; |
| 692 | } |
| 693 | |
| 694 | int rtp_history_ms_; |
| 695 | bool nack_enabled_; |
| 696 | uint32_t highest_dropped_timestamp_; |
| 697 | int frames_to_drop_; |
| 698 | bool received_pli_; |
| 699 | } test(rtp_history_ms); |
| 700 | |
| 701 | RunBaseTest(&test); |
| 702 | } |
| 703 | |
| 704 | TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) { |
| 705 | ReceivesPliAndRecovers(1000); |
| 706 | } |
| 707 | |
| 708 | // TODO(pbos): Enable this when 2250 is resolved. |
| 709 | TEST_F(EndToEndTest, DISABLED_ReceivesPliAndRecoversWithoutNack) { |
| 710 | ReceivesPliAndRecovers(0); |
| 711 | } |
| 712 | |
| 713 | TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) { |
| 714 | class PacketInputObserver : public PacketReceiver { |
| 715 | public: |
| 716 | explicit PacketInputObserver(PacketReceiver* receiver) |
| 717 | : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {} |
| 718 | |
| 719 | EventTypeWrapper Wait() { |
| 720 | return delivered_packet_->Wait(kDefaultTimeoutMs); |
| 721 | } |
| 722 | |
| 723 | private: |
| 724 | virtual DeliveryStatus DeliverPacket(const uint8_t* packet, |
| 725 | size_t length) OVERRIDE { |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 726 | if (RtpHeaderParser::IsRtcp(packet, length)) { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 727 | return receiver_->DeliverPacket(packet, length); |
| 728 | } else { |
| 729 | DeliveryStatus delivery_status = |
| 730 | receiver_->DeliverPacket(packet, length); |
| 731 | EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status); |
| 732 | delivered_packet_->Set(); |
| 733 | return delivery_status; |
| 734 | } |
| 735 | } |
| 736 | |
| 737 | PacketReceiver* receiver_; |
| 738 | scoped_ptr<EventWrapper> delivered_packet_; |
| 739 | }; |
| 740 | |
| 741 | test::DirectTransport send_transport, receive_transport; |
| 742 | |
| 743 | CreateCalls(Call::Config(&send_transport), Call::Config(&receive_transport)); |
| 744 | PacketInputObserver input_observer(receiver_call_->Receiver()); |
| 745 | |
| 746 | send_transport.SetReceiver(&input_observer); |
| 747 | receive_transport.SetReceiver(sender_call_->Receiver()); |
| 748 | |
| 749 | CreateSendConfig(1); |
| 750 | CreateMatchingReceiveConfigs(); |
| 751 | |
| 752 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 753 | CreateFrameGeneratorCapturer(); |
| 754 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 755 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 756 | receiver_call_->DestroyVideoReceiveStream(receive_streams_[0]); |
| 757 | receive_streams_.clear(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 758 | |
| 759 | // Wait() waits for a received packet. |
| 760 | EXPECT_EQ(kEventSignaled, input_observer.Wait()); |
| 761 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 762 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 763 | |
| 764 | DestroyStreams(); |
| 765 | |
| 766 | send_transport.StopSending(); |
| 767 | receive_transport.StopSending(); |
| 768 | } |
| 769 | |
| 770 | void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) { |
| 771 | static const int kNumCompoundRtcpPacketsToObserve = 10; |
| 772 | class RtcpModeObserver : public test::EndToEndTest { |
| 773 | public: |
| 774 | explicit RtcpModeObserver(newapi::RtcpMode rtcp_mode) |
| 775 | : EndToEndTest(kDefaultTimeoutMs), |
| 776 | rtcp_mode_(rtcp_mode), |
| 777 | sent_rtp_(0), |
| 778 | sent_rtcp_(0) {} |
| 779 | |
| 780 | private: |
| 781 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 782 | if (++sent_rtp_ % 3 == 0) |
| 783 | return DROP_PACKET; |
| 784 | |
| 785 | return SEND_PACKET; |
| 786 | } |
| 787 | |
| 788 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 789 | size_t length) OVERRIDE { |
| 790 | ++sent_rtcp_; |
| 791 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 792 | EXPECT_TRUE(parser.IsValid()); |
| 793 | |
| 794 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 795 | bool has_report_block = false; |
| 796 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 797 | EXPECT_NE(RTCPUtility::kRtcpSrCode, packet_type); |
| 798 | if (packet_type == RTCPUtility::kRtcpRrCode) { |
| 799 | has_report_block = true; |
| 800 | break; |
| 801 | } |
| 802 | packet_type = parser.Iterate(); |
| 803 | } |
| 804 | |
| 805 | switch (rtcp_mode_) { |
| 806 | case newapi::kRtcpCompound: |
| 807 | if (!has_report_block) { |
| 808 | ADD_FAILURE() << "Received RTCP packet without receiver report for " |
| 809 | "kRtcpCompound."; |
| 810 | observation_complete_->Set(); |
| 811 | } |
| 812 | |
| 813 | if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve) |
| 814 | observation_complete_->Set(); |
| 815 | |
| 816 | break; |
| 817 | case newapi::kRtcpReducedSize: |
| 818 | if (!has_report_block) |
| 819 | observation_complete_->Set(); |
| 820 | break; |
| 821 | } |
| 822 | |
| 823 | return SEND_PACKET; |
| 824 | } |
| 825 | |
| 826 | virtual void ModifyConfigs( |
| 827 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 828 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 829 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 830 | send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 831 | (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| 832 | (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 833 | } |
| 834 | |
| 835 | virtual void PerformTest() OVERRIDE { |
| 836 | EXPECT_EQ(kEventSignaled, Wait()) |
| 837 | << (rtcp_mode_ == newapi::kRtcpCompound |
| 838 | ? "Timed out before observing enough compound packets." |
| 839 | : "Timed out before receiving a non-compound RTCP packet."); |
| 840 | } |
| 841 | |
| 842 | newapi::RtcpMode rtcp_mode_; |
