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" |
marpan@webrtc.org | 5b88317 | 2014-11-01 06:10:48 +0000 | [diff] [blame] | 22 | #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
| 23 | #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
| 24 | #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 25 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| 26 | #include "webrtc/system_wrappers/interface/event_wrapper.h" |
| 27 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| 28 | #include "webrtc/system_wrappers/interface/sleep.h" |
| 29 | #include "webrtc/test/call_test.h" |
| 30 | #include "webrtc/test/direct_transport.h" |
| 31 | #include "webrtc/test/encoder_settings.h" |
| 32 | #include "webrtc/test/fake_audio_device.h" |
| 33 | #include "webrtc/test/fake_decoder.h" |
| 34 | #include "webrtc/test/fake_encoder.h" |
| 35 | #include "webrtc/test/frame_generator.h" |
| 36 | #include "webrtc/test/frame_generator_capturer.h" |
| 37 | #include "webrtc/test/null_transport.h" |
| 38 | #include "webrtc/test/rtp_rtcp_observer.h" |
| 39 | #include "webrtc/test/testsupport/fileutils.h" |
andresp@webrtc.org | ab071da | 2014-09-18 08:58:15 +0000 | [diff] [blame] | 40 | #include "webrtc/test/testsupport/gtest_disable.h" |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 41 | #include "webrtc/test/testsupport/perf_test.h" |
| 42 | #include "webrtc/video/transport_adapter.h" |
pbos@webrtc.org | ab990ae | 2014-09-17 09:02:25 +0000 | [diff] [blame] | 43 | #include "webrtc/video_encoder.h" |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 44 | |
| 45 | namespace webrtc { |
| 46 | |
pbos@webrtc.org | 26c0c41 | 2014-09-03 16:17:12 +0000 | [diff] [blame] | 47 | static const unsigned long kSilenceTimeoutMs = 2000; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 48 | |
| 49 | class EndToEndTest : public test::CallTest { |
| 50 | public: |
| 51 | EndToEndTest() {} |
| 52 | |
| 53 | virtual ~EndToEndTest() { |
| 54 | EXPECT_EQ(NULL, send_stream_); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 55 | EXPECT_TRUE(receive_streams_.empty()); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 56 | } |
| 57 | |
| 58 | protected: |
pbos@webrtc.org | 26c0c41 | 2014-09-03 16:17:12 +0000 | [diff] [blame] | 59 | class UnusedTransport : public newapi::Transport { |
| 60 | private: |
| 61 | virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 62 | ADD_FAILURE() << "Unexpected RTP sent."; |
| 63 | return false; |
| 64 | } |
| 65 | |
| 66 | virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 67 | ADD_FAILURE() << "Unexpected RTCP sent."; |
| 68 | return false; |
| 69 | } |
| 70 | }; |
| 71 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 72 | void DecodesRetransmittedFrame(bool retransmit_over_rtx); |
| 73 | void ReceivesPliAndRecovers(int rtp_history_ms); |
| 74 | void RespectsRtcpMode(newapi::RtcpMode rtcp_mode); |
| 75 | void TestXrReceiverReferenceTimeReport(bool enable_rrtr); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 76 | void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 77 | void TestRtpStatePreservation(bool use_rtx); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 78 | }; |
| 79 | |
| 80 | TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) { |
| 81 | test::NullTransport transport; |
| 82 | CreateCalls(Call::Config(&transport), Call::Config(&transport)); |
| 83 | |
| 84 | CreateSendConfig(1); |
| 85 | CreateMatchingReceiveConfigs(); |
| 86 | |
| 87 | CreateStreams(); |
| 88 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 89 | receive_streams_[0]->Start(); |
| 90 | receive_streams_[0]->Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 91 | |
| 92 | DestroyStreams(); |
| 93 | } |
| 94 | |
| 95 | TEST_F(EndToEndTest, ReceiverCanBeStoppedTwice) { |
| 96 | test::NullTransport transport; |
| 97 | CreateCalls(Call::Config(&transport), Call::Config(&transport)); |
| 98 | |
| 99 | CreateSendConfig(1); |
| 100 | CreateMatchingReceiveConfigs(); |
| 101 | |
| 102 | CreateStreams(); |
| 103 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 104 | receive_streams_[0]->Stop(); |
| 105 | receive_streams_[0]->Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 106 | |
| 107 | DestroyStreams(); |
| 108 | } |
| 109 | |
| 110 | TEST_F(EndToEndTest, RendersSingleDelayedFrame) { |
| 111 | static const int kWidth = 320; |
| 112 | static const int kHeight = 240; |
| 113 | // This constant is chosen to be higher than the timeout in the video_render |
| 114 | // module. This makes sure that frames aren't dropped if there are no other |
| 115 | // frames in the queue. |
| 116 | static const int kDelayRenderCallbackMs = 1000; |
| 117 | |
| 118 | class Renderer : public VideoRenderer { |
| 119 | public: |
| 120 | Renderer() : event_(EventWrapper::Create()) {} |
| 121 | |
| 122 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 123 | int /*time_to_render_ms*/) OVERRIDE { |
| 124 | event_->Set(); |
| 125 | } |
| 126 | |
| 127 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 128 | |
| 129 | scoped_ptr<EventWrapper> event_; |
| 130 | } renderer; |
| 131 | |
| 132 | class TestFrameCallback : public I420FrameCallback { |
| 133 | public: |
| 134 | TestFrameCallback() : event_(EventWrapper::Create()) {} |
| 135 | |
| 136 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 137 | |
| 138 | private: |
| 139 | virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE { |
| 140 | SleepMs(kDelayRenderCallbackMs); |
| 141 | event_->Set(); |
| 142 | } |
| 143 | |
| 144 | scoped_ptr<EventWrapper> event_; |
| 145 | }; |
| 146 | |
| 147 | test::DirectTransport sender_transport, receiver_transport; |
| 148 | |
| 149 | CreateCalls(Call::Config(&sender_transport), |
| 150 | Call::Config(&receiver_transport)); |
| 151 | |
| 152 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 153 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 154 | |
| 155 | CreateSendConfig(1); |
| 156 | CreateMatchingReceiveConfigs(); |
| 157 | |
| 158 | TestFrameCallback pre_render_callback; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 159 | receive_configs_[0].pre_render_callback = &pre_render_callback; |
| 160 | receive_configs_[0].renderer = &renderer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 161 | |
| 162 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 163 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 164 | |
| 165 | // Create frames that are smaller than the send width/height, this is done to |
| 166 | // check that the callbacks are done after processing video. |
| 167 | scoped_ptr<test::FrameGenerator> frame_generator( |
| 168 | test::FrameGenerator::Create(kWidth, kHeight)); |
| 169 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 170 | EXPECT_EQ(kEventSignaled, pre_render_callback.Wait()) |
| 171 | << "Timed out while waiting for pre-render callback."; |
| 172 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 173 | << "Timed out while waiting for the frame to render."; |
| 174 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 175 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 176 | |
| 177 | sender_transport.StopSending(); |
| 178 | receiver_transport.StopSending(); |
| 179 | |
| 180 | DestroyStreams(); |
| 181 | } |
| 182 | |
| 183 | TEST_F(EndToEndTest, TransmitsFirstFrame) { |
| 184 | class Renderer : public VideoRenderer { |
| 185 | public: |
| 186 | Renderer() : event_(EventWrapper::Create()) {} |
| 187 | |
| 188 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 189 | int /*time_to_render_ms*/) OVERRIDE { |
| 190 | event_->Set(); |
| 191 | } |
| 192 | |
| 193 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 194 | |
| 195 | scoped_ptr<EventWrapper> event_; |
| 196 | } renderer; |
| 197 | |
| 198 | test::DirectTransport sender_transport, receiver_transport; |
| 199 | |
| 200 | CreateCalls(Call::Config(&sender_transport), |
| 201 | Call::Config(&receiver_transport)); |
| 202 | |
| 203 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 204 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 205 | |
| 206 | CreateSendConfig(1); |
| 207 | CreateMatchingReceiveConfigs(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 208 | receive_configs_[0].renderer = &renderer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 209 | |
| 210 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 211 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 212 | |