| 843 | int sent_rtp_; |
| 844 | int sent_rtcp_; |
| 845 | } test(rtcp_mode); |
| 846 | |
| 847 | RunBaseTest(&test); |
| 848 | } |
| 849 | |
| 850 | TEST_F(EndToEndTest, UsesRtcpCompoundMode) { |
| 851 | RespectsRtcpMode(newapi::kRtcpCompound); |
| 852 | } |
| 853 | |
| 854 | TEST_F(EndToEndTest, UsesRtcpReducedSizeMode) { |
| 855 | RespectsRtcpMode(newapi::kRtcpReducedSize); |
| 856 | } |
| 857 | |
| 858 | // Test sets up a Call multiple senders with different resolutions and SSRCs. |
| 859 | // Another is set up to receive all three of these with different renderers. |
| 860 | // Each renderer verifies that it receives the expected resolution, and as soon |
| 861 | // as every renderer has received a frame, the test finishes. |
| 862 | TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) { |
| 863 | static const size_t kNumStreams = 3; |
| 864 | |
| 865 | class VideoOutputObserver : public VideoRenderer { |
| 866 | public: |
| 867 | VideoOutputObserver(test::FrameGeneratorCapturer** capturer, |
| 868 | int width, |
| 869 | int height) |
| 870 | : capturer_(capturer), |
| 871 | width_(width), |
| 872 | height_(height), |
| 873 | done_(EventWrapper::Create()) {} |
| 874 | |
| 875 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 876 | int time_to_render_ms) OVERRIDE { |
| 877 | EXPECT_EQ(width_, video_frame.width()); |
| 878 | EXPECT_EQ(height_, video_frame.height()); |
| 879 | (*capturer_)->Stop(); |
| 880 | done_->Set(); |
| 881 | } |
| 882 | |
| 883 | EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); } |
| 884 | |
| 885 | private: |
| 886 | test::FrameGeneratorCapturer** capturer_; |
| 887 | int width_; |
| 888 | int height_; |
| 889 | scoped_ptr<EventWrapper> done_; |
| 890 | }; |
| 891 | |
| 892 | struct { |
| 893 | uint32_t ssrc; |
| 894 | int width; |
| 895 | int height; |
| 896 | } codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}}; |
| 897 | |
| 898 | test::DirectTransport sender_transport, receiver_transport; |
| 899 | scoped_ptr<Call> sender_call(Call::Create(Call::Config(&sender_transport))); |
| 900 | scoped_ptr<Call> receiver_call( |
| 901 | Call::Create(Call::Config(&receiver_transport))); |
| 902 | sender_transport.SetReceiver(receiver_call->Receiver()); |
| 903 | receiver_transport.SetReceiver(sender_call->Receiver()); |
| 904 | |
| 905 | VideoSendStream* send_streams[kNumStreams]; |
| 906 | VideoReceiveStream* receive_streams[kNumStreams]; |
| 907 | |
| 908 | VideoOutputObserver* observers[kNumStreams]; |
| 909 | test::FrameGeneratorCapturer* frame_generators[kNumStreams]; |
| 910 | |
| 911 | scoped_ptr<VP8Encoder> encoders[kNumStreams]; |
| 912 | for (size_t i = 0; i < kNumStreams; ++i) |
| 913 | encoders[i].reset(VP8Encoder::Create()); |
| 914 | |
| 915 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 916 | uint32_t ssrc = codec_settings[i].ssrc; |
| 917 | int width = codec_settings[i].width; |
| 918 | int height = codec_settings[i].height; |
| 919 | observers[i] = new VideoOutputObserver(&frame_generators[i], width, height); |
| 920 | |
pbos@webrtc.org | bd249bc | 2014-07-07 04:45:15 +0000 | [diff] [blame] | 921 | VideoSendStream::Config send_config; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 922 | send_config.rtp.ssrcs.push_back(ssrc); |
| 923 | send_config.encoder_settings.encoder = encoders[i].get(); |
| 924 | send_config.encoder_settings.payload_name = "VP8"; |
| 925 | send_config.encoder_settings.payload_type = 124; |
| 926 | std::vector<VideoStream> video_streams = test::CreateVideoStreams(1); |
| 927 | VideoStream* stream = &video_streams[0]; |
| 928 | stream->width = width; |
| 929 | stream->height = height; |
| 930 | stream->max_framerate = 5; |
| 931 | stream->min_bitrate_bps = stream->target_bitrate_bps = |
| 932 | stream->max_bitrate_bps = 100000; |
| 933 | send_streams[i] = |
| 934 | sender_call->CreateVideoSendStream(send_config, video_streams, NULL); |
| 935 | send_streams[i]->Start(); |
| 936 | |
pbos@webrtc.org | bd249bc | 2014-07-07 04:45:15 +0000 | [diff] [blame] | 937 | VideoReceiveStream::Config receive_config; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 938 | receive_config.renderer = observers[i]; |
| 939 | receive_config.rtp.remote_ssrc = ssrc; |
| 940 | receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| 941 | VideoCodec codec = |
| 942 | test::CreateDecoderVideoCodec(send_config.encoder_settings); |
| 943 | receive_config.codecs.push_back(codec); |
| 944 | receive_streams[i] = |
| 945 | receiver_call->CreateVideoReceiveStream(receive_config); |
| 946 | receive_streams[i]->Start(); |
| 947 | |
| 948 | frame_generators[i] = test::FrameGeneratorCapturer::Create( |
| 949 | send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock()); |
| 950 | frame_generators[i]->Start(); |
| 951 | } |
| 952 | |
| 953 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 954 | EXPECT_EQ(kEventSignaled, observers[i]->Wait()) |
| 955 | << "Timed out while waiting for observer " << i << " to render."; |
| 956 | } |
| 957 | |
| 958 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 959 | frame_generators[i]->Stop(); |
| 960 | sender_call->DestroyVideoSendStream(send_streams[i]); |
| 961 | receiver_call->DestroyVideoReceiveStream(receive_streams[i]); |
| 962 | delete frame_generators[i]; |
| 963 | delete observers[i]; |
| 964 | } |
| 965 | |
| 966 | sender_transport.StopSending(); |
| 967 | receiver_transport.StopSending(); |
| 968 | } |
| 969 | |
| 970 | TEST_F(EndToEndTest, ObserversEncodedFrames) { |
| 971 | class EncodedFrameTestObserver : public EncodedFrameObserver { |
| 972 | public: |
| 973 | EncodedFrameTestObserver() |
| 974 | : length_(0), |
| 975 | frame_type_(kFrameEmpty), |
| 976 | called_(EventWrapper::Create()) {} |
| 977 | virtual ~EncodedFrameTestObserver() {} |
| 978 | |
| 979 | virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { |
| 980 | frame_type_ = encoded_frame.frame_type_; |
| 981 | length_ = encoded_frame.length_; |
| 982 | buffer_.reset(new uint8_t[length_]); |
| 983 | memcpy(buffer_.get(), encoded_frame.data_, length_); |
| 984 | called_->Set(); |
| 985 | } |
| 986 | |
| 987 | EventTypeWrapper Wait() { return called_->Wait(kDefaultTimeoutMs); } |
| 988 | |
| 989 | void ExpectEqualFrames(const EncodedFrameTestObserver& observer) { |
| 990 | ASSERT_EQ(length_, observer.length_) |
| 991 | << "Observed frames are of different lengths."; |
| 992 | EXPECT_EQ(frame_type_, observer.frame_type_) |
| 993 | << "Observed frames have different frame types."; |
| 994 | EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_)) |
| 995 | << "Observed encoded frames have different content."; |
| 996 | } |
| 997 | |
| 998 | private: |
| 999 | scoped_ptr<uint8_t[]> buffer_; |
| 1000 | size_t length_; |
| 1001 | FrameType frame_type_; |
| 1002 | scoped_ptr<EventWrapper> called_; |
| 1003 | }; |
| 1004 | |
| 1005 | EncodedFrameTestObserver post_encode_observer; |
| 1006 | EncodedFrameTestObserver pre_decode_observer; |
| 1007 | |
| 1008 | test::DirectTransport sender_transport, receiver_transport; |
| 1009 | |
| 1010 | CreateCalls(Call::Config(&sender_transport), |
| 1011 | Call::Config(&receiver_transport)); |
| 1012 | |
| 1013 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 1014 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 1015 | |
| 1016 | CreateSendConfig(1); |
| 1017 | CreateMatchingReceiveConfigs(); |
| 1018 | send_config_.post_encode_callback = &post_encode_observer; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1019 | receive_configs_[0].pre_decode_callback = &pre_decode_observer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1020 | |
| 1021 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1022 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1023 | |
| 1024 | scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create( |
| 1025 | video_streams_[0].width, video_streams_[0].height)); |
| 1026 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 1027 | |
| 1028 | EXPECT_EQ(kEventSignaled, post_encode_observer.Wait()) |
| 1029 | << "Timed out while waiting for send-side encoded-frame callback."; |
| 1030 | |
| 1031 | EXPECT_EQ(kEventSignaled, pre_decode_observer.Wait()) |
| 1032 | << "Timed out while waiting for pre-decode encoded-frame callback."; |
| 1033 | |
| 1034 | post_encode_observer.ExpectEqualFrames(pre_decode_observer); |
| 1035 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1036 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1037 | |
| 1038 | sender_transport.StopSending(); |
| 1039 | receiver_transport.StopSending(); |
| 1040 | |
| 1041 | DestroyStreams(); |
| 1042 | } |
| 1043 | |
| 1044 | TEST_F(EndToEndTest, ReceiveStreamSendsRemb) { |
| 1045 | class RembObserver : public test::EndToEndTest { |
| 1046 | public: |
| 1047 | RembObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
| 1048 | |
| 1049 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 1050 | size_t length) OVERRIDE { |
| 1051 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 1052 | EXPECT_TRUE(parser.IsValid()); |
| 1053 | |
| 1054 | bool received_psfb = false; |
| 1055 | bool received_remb = false; |
| 1056 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 1057 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 1058 | if (packet_type == RTCPUtility::kRtcpPsfbRembCode) { |
| 1059 | const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| 1060 | EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc); |
| 1061 | received_psfb = true; |
| 1062 | } else if (packet_type == RTCPUtility::kRtcpPsfbRembItemCode) { |
| 1063 | const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| 1064 | EXPECT_GT(packet.REMBItem.BitRate, 0u); |
| 1065 | EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u); |
| 1066 | EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]); |
| 1067 | received_remb = true; |
| 1068 | } |
| 1069 | packet_type = parser.Iterate(); |
| 1070 | } |
| 1071 | if (received_psfb && received_remb) |
| 1072 | observation_complete_->Set(); |
| 1073 | return SEND_PACKET; |
| 1074 | } |
| 1075 | virtual void PerformTest() OVERRIDE { |
| 1076 | EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for a " |
| 1077 | "receiver RTCP REMB packet to be " |
| 1078 | "sent."; |
| 1079 | } |
| 1080 | } test; |
| 1081 | |
| 1082 | RunBaseTest(&test); |
| 1083 | } |
| 1084 | |
| 1085 | void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) { |
| 1086 | static const int kNumRtcpReportPacketsToObserve = 5; |
| 1087 | class RtcpXrObserver : public test::EndToEndTest { |
| 1088 | public: |
| 1089 | explicit RtcpXrObserver(bool enable_rrtr) |
| 1090 | : EndToEndTest(kDefaultTimeoutMs), |
| 1091 | enable_rrtr_(enable_rrtr), |
| 1092 | sent_rtcp_sr_(0), |
| 1093 | sent_rtcp_rr_(0), |
| 1094 | sent_rtcp_rrtr_(0), |
| 1095 | sent_rtcp_dlrr_(0) {} |
| 1096 | |
| 1097 | private: |
| 1098 | // Receive stream should send RR packets (and RRTR packets if enabled). |
| 1099 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 1100 | size_t length) OVERRIDE { |
| 1101 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 1102 | EXPECT_TRUE(parser.IsValid()); |
| 1103 | |
| 1104 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 1105 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 1106 | if (packet_type == RTCPUtility::kRtcpRrCode) { |