| 213 | scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create( |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 214 | encoder_config_.streams[0].width, encoder_config_.streams[0].height)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 215 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 216 | |
| 217 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 218 | << "Timed out while waiting for the frame to render."; |
| 219 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 220 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 221 | |
| 222 | sender_transport.StopSending(); |
| 223 | receiver_transport.StopSending(); |
| 224 | |
| 225 | DestroyStreams(); |
| 226 | } |
| 227 | |
marpan@webrtc.org | 5f1e2e4 | 2014-11-06 02:02:28 +0000 | [diff] [blame] | 228 | TEST_F(EndToEndTest, SendsAndReceivesVP9) { |
marpan@webrtc.org | 5b88317 | 2014-11-01 06:10:48 +0000 | [diff] [blame] | 229 | class VP9Observer : public test::EndToEndTest, public VideoRenderer { |
| 230 | public: |
| 231 | VP9Observer() |
| 232 | : EndToEndTest(2 * kDefaultTimeoutMs), |
| 233 | encoder_(VideoEncoder::Create(VideoEncoder::kVp9)), |
| 234 | decoder_(VP9Decoder::Create()), |
| 235 | frame_counter_(0) {} |
| 236 | |
| 237 | virtual void PerformTest() OVERRIDE { |
| 238 | EXPECT_EQ(kEventSignaled, Wait()) |
| 239 | << "Timed out while waiting for enough frames to be decoded."; |
| 240 | } |
| 241 | |
| 242 | virtual void ModifyConfigs( |
| 243 | VideoSendStream::Config* send_config, |
| 244 | std::vector<VideoReceiveStream::Config>* receive_configs, |
| 245 | VideoEncoderConfig* encoder_config) OVERRIDE { |
| 246 | send_config->encoder_settings.encoder = encoder_.get(); |
| 247 | send_config->encoder_settings.payload_name = "VP9"; |
| 248 | send_config->encoder_settings.payload_type = VCM_VP9_PAYLOAD_TYPE; |
| 249 | encoder_config->streams[0].min_bitrate_bps = 50000; |
| 250 | encoder_config->streams[0].target_bitrate_bps = |
| 251 | encoder_config->streams[0].max_bitrate_bps = 2000000; |
| 252 | |
| 253 | (*receive_configs)[0].renderer = this; |
| 254 | (*receive_configs)[0].decoders.resize(1); |
| 255 | (*receive_configs)[0].decoders[0].payload_type = |
| 256 | send_config->encoder_settings.payload_type; |
| 257 | (*receive_configs)[0].decoders[0].payload_name = |
| 258 | send_config->encoder_settings.payload_name; |
| 259 | (*receive_configs)[0].decoders[0].decoder = decoder_.get(); |
| 260 | } |
| 261 | |
| 262 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 263 | int time_to_render_ms) OVERRIDE { |
| 264 | const int kRequiredFrames = 500; |
| 265 | if (++frame_counter_ == kRequiredFrames) |
| 266 | observation_complete_->Set(); |
| 267 | } |
| 268 | |
| 269 | private: |
| 270 | scoped_ptr<webrtc::VideoEncoder> encoder_; |
| 271 | scoped_ptr<webrtc::VideoDecoder> decoder_; |
| 272 | int frame_counter_; |
| 273 | } test; |
| 274 | |
| 275 | RunBaseTest(&test); |
| 276 | } |
| 277 | |
stefan@webrtc.org | 79c3359 | 2014-08-06 09:24:53 +0000 | [diff] [blame] | 278 | TEST_F(EndToEndTest, SendsAndReceivesH264) { |
| 279 | class H264Observer : public test::EndToEndTest, public VideoRenderer { |
| 280 | public: |
| 281 | H264Observer() |
| 282 | : EndToEndTest(2 * kDefaultTimeoutMs), |
| 283 | fake_encoder_(Clock::GetRealTimeClock()), |
| 284 | frame_counter_(0) {} |
| 285 | |
| 286 | virtual void PerformTest() OVERRIDE { |
| 287 | EXPECT_EQ(kEventSignaled, Wait()) |
| 288 | << "Timed out while waiting for enough frames to be decoded."; |
| 289 | } |
| 290 | |
| 291 | virtual void ModifyConfigs( |
| 292 | VideoSendStream::Config* send_config, |
| 293 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 294 | VideoEncoderConfig* encoder_config) OVERRIDE { |
stefan@webrtc.org | 79c3359 | 2014-08-06 09:24:53 +0000 | [diff] [blame] | 295 | send_config->encoder_settings.encoder = &fake_encoder_; |
| 296 | send_config->encoder_settings.payload_name = "H264"; |
| 297 | send_config->encoder_settings.payload_type = kFakeSendPayloadType; |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 298 | encoder_config->streams[0].min_bitrate_bps = 50000; |
| 299 | encoder_config->streams[0].target_bitrate_bps = |
| 300 | encoder_config->streams[0].max_bitrate_bps = 2000000; |
stefan@webrtc.org | 79c3359 | 2014-08-06 09:24:53 +0000 | [diff] [blame] | 301 | |
| 302 | (*receive_configs)[0].renderer = this; |
pbos@webrtc.org | 776e6f2 | 2014-10-29 15:28:39 +0000 | [diff] [blame] | 303 | (*receive_configs)[0].decoders.resize(1); |
| 304 | (*receive_configs)[0].decoders[0].payload_type = |
stefan@webrtc.org | 79c3359 | 2014-08-06 09:24:53 +0000 | [diff] [blame] | 305 | send_config->encoder_settings.payload_type; |
pbos@webrtc.org | 776e6f2 | 2014-10-29 15:28:39 +0000 | [diff] [blame] | 306 | (*receive_configs)[0].decoders[0].payload_name = |
| 307 | send_config->encoder_settings.payload_name; |
| 308 | (*receive_configs)[0].decoders[0].decoder = &fake_decoder_; |
stefan@webrtc.org | 79c3359 | 2014-08-06 09:24:53 +0000 | [diff] [blame] | 309 | } |
| 310 | |
| 311 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 312 | int time_to_render_ms) OVERRIDE { |
| 313 | const int kRequiredFrames = 500; |
| 314 | if (++frame_counter_ == kRequiredFrames) |
| 315 | observation_complete_->Set(); |
| 316 | } |
| 317 | |
| 318 | private: |
| 319 | test::FakeH264Decoder fake_decoder_; |
| 320 | test::FakeH264Encoder fake_encoder_; |
| 321 | int frame_counter_; |
| 322 | } test; |
| 323 | |
| 324 | RunBaseTest(&test); |
| 325 | } |
| 326 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 327 | TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { |
| 328 | class SyncRtcpObserver : public test::EndToEndTest { |
| 329 | public: |
| 330 | SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
| 331 | |
| 332 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 333 | size_t length) OVERRIDE { |
| 334 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 335 | EXPECT_TRUE(parser.IsValid()); |
| 336 | uint32_t ssrc = 0; |
| 337 | ssrc |= static_cast<uint32_t>(packet[4]) << 24; |
| 338 | ssrc |= static_cast<uint32_t>(packet[5]) << 16; |
| 339 | ssrc |= static_cast<uint32_t>(packet[6]) << 8; |
| 340 | ssrc |= static_cast<uint32_t>(packet[7]) << 0; |
| 341 | EXPECT_EQ(kReceiverLocalSsrc, ssrc); |
| 342 | observation_complete_->Set(); |
| 343 | |
| 344 | return SEND_PACKET; |
| 345 | } |
| 346 | |
| 347 | virtual void PerformTest() OVERRIDE { |
| 348 | EXPECT_EQ(kEventSignaled, Wait()) |
| 349 | << "Timed out while waiting for a receiver RTCP packet to be sent."; |
| 350 | } |
| 351 | } test; |
| 352 | |
| 353 | RunBaseTest(&test); |
| 354 | } |
| 355 | |
| 356 | TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) { |
| 357 | static const int kNumberOfNacksToObserve = 2; |
| 358 | static const int kLossBurstSize = 2; |
| 359 | static const int kPacketsBetweenLossBursts = 9; |
| 360 | class NackObserver : public test::EndToEndTest { |
| 361 | public: |
| 362 | NackObserver() |
| 363 | : EndToEndTest(kLongTimeoutMs), |
| 364 | rtp_parser_(RtpHeaderParser::Create()), |
| 365 | sent_rtp_packets_(0), |
| 366 | packets_left_to_drop_(0), |
| 367 | nacks_left_(kNumberOfNacksToObserve) {} |
| 368 | |
| 369 | private: |
| 370 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 371 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 372 | EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 373 | |
| 374 | // Never drop retransmitted packets. |
| 375 | if (dropped_packets_.find(header.sequenceNumber) != |
| 376 | dropped_packets_.end()) { |
| 377 | retransmitted_packets_.insert(header.sequenceNumber); |
| 378 | if (nacks_left_ == 0 && |
| 379 | retransmitted_packets_.size() == dropped_packets_.size()) { |
| 380 | observation_complete_->Set(); |
| 381 | } |
| 382 | return SEND_PACKET; |
| 383 | } |
| 384 | |
| 385 | ++sent_rtp_packets_; |
| 386 | |
| 387 | // Enough NACKs received, stop dropping packets. |
| 388 | if (nacks_left_ == 0) |
| 389 | return SEND_PACKET; |
| 390 | |
| 391 | // Check if it's time for a new loss burst. |
| 392 | if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0) |
| 393 | packets_left_to_drop_ = kLossBurstSize; |
| 394 | |
| 395 | if (packets_left_to_drop_ > 0) { |
| 396 | --packets_left_to_drop_; |
| 397 | dropped_packets_.insert(header.sequenceNumber); |
| 398 | return DROP_PACKET; |
| 399 | } |
| 400 | |
| 401 | return SEND_PACKET; |
| 402 | } |
| 403 | |
| 404 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 405 | size_t length) OVERRIDE { |
| 406 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 407 | EXPECT_TRUE(parser.IsValid()); |