| 1107 | ++sent_rtcp_rr_; |
| 1108 | } else if (packet_type == |
| 1109 | RTCPUtility::kRtcpXrReceiverReferenceTimeCode) { |
| 1110 | ++sent_rtcp_rrtr_; |
| 1111 | } |
| 1112 | EXPECT_NE(packet_type, RTCPUtility::kRtcpSrCode); |
| 1113 | EXPECT_NE(packet_type, RTCPUtility::kRtcpXrDlrrReportBlockItemCode); |
| 1114 | packet_type = parser.Iterate(); |
| 1115 | } |
| 1116 | return SEND_PACKET; |
| 1117 | } |
| 1118 | // Send stream should send SR packets (and DLRR packets if enabled). |
| 1119 | virtual Action OnSendRtcp(const uint8_t* packet, size_t length) { |
| 1120 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 1121 | EXPECT_TRUE(parser.IsValid()); |
| 1122 | |
| 1123 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 1124 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 1125 | if (packet_type == RTCPUtility::kRtcpSrCode) { |
| 1126 | ++sent_rtcp_sr_; |
| 1127 | } else if (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) { |
| 1128 | ++sent_rtcp_dlrr_; |
| 1129 | } |
| 1130 | EXPECT_NE(packet_type, RTCPUtility::kRtcpXrReceiverReferenceTimeCode); |
| 1131 | packet_type = parser.Iterate(); |
| 1132 | } |
| 1133 | if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve && |
| 1134 | sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) { |
| 1135 | if (enable_rrtr_) { |
| 1136 | EXPECT_GT(sent_rtcp_rrtr_, 0); |
| 1137 | EXPECT_GT(sent_rtcp_dlrr_, 0); |
| 1138 | } else { |
| 1139 | EXPECT_EQ(0, sent_rtcp_rrtr_); |
| 1140 | EXPECT_EQ(0, sent_rtcp_dlrr_); |
| 1141 | } |
| 1142 | observation_complete_->Set(); |
| 1143 | } |
| 1144 | return SEND_PACKET; |
| 1145 | } |
| 1146 | |
| 1147 | virtual void ModifyConfigs( |
| 1148 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1149 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1150 | std::vector<VideoStream>* video_streams) OVERRIDE { |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1151 | (*receive_configs)[0].rtp.rtcp_mode = newapi::kRtcpReducedSize; |
| 1152 | (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = |
| 1153 | enable_rrtr_; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1154 | } |
| 1155 | |
| 1156 | virtual void PerformTest() OVERRIDE { |
| 1157 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1158 | << "Timed out while waiting for RTCP SR/RR packets to be sent."; |
| 1159 | } |
| 1160 | |
| 1161 | bool enable_rrtr_; |
| 1162 | int sent_rtcp_sr_; |
| 1163 | int sent_rtcp_rr_; |
| 1164 | int sent_rtcp_rrtr_; |
| 1165 | int sent_rtcp_dlrr_; |
| 1166 | } test(enable_rrtr); |
| 1167 | |
| 1168 | RunBaseTest(&test); |
| 1169 | } |
| 1170 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1171 | void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs, |
| 1172 | bool send_single_ssrc_first) { |
| 1173 | class SendsSetSsrcs : public test::EndToEndTest { |
| 1174 | public: |
| 1175 | SendsSetSsrcs(const uint32_t* ssrcs, |
| 1176 | size_t num_ssrcs, |
| 1177 | bool send_single_ssrc_first) |
| 1178 | : EndToEndTest(kDefaultTimeoutMs), |
| 1179 | num_ssrcs_(num_ssrcs), |
| 1180 | send_single_ssrc_first_(send_single_ssrc_first), |
| 1181 | ssrcs_to_observe_(num_ssrcs), |
| 1182 | expect_single_ssrc_(send_single_ssrc_first) { |
| 1183 | for (size_t i = 0; i < num_ssrcs; ++i) |
| 1184 | valid_ssrcs_[ssrcs[i]] = true; |
| 1185 | } |
| 1186 | |
| 1187 | private: |
| 1188 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1189 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 1190 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1191 | |
| 1192 | EXPECT_TRUE(valid_ssrcs_[header.ssrc]) |
| 1193 | << "Received unknown SSRC: " << header.ssrc; |
| 1194 | |
| 1195 | if (!valid_ssrcs_[header.ssrc]) |
| 1196 | observation_complete_->Set(); |
| 1197 | |
| 1198 | if (!is_observed_[header.ssrc]) { |
| 1199 | is_observed_[header.ssrc] = true; |
| 1200 | --ssrcs_to_observe_; |
| 1201 | if (expect_single_ssrc_) { |
| 1202 | expect_single_ssrc_ = false; |
| 1203 | observation_complete_->Set(); |
| 1204 | } |
| 1205 | } |
| 1206 | |
| 1207 | if (ssrcs_to_observe_ == 0) |
| 1208 | observation_complete_->Set(); |
| 1209 | |
| 1210 | return SEND_PACKET; |
| 1211 | } |
| 1212 | |
| 1213 | virtual size_t GetNumStreams() const OVERRIDE { return num_ssrcs_; } |
| 1214 | |
| 1215 | virtual void ModifyConfigs( |
| 1216 | VideoSendStream::Config* send_config, |
| 1217 | std::vector<VideoReceiveStream::Config>* receive_configs, |
| 1218 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 1219 | if (num_ssrcs_ > 1) { |
| 1220 | // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. |
| 1221 | for (size_t i = 0; i < video_streams->size(); ++i) { |
| 1222 | (*video_streams)[i].min_bitrate_bps = 10000; |
| 1223 | (*video_streams)[i].target_bitrate_bps = 15000; |
| 1224 | (*video_streams)[i].max_bitrate_bps = 20000; |
| 1225 | } |
| 1226 | } |
| 1227 | |
| 1228 | all_streams_ = *video_streams; |
| 1229 | if (send_single_ssrc_first_) |
| 1230 | video_streams->resize(1); |
| 1231 | } |
| 1232 | |
| 1233 | virtual void OnStreamsCreated( |
| 1234 | VideoSendStream* send_stream, |
| 1235 | const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { |
| 1236 | send_stream_ = send_stream; |
| 1237 | } |
| 1238 | |
| 1239 | virtual void PerformTest() OVERRIDE { |
| 1240 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1241 | << "Timed out while waiting for " |
| 1242 | << (send_single_ssrc_first_ ? "first SSRC." : "SSRCs."); |
| 1243 | |
| 1244 | if (send_single_ssrc_first_) { |
| 1245 | // Set full simulcast and continue with the rest of the SSRCs. |
| 1246 | send_stream_->ReconfigureVideoEncoder(all_streams_, NULL); |