| 408 | |
| 409 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 410 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 411 | if (packet_type == RTCPUtility::kRtcpRtpfbNackCode) { |
| 412 | --nacks_left_; |
| 413 | break; |
| 414 | } |
| 415 | packet_type = parser.Iterate(); |
| 416 | } |
| 417 | return SEND_PACKET; |
| 418 | } |
| 419 | |
| 420 | virtual void ModifyConfigs( |
| 421 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 422 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 423 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 424 | send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 425 | (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 426 | } |
| 427 | |
| 428 | virtual void PerformTest() OVERRIDE { |
| 429 | EXPECT_EQ(kEventSignaled, Wait()) |
| 430 | << "Timed out waiting for packets to be NACKed, retransmitted and " |
| 431 | "rendered."; |
| 432 | } |
| 433 | |
| 434 | scoped_ptr<RtpHeaderParser> rtp_parser_; |
| 435 | std::set<uint16_t> dropped_packets_; |
| 436 | std::set<uint16_t> retransmitted_packets_; |
| 437 | uint64_t sent_rtp_packets_; |
| 438 | int packets_left_to_drop_; |
| 439 | int nacks_left_; |
| 440 | } test; |
| 441 | |
| 442 | RunBaseTest(&test); |
| 443 | } |
| 444 | |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 445 | TEST_F(EndToEndTest, CanReceiveFec) { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 446 | class FecRenderObserver : public test::EndToEndTest, public VideoRenderer { |
| 447 | public: |
| 448 | FecRenderObserver() |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 449 | : EndToEndTest(kDefaultTimeoutMs), state_(kFirstPacket) {} |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 450 | |
| 451 | private: |
| 452 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE |
| 453 | EXCLUSIVE_LOCKS_REQUIRED(crit_) { |
| 454 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 455 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 456 | |
| 457 | EXPECT_EQ(kRedPayloadType, header.payloadType); |
| 458 | int encapsulated_payload_type = |
| 459 | static_cast<int>(packet[header.headerLength]); |
| 460 | if (encapsulated_payload_type != kFakeSendPayloadType) |
| 461 | EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type); |
| 462 | |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 463 | if (protected_sequence_numbers_.count(header.sequenceNumber) != 0) { |
| 464 | // Retransmitted packet, should not count. |
| 465 | protected_sequence_numbers_.erase(header.sequenceNumber); |
| 466 | EXPECT_GT(protected_timestamps_.count(header.timestamp), 0u); |
| 467 | protected_timestamps_.erase(header.timestamp); |
| 468 | return SEND_PACKET; |
| 469 | } |
| 470 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 471 | switch (state_) { |
| 472 | case kFirstPacket: |
| 473 | state_ = kDropEveryOtherPacketUntilFec; |
| 474 | break; |
| 475 | case kDropEveryOtherPacketUntilFec: |
| 476 | if (encapsulated_payload_type == kUlpfecPayloadType) { |
| 477 | state_ = kDropNextMediaPacket; |
| 478 | return SEND_PACKET; |
| 479 | } |
| 480 | if (header.sequenceNumber % 2 == 0) |
| 481 | return DROP_PACKET; |
| 482 | break; |
| 483 | case kDropNextMediaPacket: |
| 484 | if (encapsulated_payload_type == kFakeSendPayloadType) { |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 485 | protected_sequence_numbers_.insert(header.sequenceNumber); |
| 486 | protected_timestamps_.insert(header.timestamp); |
| 487 | state_ = kDropEveryOtherPacketUntilFec; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 488 | return DROP_PACKET; |
| 489 | } |
| 490 | break; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 491 | } |
| 492 | |
| 493 | return SEND_PACKET; |
| 494 | } |
| 495 | |
| 496 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 497 | int time_to_render_ms) OVERRIDE { |
| 498 | CriticalSectionScoped lock(crit_.get()); |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 499 | // Rendering frame with timestamp of packet that was dropped -> FEC |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 500 | // protection worked. |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 501 | if (protected_timestamps_.count(video_frame.timestamp()) != 0) |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 502 | observation_complete_->Set(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 503 | } |
| 504 | |
| 505 | enum { |
| 506 | kFirstPacket, |
| 507 | kDropEveryOtherPacketUntilFec, |
| 508 | kDropNextMediaPacket, |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 509 | } state_; |
| 510 | |
| 511 | virtual void ModifyConfigs( |
| 512 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 513 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 514 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 515 | // TODO(pbos): Run this test with combined NACK/FEC enabled as well. |
| 516 | // int rtp_history_ms = 1000; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 517 | // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 518 | // send_config->rtp.nack.rtp_history_ms = rtp_history_ms; |
| 519 | send_config->rtp.fec.red_payload_type = kRedPayloadType; |
| 520 | send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| 521 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 522 | (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType; |
| 523 | (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| 524 | (*receive_configs)[0].renderer = this; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 525 | } |
| 526 | |
| 527 | virtual void PerformTest() OVERRIDE { |
| 528 | EXPECT_EQ(kEventSignaled, Wait()) |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 529 | << "Timed out waiting for dropped frames frames to be rendered."; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 530 | } |
| 531 | |
pbos@webrtc.org | a9c2d45 | 2014-11-13 14:40:15 +0000 | [diff] [blame] | 532 | std::set<uint32_t> protected_sequence_numbers_ GUARDED_BY(crit_); |
| 533 | std::set<uint32_t> protected_timestamps_ GUARDED_BY(crit_); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 534 | } test; |
| 535 | |
| 536 | RunBaseTest(&test); |
| 537 | } |
| 538 | |
| 539 | // This test drops second RTP packet with a marker bit set, makes sure it's |
| 540 | // retransmitted and renders. Retransmission SSRCs are also checked. |
| 541 | void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) { |
| 542 | static const int kDroppedFrameNumber = 2; |
| 543 | class RetransmissionObserver : public test::EndToEndTest, |
| 544 | public I420FrameCallback { |
| 545 | public: |
| 546 | explicit RetransmissionObserver(bool expect_rtx) |
| 547 | : EndToEndTest(kDefaultTimeoutMs), |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 548 | retransmission_ssrc_(expect_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]), |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 549 | retransmission_payload_type_(expect_rtx ? kSendRtxPayloadType |
| 550 | : kFakeSendPayloadType), |
| 551 | marker_bits_observed_(0), |
| 552 | retransmitted_timestamp_(0), |
| 553 | frame_retransmitted_(false) {} |
| 554 | |
| 555 | private: |
| 556 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 557 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 558 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 559 | |
| 560 | if (header.timestamp == retransmitted_timestamp_) { |
| 561 | EXPECT_EQ(retransmission_ssrc_, header.ssrc); |
| 562 | EXPECT_EQ(retransmission_payload_type_, header.payloadType); |
| 563 | frame_retransmitted_ = true; |
| 564 | return SEND_PACKET; |
| 565 | } |
| 566 | |
| 567 | EXPECT_EQ(kSendSsrcs[0], header.ssrc); |
| 568 | EXPECT_EQ(kFakeSendPayloadType, header.payloadType); |
| 569 | |
| 570 | // Found the second frame's final packet, drop this and expect a |
| 571 | // retransmission. |
| 572 | if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) { |
| 573 | retransmitted_timestamp_ = header.timestamp; |
| 574 | return DROP_PACKET; |
| 575 | } |
| 576 | |
| 577 | return SEND_PACKET; |
| 578 | } |
| 579 | |
| 580 | virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE { |
| 581 | CriticalSectionScoped lock(crit_.get()); |
| 582 | if (frame->timestamp() == retransmitted_timestamp_) { |
| 583 | EXPECT_TRUE(frame_retransmitted_); |
| 584 | observation_complete_->Set(); |
| 585 | } |
| 586 | } |
| 587 | |
| 588 | virtual void ModifyConfigs( |
| 589 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 590 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 591 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 592 | send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 593 | (*receive_configs)[0].pre_render_callback = this; |
| 594 | (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 595 | if (retransmission_ssrc_ == kSendRtxSsrcs[0]) { |
| 596 | send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 597 | send_config->rtp.rtx.payload_type = kSendRtxPayloadType; |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 598 | (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc = |
| 599 | kSendRtxSsrcs[0]; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 600 | (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].payload_type = |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 601 | kSendRtxPayloadType; |
| 602 | } |
| 603 | } |
| 604 | |
| 605 | virtual void PerformTest() OVERRIDE { |
| 606 | EXPECT_EQ(kEventSignaled, Wait()) |
| 607 | << "Timed out while waiting for retransmission to render."; |
| 608 | } |
| 609 | |
| 610 | const uint32_t retransmission_ssrc_; |
| 611 | const int retransmission_payload_type_; |
| 612 | int marker_bits_observed_; |
| 613 | uint32_t retransmitted_timestamp_; |
| 614 | bool frame_retransmitted_; |
| 615 | } test(retransmit_over_rtx); |
| 616 | |
| 617 | RunBaseTest(&test); |
| 618 | } |
| 619 | |
| 620 | TEST_F(EndToEndTest, DecodesRetransmittedFrame) { |
| 621 | DecodesRetransmittedFrame(false); |
| 622 | } |
| 623 | |
| 624 | TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) { |
| 625 | DecodesRetransmittedFrame(true); |
| 626 | } |
| 627 | |
andresp@webrtc.org | 0268611 | 2014-09-19 08:24:19 +0000 | [diff] [blame] | 628 | TEST_F(EndToEndTest, UsesFrameCallbacks) { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 629 | static const int kWidth = 320; |
| 630 | static const int kHeight = 240; |
| 631 | |
| 632 | class Renderer : public VideoRenderer { |
| 633 | public: |
| 634 | Renderer() : event_(EventWrapper::Create()) {} |
| 635 | |
| 636 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 637 | int /*time_to_render_ms*/) OVERRIDE { |
| 638 | EXPECT_EQ(0, *video_frame.buffer(kYPlane)) |
| 639 | << "Rendered frame should have zero luma which is applied by the " |
| 640 | "pre-render callback."; |
| 641 | event_->Set(); |
| 642 | } |
| 643 | |
| 644 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 645 | scoped_ptr<EventWrapper> event_; |
| 646 | } renderer; |
| 647 | |
| 648 | class TestFrameCallback : public I420FrameCallback { |
| 649 | public: |
| 650 | TestFrameCallback(int expected_luma_byte, int next_luma_byte) |
| 651 | : event_(EventWrapper::Create()), |
| 652 | expected_luma_byte_(expected_luma_byte), |
| 653 | next_luma_byte_(next_luma_byte) {} |
| 654 | |
| 655 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 656 | |
| 657 | private: |
| 658 | virtual void FrameCallback(I420VideoFrame* frame) { |
| 659 | EXPECT_EQ(kWidth, frame->width()) |
| 660 | << "Width not as expected, callback done before resize?"; |
| 661 | EXPECT_EQ(kHeight, frame->height()) |
| 662 | << "Height not as expected, callback done before resize?"; |
| 663 | |
| 664 | // Previous luma specified, observed luma should be fairly close. |
| 665 | if (expected_luma_byte_ != -1) { |
| 666 | EXPECT_NEAR(expected_luma_byte_, *frame->buffer(kYPlane), 10); |
| 667 | } |
| 668 | |
| 669 | memset(frame->buffer(kYPlane), |
| 670 | next_luma_byte_, |
| 671 | frame->allocated_size(kYPlane)); |
| 672 | |
| 673 | event_->Set(); |
| 674 | } |
| 675 | |
| 676 | scoped_ptr<EventWrapper> event_; |
| 677 | int expected_luma_byte_; |
| 678 | int next_luma_byte_; |
| 679 | }; |
| 680 | |
| 681 | TestFrameCallback pre_encode_callback(-1, 255); // Changes luma to 255. |
| 682 | TestFrameCallback pre_render_callback(255, 0); // Changes luma from 255 to 0. |
| 683 | |
| 684 | test::DirectTransport sender_transport, receiver_transport; |
| 685 | |
| 686 | CreateCalls(Call::Config(&sender_transport), |
| 687 | Call::Config(&receiver_transport)); |
| 688 | |
| 689 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 690 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 691 | |
| 692 | CreateSendConfig(1); |
pbos@webrtc.org | ab990ae | 2014-09-17 09:02:25 +0000 | [diff] [blame] | 693 | scoped_ptr<VideoEncoder> encoder( |
| 694 | VideoEncoder::Create(VideoEncoder::kVp8)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 695 | send_config_.encoder_settings.encoder = encoder.get(); |
| 696 | send_config_.encoder_settings.payload_name = "VP8"; |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 697 | ASSERT_EQ(1u, encoder_config_.streams.size()) << "Test setup error."; |
| 698 | encoder_config_.streams[0].width = kWidth; |
| 699 | encoder_config_.streams[0].height = kHeight; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 700 | send_config_.pre_encode_callback = &pre_encode_callback; |
| 701 | |
| 702 | CreateMatchingReceiveConfigs(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 703 | receive_configs_[0].pre_render_callback = &pre_render_callback; |
| 704 | receive_configs_[0].renderer = &renderer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 705 | |
| 706 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 707 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 708 | |
| 709 | // Create frames that are smaller than the send width/height, this is done to |
| 710 | // check that the callbacks are done after processing video. |
| 711 | scoped_ptr<test::FrameGenerator> frame_generator( |
| 712 | test::FrameGenerator::Create(kWidth / 2, kHeight / 2)); |
| 713 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 714 | |
| 715 | EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait()) |
| 716 | << "Timed out while waiting for pre-encode callback."; |
| 717 | EXPECT_EQ(kEventSignaled, pre_render_callback.Wait()) |
| 718 | << "Timed out while waiting for pre-render callback."; |
| 719 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 720 | << "Timed out while waiting for the frame to render."; |
| 721 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 722 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 723 | |
| 724 | sender_transport.StopSending(); |
| 725 | receiver_transport.StopSending(); |
| 726 | |
| 727 | DestroyStreams(); |
| 728 | } |
| 729 | |
| 730 | void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) { |
| 731 | static const int kPacketsToDrop = 1; |
| 732 | |
| 733 | class PliObserver : public test::EndToEndTest, public VideoRenderer { |
| 734 | public: |
| 735 | explicit PliObserver(int rtp_history_ms) |
| 736 | : EndToEndTest(kLongTimeoutMs), |
| 737 | rtp_history_ms_(rtp_history_ms), |
| 738 | nack_enabled_(rtp_history_ms > 0), |
| 739 | highest_dropped_timestamp_(0), |
| 740 | frames_to_drop_(0), |
| 741 | received_pli_(false) {} |
| 742 | |
| 743 | private: |
| 744 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 745 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 746 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 747 | |
| 748 | // Drop all retransmitted packets to force a PLI. |
| 749 | if (header.timestamp <= highest_dropped_timestamp_) |
| 750 | return DROP_PACKET; |
| 751 | |
| 752 | if (frames_to_drop_ > 0) { |
| 753 | highest_dropped_timestamp_ = header.timestamp; |
| 754 | --frames_to_drop_; |
| 755 | return DROP_PACKET; |
| 756 | } |
| 757 | |
| 758 | return SEND_PACKET; |
| 759 | } |
| 760 | |
| 761 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 762 | size_t length) OVERRIDE { |
| 763 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 764 | EXPECT_TRUE(parser.IsValid()); |
| 765 | |
| 766 | for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 767 | packet_type != RTCPUtility::kRtcpNotValidCode; |
| 768 | packet_type = parser.Iterate()) { |
| 769 | if (!nack_enabled_) |
| 770 | EXPECT_NE(packet_type, RTCPUtility::kRtcpRtpfbNackCode); |
| 771 | |
| 772 | if (packet_type == RTCPUtility::kRtcpPsfbPliCode) { |
| 773 | received_pli_ = true; |
| 774 | break; |
| 775 | } |
| 776 | } |
| 777 | return SEND_PACKET; |
| 778 | } |
| 779 | |
| 780 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 781 | int time_to_render_ms) OVERRIDE { |
| 782 | CriticalSectionScoped lock(crit_.get()); |
| 783 | if (received_pli_ && |
| 784 | video_frame.timestamp() > highest_dropped_timestamp_) { |
| 785 | observation_complete_->Set(); |
| 786 | } |
| 787 | if (!received_pli_) |
| 788 | frames_to_drop_ = kPacketsToDrop; |
| 789 | } |
| 790 | |