| 1247 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1248 | << "Timed out while waiting on additional SSRCs."; |
| 1249 | } |
| 1250 | } |
| 1251 | |
| 1252 | private: |
| 1253 | std::map<uint32_t, bool> valid_ssrcs_; |
| 1254 | std::map<uint32_t, bool> is_observed_; |
| 1255 | |
| 1256 | const size_t num_ssrcs_; |
| 1257 | const bool send_single_ssrc_first_; |
| 1258 | |
| 1259 | size_t ssrcs_to_observe_; |
| 1260 | bool expect_single_ssrc_; |
| 1261 | |
| 1262 | VideoSendStream* send_stream_; |
| 1263 | std::vector<VideoStream> all_streams_; |
| 1264 | } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first); |
| 1265 | |
| 1266 | RunBaseTest(&test); |
| 1267 | } |
| 1268 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1269 | TEST_F(EndToEndTest, GetStats) { |
| 1270 | class StatsObserver : public test::EndToEndTest, public I420FrameCallback { |
| 1271 | public: |
| 1272 | StatsObserver() |
| 1273 | : EndToEndTest(kLongTimeoutMs), |
| 1274 | receive_stream_(NULL), |
| 1275 | send_stream_(NULL), |
| 1276 | expected_receive_ssrc_(), |
| 1277 | expected_send_ssrcs_(), |
| 1278 | check_stats_event_(EventWrapper::Create()) {} |
| 1279 | |
| 1280 | private: |
| 1281 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1282 | check_stats_event_->Set(); |
| 1283 | return SEND_PACKET; |
| 1284 | } |
| 1285 | |
| 1286 | virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1287 | check_stats_event_->Set(); |
| 1288 | return SEND_PACKET; |
| 1289 | } |
| 1290 | |
| 1291 | virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1292 | check_stats_event_->Set(); |
| 1293 | return SEND_PACKET; |
| 1294 | } |
| 1295 | |
| 1296 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 1297 | size_t length) OVERRIDE { |
| 1298 | check_stats_event_->Set(); |
| 1299 | return SEND_PACKET; |
| 1300 | } |
| 1301 | |
| 1302 | virtual void FrameCallback(I420VideoFrame* video_frame) OVERRIDE { |
| 1303 | // Ensure that we have at least 5ms send side delay. |
| 1304 | int64_t render_time = video_frame->render_time_ms(); |
| 1305 | if (render_time > 0) |
| 1306 | video_frame->set_render_time_ms(render_time - 5); |
| 1307 | } |
| 1308 | |
| 1309 | bool CheckReceiveStats() { |
| 1310 | assert(receive_stream_ != NULL); |
| 1311 | VideoReceiveStream::Stats stats = receive_stream_->GetStats(); |
| 1312 | EXPECT_EQ(expected_receive_ssrc_, stats.ssrc); |
| 1313 | |
| 1314 | // Make sure all fields have been populated. |
| 1315 | |
| 1316 | receive_stats_filled_["IncomingRate"] |= |
| 1317 | stats.network_frame_rate != 0 || stats.bitrate_bps != 0; |
| 1318 | |
| 1319 | receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0; |
| 1320 | |
| 1321 | receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0; |
| 1322 | |
| 1323 | receive_stats_filled_["StatisticsUpdated"] |= |
| 1324 | stats.rtcp_stats.cumulative_lost != 0 || |
| 1325 | stats.rtcp_stats.extended_max_sequence_number != 0 || |
| 1326 | stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0; |
| 1327 | |
| 1328 | receive_stats_filled_["DataCountersUpdated"] |= |
| 1329 | stats.rtp_stats.bytes != 0 || stats.rtp_stats.fec_packets != 0 || |
| 1330 | stats.rtp_stats.header_bytes != 0 || stats.rtp_stats.packets != 0 || |
| 1331 | stats.rtp_stats.padding_bytes != 0 || |
| 1332 | stats.rtp_stats.retransmitted_packets != 0; |
| 1333 | |
| 1334 | receive_stats_filled_["CodecStats"] |= |
| 1335 | stats.avg_delay_ms != 0 || stats.discarded_packets != 0 || |
| 1336 | stats.key_frames != 0 || stats.delta_frames != 0; |
| 1337 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1338 | return AllStatsFilled(receive_stats_filled_); |
| 1339 | } |
| 1340 | |
| 1341 | bool CheckSendStats() { |
| 1342 | assert(send_stream_ != NULL); |
| 1343 | VideoSendStream::Stats stats = send_stream_->GetStats(); |
| 1344 | |
| 1345 | send_stats_filled_["NumStreams"] |= |
| 1346 | stats.substreams.size() == expected_send_ssrcs_.size(); |
| 1347 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1348 | for (std::map<uint32_t, StreamStats>::const_iterator it = |
| 1349 | stats.substreams.begin(); |
| 1350 | it != stats.substreams.end(); |
| 1351 | ++it) { |
| 1352 | EXPECT_TRUE(expected_send_ssrcs_.find(it->first) != |
| 1353 | expected_send_ssrcs_.end()); |
| 1354 | |
| 1355 | send_stats_filled_[CompoundKey("IncomingRate", it->first)] |= |
| 1356 | stats.input_frame_rate != 0; |
| 1357 | |
| 1358 | const StreamStats& stream_stats = it->second; |
| 1359 | |
| 1360 | send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |= |
| 1361 | stream_stats.rtcp_stats.cumulative_lost != 0 || |
| 1362 | stream_stats.rtcp_stats.extended_max_sequence_number != 0 || |
| 1363 | stream_stats.rtcp_stats.fraction_lost != 0; |
| 1364 | |
| 1365 | send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |= |
| 1366 | stream_stats.rtp_stats.fec_packets != 0 || |
| 1367 | stream_stats.rtp_stats.padding_bytes != 0 || |
| 1368 | stream_stats.rtp_stats.retransmitted_packets != 0 || |
| 1369 | stream_stats.rtp_stats.packets != 0; |
| 1370 | |
| 1371 | send_stats_filled_[CompoundKey("BitrateStatisticsObserver", |
| 1372 | it->first)] |= |
| 1373 | stream_stats.bitrate_bps != 0; |
| 1374 | |
| 1375 | send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |= |
| 1376 | stream_stats.delta_frames != 0 || stream_stats.key_frames != 0; |
| 1377 | |
| 1378 | send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |= |
| 1379 | stats.encode_frame_rate != 0; |
stefan@webrtc.org | 168f23f | 2014-07-11 13:44:02 +0000 | [diff] [blame] | 1380 | |
| 1381 | send_stats_filled_[CompoundKey("Delay", it->first)] |= |
| 1382 | stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1383 | } |