| 791 | virtual void ModifyConfigs( |
| 792 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 793 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 794 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 795 | send_config->rtp.nack.rtp_history_ms = rtp_history_ms_; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 796 | (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_; |
| 797 | (*receive_configs)[0].renderer = this; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 798 | } |
| 799 | |
| 800 | virtual void PerformTest() OVERRIDE { |
| 801 | EXPECT_EQ(kEventSignaled, Wait()) << "Timed out waiting for PLI to be " |
| 802 | "received and a frame to be " |
| 803 | "rendered afterwards."; |
| 804 | } |
| 805 | |
| 806 | int rtp_history_ms_; |
| 807 | bool nack_enabled_; |
| 808 | uint32_t highest_dropped_timestamp_; |
| 809 | int frames_to_drop_; |
| 810 | bool received_pli_; |
| 811 | } test(rtp_history_ms); |
| 812 | |
| 813 | RunBaseTest(&test); |
| 814 | } |
| 815 | |
| 816 | TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) { |
| 817 | ReceivesPliAndRecovers(1000); |
| 818 | } |
| 819 | |
| 820 | // TODO(pbos): Enable this when 2250 is resolved. |
| 821 | TEST_F(EndToEndTest, DISABLED_ReceivesPliAndRecoversWithoutNack) { |
| 822 | ReceivesPliAndRecovers(0); |
| 823 | } |
| 824 | |
| 825 | TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) { |
| 826 | class PacketInputObserver : public PacketReceiver { |
| 827 | public: |
| 828 | explicit PacketInputObserver(PacketReceiver* receiver) |
| 829 | : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {} |
| 830 | |
| 831 | EventTypeWrapper Wait() { |
| 832 | return delivered_packet_->Wait(kDefaultTimeoutMs); |
| 833 | } |
| 834 | |
| 835 | private: |
| 836 | virtual DeliveryStatus DeliverPacket(const uint8_t* packet, |
| 837 | size_t length) OVERRIDE { |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 838 | if (RtpHeaderParser::IsRtcp(packet, length)) { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 839 | return receiver_->DeliverPacket(packet, length); |
| 840 | } else { |
| 841 | DeliveryStatus delivery_status = |
| 842 | receiver_->DeliverPacket(packet, length); |
| 843 | EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status); |
| 844 | delivered_packet_->Set(); |
| 845 | return delivery_status; |
| 846 | } |
| 847 | } |
| 848 | |
| 849 | PacketReceiver* receiver_; |
| 850 | scoped_ptr<EventWrapper> delivered_packet_; |
| 851 | }; |
| 852 | |
| 853 | test::DirectTransport send_transport, receive_transport; |
| 854 | |
| 855 | CreateCalls(Call::Config(&send_transport), Call::Config(&receive_transport)); |
| 856 | PacketInputObserver input_observer(receiver_call_->Receiver()); |
| 857 | |
| 858 | send_transport.SetReceiver(&input_observer); |
| 859 | receive_transport.SetReceiver(sender_call_->Receiver()); |
| 860 | |
| 861 | CreateSendConfig(1); |
| 862 | CreateMatchingReceiveConfigs(); |
| 863 | |
| 864 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 865 | CreateFrameGeneratorCapturer(); |
| 866 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 867 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 868 | receiver_call_->DestroyVideoReceiveStream(receive_streams_[0]); |
| 869 | receive_streams_.clear(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 870 | |
| 871 | // Wait() waits for a received packet. |
| 872 | EXPECT_EQ(kEventSignaled, input_observer.Wait()); |
| 873 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 874 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 875 | |
| 876 | DestroyStreams(); |
| 877 | |
| 878 | send_transport.StopSending(); |
| 879 | receive_transport.StopSending(); |
| 880 | } |
| 881 | |
| 882 | void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) { |
| 883 | static const int kNumCompoundRtcpPacketsToObserve = 10; |
| 884 | class RtcpModeObserver : public test::EndToEndTest { |
| 885 | public: |
| 886 | explicit RtcpModeObserver(newapi::RtcpMode rtcp_mode) |
| 887 | : EndToEndTest(kDefaultTimeoutMs), |
| 888 | rtcp_mode_(rtcp_mode), |
| 889 | sent_rtp_(0), |
| 890 | sent_rtcp_(0) {} |
| 891 | |
| 892 | private: |
| 893 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 894 | if (++sent_rtp_ % 3 == 0) |
| 895 | return DROP_PACKET; |
| 896 | |
| 897 | return SEND_PACKET; |
| 898 | } |
| 899 | |
| 900 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 901 | size_t length) OVERRIDE { |
| 902 | ++sent_rtcp_; |
| 903 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 904 | EXPECT_TRUE(parser.IsValid()); |
| 905 | |
| 906 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 907 | bool has_report_block = false; |
| 908 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 909 | EXPECT_NE(RTCPUtility::kRtcpSrCode, packet_type); |
| 910 | if (packet_type == RTCPUtility::kRtcpRrCode) { |
| 911 | has_report_block = true; |
| 912 | break; |
| 913 | } |
| 914 | packet_type = parser.Iterate(); |
| 915 | } |
| 916 | |
| 917 | switch (rtcp_mode_) { |
| 918 | case newapi::kRtcpCompound: |
| 919 | if (!has_report_block) { |
| 920 | ADD_FAILURE() << "Received RTCP packet without receiver report for " |
| 921 | "kRtcpCompound."; |
| 922 | observation_complete_->Set(); |
| 923 | } |
| 924 | |
| 925 | if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve) |
| 926 | observation_complete_->Set(); |
| 927 | |
| 928 | break; |
| 929 | case newapi::kRtcpReducedSize: |
| 930 | if (!has_report_block) |
| 931 | observation_complete_->Set(); |
| 932 | break; |
| 933 | } |
| 934 | |
| 935 | return SEND_PACKET; |
| 936 | } |
| 937 | |
| 938 | virtual void ModifyConfigs( |
| 939 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 940 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 941 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 942 | send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 943 | (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| 944 | (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 945 | } |
| 946 | |
| 947 | virtual void PerformTest() OVERRIDE { |
| 948 | EXPECT_EQ(kEventSignaled, Wait()) |
| 949 | << (rtcp_mode_ == newapi::kRtcpCompound |
| 950 | ? "Timed out before observing enough compound packets." |
| 951 | : "Timed out before receiving a non-compound RTCP packet."); |
| 952 | } |
| 953 | |
| 954 | newapi::RtcpMode rtcp_mode_; |
| 955 | int sent_rtp_; |
| 956 | int sent_rtcp_; |
| 957 | } test(rtcp_mode); |
| 958 | |
| 959 | RunBaseTest(&test); |
| 960 | } |
| 961 | |
| 962 | TEST_F(EndToEndTest, UsesRtcpCompoundMode) { |
| 963 | RespectsRtcpMode(newapi::kRtcpCompound); |
| 964 | } |
| 965 | |
| 966 | TEST_F(EndToEndTest, UsesRtcpReducedSizeMode) { |
| 967 | RespectsRtcpMode(newapi::kRtcpReducedSize); |
| 968 | } |
| 969 | |
| 970 | // Test sets up a Call multiple senders with different resolutions and SSRCs. |
| 971 | // Another is set up to receive all three of these with different renderers. |
| 972 | // Each renderer verifies that it receives the expected resolution, and as soon |
| 973 | // as every renderer has received a frame, the test finishes. |
andresp@webrtc.org | 0268611 | 2014-09-19 08:24:19 +0000 | [diff] [blame] | 974 | TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 975 | static const size_t kNumStreams = 3; |
| 976 | |
| 977 | class VideoOutputObserver : public VideoRenderer { |
| 978 | public: |
| 979 | VideoOutputObserver(test::FrameGeneratorCapturer** capturer, |
| 980 | int width, |
| 981 | int height) |
| 982 | : capturer_(capturer), |
| 983 | width_(width), |
| 984 | height_(height), |
| 985 | done_(EventWrapper::Create()) {} |
| 986 | |
| 987 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 988 | int time_to_render_ms) OVERRIDE { |
| 989 | EXPECT_EQ(width_, video_frame.width()); |
| 990 | EXPECT_EQ(height_, video_frame.height()); |
| 991 | (*capturer_)->Stop(); |
| 992 | done_->Set(); |
| 993 | } |
| 994 | |
| 995 | EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); } |
| 996 | |
| 997 | private: |
| 998 | test::FrameGeneratorCapturer** capturer_; |
| 999 | int width_; |
| 1000 | int height_; |
| 1001 | scoped_ptr<EventWrapper> done_; |
| 1002 | }; |
| 1003 | |
| 1004 | struct { |
| 1005 | uint32_t ssrc; |
| 1006 | int width; |
| 1007 | int height; |
| 1008 | } codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}}; |
| 1009 | |
| 1010 | test::DirectTransport sender_transport, receiver_transport; |
| 1011 | scoped_ptr<Call> sender_call(Call::Create(Call::Config(&sender_transport))); |
| 1012 | scoped_ptr<Call> receiver_call( |
| 1013 | Call::Create(Call::Config(&receiver_transport))); |
| 1014 | sender_transport.SetReceiver(receiver_call->Receiver()); |
| 1015 | receiver_transport.SetReceiver(sender_call->Receiver()); |
| 1016 | |
| 1017 | VideoSendStream* send_streams[kNumStreams]; |
| 1018 | VideoReceiveStream* receive_streams[kNumStreams]; |
| 1019 | |
| 1020 | VideoOutputObserver* observers[kNumStreams]; |
| 1021 | test::FrameGeneratorCapturer* frame_generators[kNumStreams]; |
| 1022 | |
pbos@webrtc.org | ab990ae | 2014-09-17 09:02:25 +0000 | [diff] [blame] | 1023 | scoped_ptr<VideoEncoder> encoders[kNumStreams]; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1024 | for (size_t i = 0; i < kNumStreams; ++i) |
pbos@webrtc.org | ab990ae | 2014-09-17 09:02:25 +0000 | [diff] [blame] | 1025 | encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1026 | |
pbos@webrtc.org | 776e6f2 | 2014-10-29 15:28:39 +0000 | [diff] [blame] | 1027 | ScopedVector<VideoDecoder> allocated_decoders; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1028 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 1029 | uint32_t ssrc = codec_settings[i].ssrc; |
| 1030 | int width = codec_settings[i].width; |
| 1031 | int height = codec_settings[i].height; |
| 1032 | observers[i] = new VideoOutputObserver(&frame_generators[i], width, height); |
| 1033 | |
pbos@webrtc.org | bd249bc | 2014-07-07 04:45:15 +0000 | [diff] [blame] | 1034 | VideoSendStream::Config send_config; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1035 | send_config.rtp.ssrcs.push_back(ssrc); |
| 1036 | send_config.encoder_settings.encoder = encoders[i].get(); |
| 1037 | send_config.encoder_settings.payload_name = "VP8"; |
| 1038 | send_config.encoder_settings.payload_type = 124; |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1039 | VideoEncoderConfig encoder_config; |
| 1040 | encoder_config.streams = test::CreateVideoStreams(1); |
| 1041 | VideoStream* stream = &encoder_config.streams[0]; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1042 | stream->width = width; |
| 1043 | stream->height = height; |
| 1044 | stream->max_framerate = 5; |
| 1045 | stream->min_bitrate_bps = stream->target_bitrate_bps = |
| 1046 | stream->max_bitrate_bps = 100000; |
| 1047 | send_streams[i] = |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1048 | sender_call->CreateVideoSendStream(send_config, encoder_config); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1049 | send_streams[i]->Start(); |
| 1050 | |
pbos@webrtc.org | bd249bc | 2014-07-07 04:45:15 +0000 | [diff] [blame] | 1051 | VideoReceiveStream::Config receive_config; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1052 | receive_config.renderer = observers[i]; |
| 1053 | receive_config.rtp.remote_ssrc = ssrc; |
| 1054 | receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
pbos@webrtc.org | 776e6f2 | 2014-10-29 15:28:39 +0000 | [diff] [blame] | 1055 | VideoReceiveStream::Decoder decoder = |
| 1056 | test::CreateMatchingDecoder(send_config.encoder_settings); |
| 1057 | allocated_decoders.push_back(decoder.decoder); |
| 1058 | receive_config.decoders.push_back(decoder); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1059 | receive_streams[i] = |
| 1060 | receiver_call->CreateVideoReceiveStream(receive_config); |
| 1061 | receive_streams[i]->Start(); |
| 1062 | |
| 1063 | frame_generators[i] = test::FrameGeneratorCapturer::Create( |
| 1064 | send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock()); |
| 1065 | frame_generators[i]->Start(); |
| 1066 | } |
| 1067 | |
| 1068 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 1069 | EXPECT_EQ(kEventSignaled, observers[i]->Wait()) |
| 1070 | << "Timed out while waiting for observer " << i << " to render."; |
| 1071 | } |
| 1072 | |
| 1073 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 1074 | frame_generators[i]->Stop(); |
| 1075 | sender_call->DestroyVideoSendStream(send_streams[i]); |
| 1076 | receiver_call->DestroyVideoReceiveStream(receive_streams[i]); |
| 1077 | delete frame_generators[i]; |
| 1078 | delete observers[i]; |
| 1079 | } |
| 1080 | |
| 1081 | sender_transport.StopSending(); |
| 1082 | receiver_transport.StopSending(); |
| 1083 | } |
| 1084 | |
| 1085 | TEST_F(EndToEndTest, ObserversEncodedFrames) { |
| 1086 | class EncodedFrameTestObserver : public EncodedFrameObserver { |
| 1087 | public: |
| 1088 | EncodedFrameTestObserver() |
| 1089 | : length_(0), |
| 1090 | frame_type_(kFrameEmpty), |
| 1091 | called_(EventWrapper::Create()) {} |
| 1092 | virtual ~EncodedFrameTestObserver() {} |
| 1093 | |
| 1094 | virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { |
| 1095 | frame_type_ = encoded_frame.frame_type_; |
| 1096 | length_ = encoded_frame.length_; |
| 1097 | buffer_.reset(new uint8_t[length_]); |
| 1098 | memcpy(buffer_.get(), encoded_frame.data_, length_); |
| 1099 | called_->Set(); |
| 1100 | } |
| 1101 | |
| 1102 | EventTypeWrapper Wait() { return called_->Wait(kDefaultTimeoutMs); } |
| 1103 | |
| 1104 | void ExpectEqualFrames(const EncodedFrameTestObserver& observer) { |
| 1105 | ASSERT_EQ(length_, observer.length_) |
| 1106 | << "Observed frames are of different lengths."; |
| 1107 | EXPECT_EQ(frame_type_, observer.frame_type_) |
| 1108 | << "Observed frames have different frame types."; |
| 1109 | EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_)) |
| 1110 | << "Observed encoded frames have different content."; |
| 1111 | } |
| 1112 | |
| 1113 | private: |
| 1114 | scoped_ptr<uint8_t[]> buffer_; |
| 1115 | size_t length_; |
| 1116 | FrameType frame_type_; |
| 1117 | scoped_ptr<EventWrapper> called_; |
| 1118 | }; |
| 1119 | |
| 1120 | EncodedFrameTestObserver post_encode_observer; |
| 1121 | EncodedFrameTestObserver pre_decode_observer; |
| 1122 | |
| 1123 | test::DirectTransport sender_transport, receiver_transport; |
| 1124 | |
| 1125 | CreateCalls(Call::Config(&sender_transport), |
| 1126 | Call::Config(&receiver_transport)); |
| 1127 | |
| 1128 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 1129 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 1130 | |
| 1131 | CreateSendConfig(1); |
| 1132 | CreateMatchingReceiveConfigs(); |
| 1133 | send_config_.post_encode_callback = &post_encode_observer; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1134 | receive_configs_[0].pre_decode_callback = &pre_decode_observer; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1135 | |
| 1136 | CreateStreams(); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1137 | Start(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1138 | |
| 1139 | scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create( |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1140 | encoder_config_.streams[0].width, encoder_config_.streams[0].height)); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1141 | send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); |
| 1142 | |
| 1143 | EXPECT_EQ(kEventSignaled, post_encode_observer.Wait()) |
| 1144 | << "Timed out while waiting for send-side encoded-frame callback."; |
| 1145 | |
| 1146 | EXPECT_EQ(kEventSignaled, pre_decode_observer.Wait()) |
| 1147 | << "Timed out while waiting for pre-decode encoded-frame callback."; |
| 1148 | |
| 1149 | post_encode_observer.ExpectEqualFrames(pre_decode_observer); |
| 1150 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1151 | Stop(); |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1152 | |
| 1153 | sender_transport.StopSending(); |
| 1154 | receiver_transport.StopSending(); |
| 1155 | |
| 1156 | DestroyStreams(); |
| 1157 | } |
| 1158 | |
| 1159 | TEST_F(EndToEndTest, ReceiveStreamSendsRemb) { |
| 1160 | class RembObserver : public test::EndToEndTest { |
| 1161 | public: |
| 1162 | RembObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
| 1163 | |
| 1164 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 1165 | size_t length) OVERRIDE { |
| 1166 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 1167 | EXPECT_TRUE(parser.IsValid()); |
| 1168 | |
| 1169 | bool received_psfb = false; |
| 1170 | bool received_remb = false; |
| 1171 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 1172 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 1173 | if (packet_type == RTCPUtility::kRtcpPsfbRembCode) { |
| 1174 | const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| 1175 | EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc); |
| 1176 | received_psfb = true; |
| 1177 | } else if (packet_type == RTCPUtility::kRtcpPsfbRembItemCode) { |