| 1384 | |
| 1385 | return AllStatsFilled(send_stats_filled_); |
| 1386 | } |
| 1387 | |
| 1388 | std::string CompoundKey(const char* name, uint32_t ssrc) { |
| 1389 | std::ostringstream oss; |
| 1390 | oss << name << "_" << ssrc; |
| 1391 | return oss.str(); |
| 1392 | } |
| 1393 | |
| 1394 | bool AllStatsFilled(const std::map<std::string, bool>& stats_map) { |
| 1395 | for (std::map<std::string, bool>::const_iterator it = stats_map.begin(); |
| 1396 | it != stats_map.end(); |
| 1397 | ++it) { |
| 1398 | if (!it->second) |
| 1399 | return false; |
| 1400 | } |
| 1401 | return true; |
| 1402 | } |
| 1403 | |
| 1404 | virtual void ModifyConfigs( |
| 1405 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1406 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1407 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 1408 | send_config->pre_encode_callback = this; // Used to inject delay. |
| 1409 | send_config->rtp.c_name = "SomeCName"; |
| 1410 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1411 | expected_receive_ssrc_ = (*receive_configs)[0].rtp.local_ssrc; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1412 | const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs; |
| 1413 | for (size_t i = 0; i < ssrcs.size(); ++i) |
| 1414 | expected_send_ssrcs_.insert(ssrcs[i]); |
| 1415 | |
| 1416 | expected_cname_ = send_config->rtp.c_name; |
| 1417 | } |
| 1418 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1419 | virtual void OnStreamsCreated( |
| 1420 | VideoSendStream* send_stream, |
| 1421 | const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1422 | send_stream_ = send_stream; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1423 | receive_stream_ = receive_streams[0]; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1424 | } |
| 1425 | |
| 1426 | virtual void PerformTest() OVERRIDE { |
| 1427 | Clock* clock = Clock::GetRealTimeClock(); |
| 1428 | int64_t now = clock->TimeInMilliseconds(); |
| 1429 | int64_t stop_time = now + test::CallTest::kLongTimeoutMs; |
| 1430 | bool receive_ok = false; |
| 1431 | bool send_ok = false; |
| 1432 | |
| 1433 | while (now < stop_time) { |
| 1434 | if (!receive_ok) |
| 1435 | receive_ok = CheckReceiveStats(); |
| 1436 | if (!send_ok) |
| 1437 | send_ok = CheckSendStats(); |
| 1438 | |
| 1439 | if (receive_ok && send_ok) |
| 1440 | return; |
| 1441 | |
| 1442 | int64_t time_until_timout_ = stop_time - now; |
| 1443 | if (time_until_timout_ > 0) |
| 1444 | check_stats_event_->Wait(time_until_timout_); |
| 1445 | now = clock->TimeInMilliseconds(); |
| 1446 | } |
| 1447 | |
| 1448 | ADD_FAILURE() << "Timed out waiting for filled stats."; |
| 1449 | for (std::map<std::string, bool>::const_iterator it = |
| 1450 | receive_stats_filled_.begin(); |
| 1451 | it != receive_stats_filled_.end(); |
| 1452 | ++it) { |
| 1453 | if (!it->second) { |
| 1454 | ADD_FAILURE() << "Missing receive stats: " << it->first; |
| 1455 | } |
| 1456 | } |
| 1457 | |
| 1458 | for (std::map<std::string, bool>::const_iterator it = |
| 1459 | send_stats_filled_.begin(); |
| 1460 | it != send_stats_filled_.end(); |
| 1461 | ++it) { |
| 1462 | if (!it->second) { |
| 1463 | ADD_FAILURE() << "Missing send stats: " << it->first; |
| 1464 | } |
| 1465 | } |
| 1466 | } |
| 1467 | |
| 1468 | VideoReceiveStream* receive_stream_; |
| 1469 | std::map<std::string, bool> receive_stats_filled_; |
| 1470 | |
| 1471 | VideoSendStream* send_stream_; |
| 1472 | std::map<std::string, bool> send_stats_filled_; |
| 1473 | |
| 1474 | uint32_t expected_receive_ssrc_; |
| 1475 | std::set<uint32_t> expected_send_ssrcs_; |
| 1476 | std::string expected_cname_; |
| 1477 | |
| 1478 | scoped_ptr<EventWrapper> check_stats_event_; |
| 1479 | } test; |
| 1480 | |
| 1481 | RunBaseTest(&test); |
| 1482 | } |
| 1483 | |
| 1484 | TEST_F(EndToEndTest, ReceiverReferenceTimeReportEnabled) { |
| 1485 | TestXrReceiverReferenceTimeReport(true); |
| 1486 | } |
| 1487 | |
| 1488 | TEST_F(EndToEndTest, ReceiverReferenceTimeReportDisabled) { |
| 1489 | TestXrReceiverReferenceTimeReport(false); |
| 1490 | } |
| 1491 | |
| 1492 | TEST_F(EndToEndTest, TestReceivedRtpPacketStats) { |
| 1493 | static const size_t kNumRtpPacketsToSend = 5; |
| 1494 | class ReceivedRtpStatsObserver : public test::EndToEndTest { |
| 1495 | public: |
| 1496 | ReceivedRtpStatsObserver() |
| 1497 | : EndToEndTest(kDefaultTimeoutMs), |
| 1498 | receive_stream_(NULL), |
| 1499 | sent_rtp_(0) {} |
| 1500 | |
| 1501 | private: |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1502 | virtual void OnStreamsCreated( |
| 1503 | VideoSendStream* send_stream, |
| 1504 | const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { |
| 1505 | receive_stream_ = receive_streams[0]; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1506 | } |
| 1507 | |
| 1508 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1509 | if (sent_rtp_ >= kNumRtpPacketsToSend) { |
| 1510 | VideoReceiveStream::Stats stats = receive_stream_->GetStats(); |
| 1511 | if (kNumRtpPacketsToSend == stats.rtp_stats.packets) { |
| 1512 | observation_complete_->Set(); |
| 1513 | } |
| 1514 | return DROP_PACKET; |
| 1515 | } |
| 1516 | ++sent_rtp_; |
| 1517 | return SEND_PACKET; |
| 1518 | } |
| 1519 | |
| 1520 | virtual void PerformTest() OVERRIDE { |
| 1521 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1522 | << "Timed out while verifying number of received RTP packets."; |
| 1523 | } |
| 1524 | |
| 1525 | VideoReceiveStream* receive_stream_; |
| 1526 | uint32_t sent_rtp_; |
| 1527 | } test; |
| 1528 | |
| 1529 | RunBaseTest(&test); |
| 1530 | } |