| 1178 | const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| 1179 | EXPECT_GT(packet.REMBItem.BitRate, 0u); |
| 1180 | EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u); |
| 1181 | EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]); |
| 1182 | received_remb = true; |
| 1183 | } |
| 1184 | packet_type = parser.Iterate(); |
| 1185 | } |
| 1186 | if (received_psfb && received_remb) |
| 1187 | observation_complete_->Set(); |
| 1188 | return SEND_PACKET; |
| 1189 | } |
| 1190 | virtual void PerformTest() OVERRIDE { |
| 1191 | EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for a " |
| 1192 | "receiver RTCP REMB packet to be " |
| 1193 | "sent."; |
| 1194 | } |
| 1195 | } test; |
| 1196 | |
| 1197 | RunBaseTest(&test); |
| 1198 | } |
| 1199 | |
stefan@webrtc.org | 0bae1fa | 2014-11-05 14:05:29 +0000 | [diff] [blame] | 1200 | TEST_F(EndToEndTest, VerifyBandwidthStats) { |
| 1201 | class RtcpObserver : public test::EndToEndTest, public PacketReceiver { |
| 1202 | public: |
| 1203 | RtcpObserver() |
| 1204 | : EndToEndTest(kDefaultTimeoutMs), |
| 1205 | sender_call_(NULL), |
| 1206 | receiver_call_(NULL), |
| 1207 | has_seen_pacer_delay_(false) {} |
| 1208 | |
| 1209 | virtual DeliveryStatus DeliverPacket(const uint8_t* packet, |
| 1210 | size_t length) OVERRIDE { |
| 1211 | Call::Stats sender_stats = sender_call_->GetStats(); |
| 1212 | Call::Stats receiver_stats = receiver_call_->GetStats(); |
| 1213 | if (!has_seen_pacer_delay_) |
| 1214 | has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0; |
| 1215 | if (sender_stats.send_bandwidth_bps > 0 && |
| 1216 | receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_) |
| 1217 | observation_complete_->Set(); |
| 1218 | return receiver_call_->Receiver()->DeliverPacket(packet, length); |
| 1219 | } |
| 1220 | |
| 1221 | virtual void OnCallsCreated(Call* sender_call, |
| 1222 | Call* receiver_call) OVERRIDE { |
| 1223 | sender_call_ = sender_call; |
| 1224 | receiver_call_ = receiver_call; |
| 1225 | } |
| 1226 | |
| 1227 | virtual void PerformTest() OVERRIDE { |
| 1228 | EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for " |
| 1229 | "non-zero bandwidth stats."; |
| 1230 | } |
| 1231 | |
| 1232 | virtual void SetReceivers( |
| 1233 | PacketReceiver* send_transport_receiver, |
| 1234 | PacketReceiver* receive_transport_receiver) OVERRIDE { |
| 1235 | test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver); |
| 1236 | } |
| 1237 | |
| 1238 | private: |
| 1239 | Call* sender_call_; |
| 1240 | Call* receiver_call_; |
| 1241 | bool has_seen_pacer_delay_; |
| 1242 | } test; |
| 1243 | |
| 1244 | RunBaseTest(&test); |
| 1245 | } |
| 1246 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1247 | void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) { |
| 1248 | static const int kNumRtcpReportPacketsToObserve = 5; |
| 1249 | class RtcpXrObserver : public test::EndToEndTest { |
| 1250 | public: |
| 1251 | explicit RtcpXrObserver(bool enable_rrtr) |
| 1252 | : EndToEndTest(kDefaultTimeoutMs), |
| 1253 | enable_rrtr_(enable_rrtr), |
| 1254 | sent_rtcp_sr_(0), |
| 1255 | sent_rtcp_rr_(0), |
| 1256 | sent_rtcp_rrtr_(0), |
| 1257 | sent_rtcp_dlrr_(0) {} |
| 1258 | |
| 1259 | private: |
| 1260 | // Receive stream should send RR packets (and RRTR packets if enabled). |
| 1261 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 1262 | size_t length) OVERRIDE { |
| 1263 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 1264 | EXPECT_TRUE(parser.IsValid()); |
| 1265 | |
| 1266 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 1267 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 1268 | if (packet_type == RTCPUtility::kRtcpRrCode) { |
| 1269 | ++sent_rtcp_rr_; |
| 1270 | } else if (packet_type == |
| 1271 | RTCPUtility::kRtcpXrReceiverReferenceTimeCode) { |
| 1272 | ++sent_rtcp_rrtr_; |
| 1273 | } |
| 1274 | EXPECT_NE(packet_type, RTCPUtility::kRtcpSrCode); |
| 1275 | EXPECT_NE(packet_type, RTCPUtility::kRtcpXrDlrrReportBlockItemCode); |
| 1276 | packet_type = parser.Iterate(); |
| 1277 | } |
| 1278 | return SEND_PACKET; |
| 1279 | } |
| 1280 | // Send stream should send SR packets (and DLRR packets if enabled). |
| 1281 | virtual Action OnSendRtcp(const uint8_t* packet, size_t length) { |
| 1282 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 1283 | EXPECT_TRUE(parser.IsValid()); |
| 1284 | |
| 1285 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 1286 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 1287 | if (packet_type == RTCPUtility::kRtcpSrCode) { |
| 1288 | ++sent_rtcp_sr_; |
| 1289 | } else if (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) { |
| 1290 | ++sent_rtcp_dlrr_; |
| 1291 | } |
| 1292 | EXPECT_NE(packet_type, RTCPUtility::kRtcpXrReceiverReferenceTimeCode); |
| 1293 | packet_type = parser.Iterate(); |
| 1294 | } |
| 1295 | if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve && |
| 1296 | sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) { |
| 1297 | if (enable_rrtr_) { |
| 1298 | EXPECT_GT(sent_rtcp_rrtr_, 0); |
| 1299 | EXPECT_GT(sent_rtcp_dlrr_, 0); |
| 1300 | } else { |
| 1301 | EXPECT_EQ(0, sent_rtcp_rrtr_); |
| 1302 | EXPECT_EQ(0, sent_rtcp_dlrr_); |
| 1303 | } |
| 1304 | observation_complete_->Set(); |
| 1305 | } |
| 1306 | return SEND_PACKET; |
| 1307 | } |
| 1308 | |
| 1309 | virtual void ModifyConfigs( |
| 1310 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1311 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1312 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1313 | (*receive_configs)[0].rtp.rtcp_mode = newapi::kRtcpReducedSize; |
| 1314 | (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = |
| 1315 | enable_rrtr_; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1316 | } |
| 1317 | |
| 1318 | virtual void PerformTest() OVERRIDE { |
| 1319 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1320 | << "Timed out while waiting for RTCP SR/RR packets to be sent."; |
| 1321 | } |
| 1322 | |
| 1323 | bool enable_rrtr_; |
| 1324 | int sent_rtcp_sr_; |
| 1325 | int sent_rtcp_rr_; |
| 1326 | int sent_rtcp_rrtr_; |
| 1327 | int sent_rtcp_dlrr_; |
| 1328 | } test(enable_rrtr); |
| 1329 | |
| 1330 | RunBaseTest(&test); |
| 1331 | } |
| 1332 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1333 | void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs, |
| 1334 | bool send_single_ssrc_first) { |
| 1335 | class SendsSetSsrcs : public test::EndToEndTest { |
| 1336 | public: |
| 1337 | SendsSetSsrcs(const uint32_t* ssrcs, |
| 1338 | size_t num_ssrcs, |
| 1339 | bool send_single_ssrc_first) |
| 1340 | : EndToEndTest(kDefaultTimeoutMs), |
| 1341 | num_ssrcs_(num_ssrcs), |
| 1342 | send_single_ssrc_first_(send_single_ssrc_first), |
| 1343 | ssrcs_to_observe_(num_ssrcs), |
| 1344 | expect_single_ssrc_(send_single_ssrc_first) { |
| 1345 | for (size_t i = 0; i < num_ssrcs; ++i) |
| 1346 | valid_ssrcs_[ssrcs[i]] = true; |
| 1347 | } |
| 1348 | |
| 1349 | private: |
| 1350 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1351 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 1352 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1353 | |
| 1354 | EXPECT_TRUE(valid_ssrcs_[header.ssrc]) |
| 1355 | << "Received unknown SSRC: " << header.ssrc; |
| 1356 | |
| 1357 | if (!valid_ssrcs_[header.ssrc]) |
| 1358 | observation_complete_->Set(); |
| 1359 | |
| 1360 | if (!is_observed_[header.ssrc]) { |
| 1361 | is_observed_[header.ssrc] = true; |
| 1362 | --ssrcs_to_observe_; |
| 1363 | if (expect_single_ssrc_) { |
| 1364 | expect_single_ssrc_ = false; |
| 1365 | observation_complete_->Set(); |
| 1366 | } |
| 1367 | } |
| 1368 | |
| 1369 | if (ssrcs_to_observe_ == 0) |
| 1370 | observation_complete_->Set(); |
| 1371 | |
| 1372 | return SEND_PACKET; |
| 1373 | } |
| 1374 | |
| 1375 | virtual size_t GetNumStreams() const OVERRIDE { return num_ssrcs_; } |
| 1376 | |
| 1377 | virtual void ModifyConfigs( |
| 1378 | VideoSendStream::Config* send_config, |
| 1379 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1380 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1381 | if (num_ssrcs_ > 1) { |
| 1382 | // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1383 | for (size_t i = 0; i < encoder_config->streams.size(); ++i) { |
| 1384 | encoder_config->streams[i].min_bitrate_bps = 10000; |
| 1385 | encoder_config->streams[i].target_bitrate_bps = 15000; |
| 1386 | encoder_config->streams[i].max_bitrate_bps = 20000; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1387 | } |