| 1531 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1532 | TEST_F(EndToEndTest, SendsSetSsrc) { TestSendsSetSsrcs(1, false); } |
| 1533 | |
| 1534 | TEST_F(EndToEndTest, SendsSetSimulcastSsrcs) { |
| 1535 | TestSendsSetSsrcs(kNumSsrcs, false); |
| 1536 | } |
| 1537 | |
| 1538 | TEST_F(EndToEndTest, CanSwitchToUseAllSsrcs) { |
| 1539 | TestSendsSetSsrcs(kNumSsrcs, true); |
| 1540 | } |
| 1541 | |
mflodman@webrtc.org | f946068 | 2014-07-24 16:41:25 +0000 | [diff] [blame^] | 1542 | TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) { |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1543 | class ObserveRedundantPayloads: public test::EndToEndTest { |
| 1544 | public: |
| 1545 | ObserveRedundantPayloads() |
| 1546 | : EndToEndTest(kDefaultTimeoutMs), ssrcs_to_observe_(kNumSsrcs) { |
| 1547 | for(size_t i = 0; i < kNumSsrcs; ++i) { |
| 1548 | registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true; |
| 1549 | } |
| 1550 | } |
| 1551 | |
| 1552 | private: |
| 1553 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1554 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 1555 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1556 | |
| 1557 | if (!registered_rtx_ssrc_[header.ssrc]) |
| 1558 | return SEND_PACKET; |
| 1559 | |
| 1560 | EXPECT_LE(static_cast<size_t>(header.headerLength + header.paddingLength), |
| 1561 | length); |
| 1562 | const bool packet_is_redundant_payload = |
| 1563 | static_cast<size_t>(header.headerLength + header.paddingLength) < |
| 1564 | length; |
| 1565 | |
| 1566 | if (!packet_is_redundant_payload) |
| 1567 | return SEND_PACKET; |
| 1568 | |
| 1569 | if (!observed_redundant_retransmission_[header.ssrc]) { |
| 1570 | observed_redundant_retransmission_[header.ssrc] = true; |
| 1571 | if (--ssrcs_to_observe_ == 0) |
| 1572 | observation_complete_->Set(); |
| 1573 | } |
| 1574 | |
| 1575 | return SEND_PACKET; |
| 1576 | } |
| 1577 | |
| 1578 | virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; } |
| 1579 | |
| 1580 | virtual void ModifyConfigs( |
| 1581 | VideoSendStream::Config* send_config, |
| 1582 | std::vector<VideoReceiveStream::Config>* receive_configs, |
| 1583 | std::vector<VideoStream>* video_streams) OVERRIDE { |
| 1584 | // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. |
| 1585 | for (size_t i = 0; i < video_streams->size(); ++i) { |
| 1586 | (*video_streams)[i].min_bitrate_bps = 10000; |
| 1587 | (*video_streams)[i].target_bitrate_bps = 15000; |
| 1588 | (*video_streams)[i].max_bitrate_bps = 20000; |
| 1589 | } |
| 1590 | // Significantly higher than max bitrates for all video streams -> forcing |
| 1591 | // padding to trigger redundant padding on all RTX SSRCs. |
| 1592 | send_config->rtp.min_transmit_bitrate_bps = 100000; |
| 1593 | |
| 1594 | send_config->rtp.rtx.payload_type = kSendRtxPayloadType; |
| 1595 | send_config->rtp.rtx.pad_with_redundant_payloads = true; |
| 1596 | |
| 1597 | for (size_t i = 0; i < kNumSsrcs; ++i) |
| 1598 | send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); |
| 1599 | } |
| 1600 | |
| 1601 | virtual void PerformTest() OVERRIDE { |
| 1602 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1603 | << "Timed out while waiting for redundant payloads on all SSRCs."; |
| 1604 | } |
| 1605 | |
| 1606 | private: |
| 1607 | size_t ssrcs_to_observe_; |
| 1608 | std::map<uint32_t, bool> observed_redundant_retransmission_; |
| 1609 | std::map<uint32_t, bool> registered_rtx_ssrc_; |
| 1610 | } test; |
| 1611 | |
| 1612 | RunBaseTest(&test); |
| 1613 | } |
| 1614 | |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1615 | void EndToEndTest::TestRtpStatePreservation(bool use_rtx) { |
| 1616 | static const uint32_t kMaxSequenceNumberGap = 100; |
| 1617 | static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90; |
| 1618 | class RtpSequenceObserver : public test::RtpRtcpObserver { |
| 1619 | public: |
| 1620 | RtpSequenceObserver(bool use_rtx) |
| 1621 | : test::RtpRtcpObserver(kDefaultTimeoutMs), |
| 1622 | crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| 1623 | ssrcs_to_observe_(kNumSsrcs) { |
| 1624 | for (size_t i = 0; i < kNumSsrcs; ++i) { |
| 1625 | configured_ssrcs_[kSendSsrcs[i]] = true; |
| 1626 | if (use_rtx) |
| 1627 | configured_ssrcs_[kSendRtxSsrcs[i]] = true; |
| 1628 | } |
| 1629 | } |
| 1630 | |
| 1631 | void ResetExpectedSsrcs(size_t num_expected_ssrcs) { |
| 1632 | CriticalSectionScoped lock(crit_.get()); |
| 1633 | ssrc_observed_.clear(); |
| 1634 | ssrcs_to_observe_ = num_expected_ssrcs; |
| 1635 | } |
| 1636 | |
| 1637 | private: |
| 1638 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1639 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 1640 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1641 | const uint32_t ssrc = header.ssrc; |
| 1642 | const uint16_t sequence_number = header.sequenceNumber; |
| 1643 | const uint32_t timestamp = header.timestamp; |
| 1644 | const bool only_padding = |
| 1645 | static_cast<size_t>(header.headerLength + header.paddingLength) == |
| 1646 | length; |
| 1647 | |
| 1648 | EXPECT_TRUE(configured_ssrcs_[ssrc]) |
| 1649 | << "Received SSRC that wasn't configured: " << ssrc; |
| 1650 | |
| 1651 | std::map<uint32_t, uint16_t>::iterator it = |
| 1652 | last_observed_sequence_number_.find(header.ssrc); |
| 1653 | if (it == last_observed_sequence_number_.end()) { |
| 1654 | last_observed_sequence_number_[ssrc] = sequence_number; |
| 1655 | last_observed_timestamp_[ssrc] = timestamp; |
| 1656 | } else { |
| 1657 | // Verify sequence numbers are reasonably close. |
| 1658 | uint32_t extended_sequence_number = sequence_number; |
| 1659 | // Check for roll-over. |
| 1660 | if (sequence_number < last_observed_sequence_number_[ssrc]) |
| 1661 | extended_sequence_number += 0xFFFFu + 1; |
| 1662 | EXPECT_LE( |
| 1663 | extended_sequence_number - last_observed_sequence_number_[ssrc], |
| 1664 | kMaxSequenceNumberGap) |
| 1665 | << "Gap in sequence numbers (" |
| 1666 | << last_observed_sequence_number_[ssrc] << " -> " << sequence_number |
| 1667 | << ") too large for SSRC: " << ssrc << "."; |
| 1668 | last_observed_sequence_number_[ssrc] = sequence_number; |
| 1669 | |
| 1670 | // TODO(pbos): Remove this check if we ever have monotonically |
| 1671 | // increasing timestamps. Right now padding packets add a delta which |
| 1672 | // can cause reordering between padding packets and regular packets, |
| 1673 | // hence we drop padding-only packets to not flake. |
| 1674 | if (only_padding) { |
| 1675 | // Verify that timestamps are reasonably close. |
| 1676 | uint64_t extended_timestamp = timestamp; |
| 1677 | // Check for roll-over. |
| 1678 | if (timestamp < last_observed_timestamp_[ssrc]) |
| 1679 | extended_timestamp += static_cast<uint64_t>(0xFFFFFFFFu) + 1; |
| 1680 | EXPECT_LE(extended_timestamp - last_observed_timestamp_[ssrc], |
| 1681 | kMaxTimestampGap) |
| 1682 | << "Gap in timestamps (" << last_observed_timestamp_[ssrc] |
| 1683 | << " -> " << timestamp << ") too large for SSRC: " << ssrc << "."; |
| 1684 | } |
| 1685 | last_observed_timestamp_[ssrc] = timestamp; |
| 1686 | } |
| 1687 | |
| 1688 | CriticalSectionScoped lock(crit_.get()); |
| 1689 | // Wait for media packets on all ssrcs. |
| 1690 | if (!ssrc_observed_[ssrc] && !only_padding) { |
| 1691 | ssrc_observed_[ssrc] = true; |
| 1692 | if (--ssrcs_to_observe_ == 0) |
| 1693 | observation_complete_->Set(); |
| 1694 | } |
| 1695 | |
| 1696 | return SEND_PACKET; |
| 1697 | } |
| 1698 | |
| 1699 | std::map<uint32_t, uint16_t> last_observed_sequence_number_; |
| 1700 | std::map<uint32_t, uint32_t> last_observed_timestamp_; |
| 1701 | std::map<uint32_t, bool> configured_ssrcs_; |
| 1702 | |
| 1703 | scoped_ptr<CriticalSectionWrapper> crit_; |
| 1704 | size_t ssrcs_to_observe_ GUARDED_BY(crit_); |
| 1705 | std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_); |
| 1706 | } observer(use_rtx); |
| 1707 | |
| 1708 | CreateCalls(Call::Config(observer.SendTransport()), |
| 1709 | Call::Config(observer.ReceiveTransport())); |
| 1710 | observer.SetReceivers(sender_call_->Receiver(), NULL); |
| 1711 | |
| 1712 | CreateSendConfig(kNumSsrcs); |
| 1713 | |
| 1714 | if (use_rtx) { |
| 1715 | for (size_t i = 0; i < kNumSsrcs; ++i) { |
| 1716 | send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); |
| 1717 | } |
| 1718 | send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; |
| 1719 | } |
| 1720 | |
| 1721 | // Lower bitrates so that all streams send initially. |
| 1722 | for (size_t i = 0; i < video_streams_.size(); ++i) { |
| 1723 | video_streams_[i].min_bitrate_bps = 10000; |
| 1724 | video_streams_[i].target_bitrate_bps = 15000; |
| 1725 | video_streams_[i].max_bitrate_bps = 20000; |
| 1726 | } |
| 1727 | |
| 1728 | CreateMatchingReceiveConfigs(); |
| 1729 | |
| 1730 | CreateStreams(); |
| 1731 | CreateFrameGeneratorCapturer(); |
| 1732 | |
| 1733 | Start(); |
| 1734 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1735 | << "Timed out waiting for all SSRCs to send packets."; |
| 1736 | |
| 1737 | // Test stream resetting more than once to make sure that the state doesn't |
| 1738 | // get set once (this could be due to using std::map::insert for instance). |
| 1739 | for (size_t i = 0; i < 3; ++i) { |
| 1740 | frame_generator_capturer_->Stop(); |
| 1741 | sender_call_->DestroyVideoSendStream(send_stream_); |
| 1742 | |
| 1743 | // Re-create VideoSendStream with only one stream. |
| 1744 | std::vector<VideoStream> one_stream = video_streams_; |
| 1745 | one_stream.resize(1); |
| 1746 | send_stream_ = |
| 1747 | sender_call_->CreateVideoSendStream(send_config_, one_stream, NULL); |
| 1748 | send_stream_->Start(); |
| 1749 | CreateFrameGeneratorCapturer(); |
| 1750 | frame_generator_capturer_->Start(); |
| 1751 | |
| 1752 | observer.ResetExpectedSsrcs(1); |
| 1753 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1754 | << "Timed out waiting for single RTP packet."; |
| 1755 | |
| 1756 | // Reconfigure back to use all streams. |
| 1757 | send_stream_->ReconfigureVideoEncoder(video_streams_, NULL); |
| 1758 | observer.ResetExpectedSsrcs(kNumSsrcs); |
| 1759 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1760 | << "Timed out waiting for all SSRCs to send packets."; |
| 1761 | |
| 1762 | // Reconfigure down to one stream. |
| 1763 | send_stream_->ReconfigureVideoEncoder(one_stream, NULL); |
| 1764 | observer.ResetExpectedSsrcs(1); |
| 1765 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1766 | << "Timed out waiting for single RTP packet."; |
| 1767 | |
| 1768 | // Reconfigure back to use all streams. |
| 1769 | send_stream_->ReconfigureVideoEncoder(video_streams_, NULL); |
| 1770 | observer.ResetExpectedSsrcs(kNumSsrcs); |
| 1771 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1772 | << "Timed out waiting for all SSRCs to send packets."; |
| 1773 | } |
| 1774 | |
| 1775 | observer.StopSending(); |
| 1776 | |
| 1777 | Stop(); |
| 1778 | DestroyStreams(); |
| 1779 | } |
| 1780 | |
| 1781 | TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpState) { |
| 1782 | TestRtpStatePreservation(false); |
| 1783 | } |
| 1784 | |
| 1785 | TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) { |
| 1786 | TestRtpStatePreservation(true); |
| 1787 | } |
| 1788 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1789 | } // namespace webrtc |