| 1388 | } |
| 1389 | |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1390 | encoder_config_all_streams_ = *encoder_config; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1391 | if (send_single_ssrc_first_) |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1392 | encoder_config->streams.resize(1); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1393 | } |
| 1394 | |
| 1395 | virtual void OnStreamsCreated( |
| 1396 | VideoSendStream* send_stream, |
| 1397 | const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { |
| 1398 | send_stream_ = send_stream; |
| 1399 | } |
| 1400 | |
| 1401 | virtual void PerformTest() OVERRIDE { |
| 1402 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1403 | << "Timed out while waiting for " |
| 1404 | << (send_single_ssrc_first_ ? "first SSRC." : "SSRCs."); |
| 1405 | |
| 1406 | if (send_single_ssrc_first_) { |
| 1407 | // Set full simulcast and continue with the rest of the SSRCs. |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1408 | send_stream_->ReconfigureVideoEncoder(encoder_config_all_streams_); |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1409 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1410 | << "Timed out while waiting on additional SSRCs."; |
| 1411 | } |
| 1412 | } |
| 1413 | |
| 1414 | private: |
| 1415 | std::map<uint32_t, bool> valid_ssrcs_; |
| 1416 | std::map<uint32_t, bool> is_observed_; |
| 1417 | |
| 1418 | const size_t num_ssrcs_; |
| 1419 | const bool send_single_ssrc_first_; |
| 1420 | |
| 1421 | size_t ssrcs_to_observe_; |
| 1422 | bool expect_single_ssrc_; |
| 1423 | |
| 1424 | VideoSendStream* send_stream_; |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1425 | VideoEncoderConfig encoder_config_all_streams_; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1426 | } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first); |
| 1427 | |
| 1428 | RunBaseTest(&test); |
| 1429 | } |
| 1430 | |
pbos@webrtc.org | 67c2247 | 2014-11-14 17:42:51 +0000 | [diff] [blame] | 1431 | // TODO(pbos): Reenable test, it exposes an assert in the jitter buffer. |
| 1432 | // https://code.google.com/p/webrtc/issues/detail?id=4014 |
| 1433 | TEST_F(EndToEndTest, DISABLED_GetStats) { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1434 | class StatsObserver : public test::EndToEndTest, public I420FrameCallback { |
| 1435 | public: |
| 1436 | StatsObserver() |
| 1437 | : EndToEndTest(kLongTimeoutMs), |
| 1438 | receive_stream_(NULL), |
| 1439 | send_stream_(NULL), |
| 1440 | expected_receive_ssrc_(), |
| 1441 | expected_send_ssrcs_(), |
| 1442 | check_stats_event_(EventWrapper::Create()) {} |
| 1443 | |
| 1444 | private: |
| 1445 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1446 | check_stats_event_->Set(); |
| 1447 | return SEND_PACKET; |
| 1448 | } |
| 1449 | |
| 1450 | virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1451 | check_stats_event_->Set(); |
| 1452 | return SEND_PACKET; |
| 1453 | } |
| 1454 | |
| 1455 | virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1456 | check_stats_event_->Set(); |
| 1457 | return SEND_PACKET; |
| 1458 | } |
| 1459 | |
| 1460 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 1461 | size_t length) OVERRIDE { |
| 1462 | check_stats_event_->Set(); |
| 1463 | return SEND_PACKET; |
| 1464 | } |
| 1465 | |
| 1466 | virtual void FrameCallback(I420VideoFrame* video_frame) OVERRIDE { |
| 1467 | // Ensure that we have at least 5ms send side delay. |
| 1468 | int64_t render_time = video_frame->render_time_ms(); |
| 1469 | if (render_time > 0) |
| 1470 | video_frame->set_render_time_ms(render_time - 5); |
| 1471 | } |
| 1472 | |
| 1473 | bool CheckReceiveStats() { |
| 1474 | assert(receive_stream_ != NULL); |
| 1475 | VideoReceiveStream::Stats stats = receive_stream_->GetStats(); |
| 1476 | EXPECT_EQ(expected_receive_ssrc_, stats.ssrc); |
| 1477 | |
| 1478 | // Make sure all fields have been populated. |
| 1479 | |
| 1480 | receive_stats_filled_["IncomingRate"] |= |
stefan@webrtc.org | 0bae1fa | 2014-11-05 14:05:29 +0000 | [diff] [blame] | 1481 | stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1482 | |
| 1483 | receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0; |
| 1484 | |
| 1485 | receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0; |
| 1486 | |
| 1487 | receive_stats_filled_["StatisticsUpdated"] |= |
| 1488 | stats.rtcp_stats.cumulative_lost != 0 || |
| 1489 | stats.rtcp_stats.extended_max_sequence_number != 0 || |
| 1490 | stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0; |
| 1491 | |
| 1492 | receive_stats_filled_["DataCountersUpdated"] |= |
| 1493 | stats.rtp_stats.bytes != 0 || stats.rtp_stats.fec_packets != 0 || |
| 1494 | stats.rtp_stats.header_bytes != 0 || stats.rtp_stats.packets != 0 || |
| 1495 | stats.rtp_stats.padding_bytes != 0 || |
| 1496 | stats.rtp_stats.retransmitted_packets != 0; |
| 1497 | |
| 1498 | receive_stats_filled_["CodecStats"] |= |
| 1499 | stats.avg_delay_ms != 0 || stats.discarded_packets != 0 || |
| 1500 | stats.key_frames != 0 || stats.delta_frames != 0; |
| 1501 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1502 | return AllStatsFilled(receive_stats_filled_); |
| 1503 | } |
| 1504 | |
| 1505 | bool CheckSendStats() { |
| 1506 | assert(send_stream_ != NULL); |
| 1507 | VideoSendStream::Stats stats = send_stream_->GetStats(); |
| 1508 | |
| 1509 | send_stats_filled_["NumStreams"] |= |
| 1510 | stats.substreams.size() == expected_send_ssrcs_.size(); |
| 1511 | |
stefan@webrtc.org | 0bae1fa | 2014-11-05 14:05:29 +0000 | [diff] [blame] | 1512 | for (std::map<uint32_t, SsrcStats>::const_iterator it = |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1513 | stats.substreams.begin(); |
| 1514 | it != stats.substreams.end(); |
| 1515 | ++it) { |
| 1516 | EXPECT_TRUE(expected_send_ssrcs_.find(it->first) != |
| 1517 | expected_send_ssrcs_.end()); |
| 1518 | |
| 1519 | send_stats_filled_[CompoundKey("IncomingRate", it->first)] |= |
| 1520 | stats.input_frame_rate != 0; |
| 1521 | |
stefan@webrtc.org | 0bae1fa | 2014-11-05 14:05:29 +0000 | [diff] [blame] | 1522 | const SsrcStats& stream_stats = it->second; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1523 | |
| 1524 | send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |= |
| 1525 | stream_stats.rtcp_stats.cumulative_lost != 0 || |
| 1526 | stream_stats.rtcp_stats.extended_max_sequence_number != 0 || |
| 1527 | stream_stats.rtcp_stats.fraction_lost != 0; |
| 1528 | |
| 1529 | send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |= |
| 1530 | stream_stats.rtp_stats.fec_packets != 0 || |
| 1531 | stream_stats.rtp_stats.padding_bytes != 0 || |
| 1532 | stream_stats.rtp_stats.retransmitted_packets != 0 || |
| 1533 | stream_stats.rtp_stats.packets != 0; |
| 1534 | |
| 1535 | send_stats_filled_[CompoundKey("BitrateStatisticsObserver", |
| 1536 | it->first)] |= |
stefan@webrtc.org | 0bae1fa | 2014-11-05 14:05:29 +0000 | [diff] [blame] | 1537 | stream_stats.total_bitrate_bps != 0; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1538 | |
| 1539 | send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |= |
| 1540 | stream_stats.delta_frames != 0 || stream_stats.key_frames != 0; |
| 1541 | |
| 1542 | send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |= |
| 1543 | stats.encode_frame_rate != 0; |
stefan@webrtc.org | 168f23f | 2014-07-11 13:44:02 +0000 | [diff] [blame] | 1544 | |
| 1545 | send_stats_filled_[CompoundKey("Delay", it->first)] |= |
| 1546 | 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] | 1547 | } |
| 1548 | |
| 1549 | return AllStatsFilled(send_stats_filled_); |
| 1550 | } |
| 1551 | |
| 1552 | std::string CompoundKey(const char* name, uint32_t ssrc) { |
| 1553 | std::ostringstream oss; |
| 1554 | oss << name << "_" << ssrc; |
| 1555 | return oss.str(); |
| 1556 | } |
| 1557 | |
| 1558 | bool AllStatsFilled(const std::map<std::string, bool>& stats_map) { |
| 1559 | for (std::map<std::string, bool>::const_iterator it = stats_map.begin(); |
| 1560 | it != stats_map.end(); |
| 1561 | ++it) { |
| 1562 | if (!it->second) |
| 1563 | return false; |
| 1564 | } |
| 1565 | return true; |
| 1566 | } |
| 1567 | |
| 1568 | virtual void ModifyConfigs( |
| 1569 | VideoSendStream::Config* send_config, |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1570 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1571 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1572 | send_config->pre_encode_callback = this; // Used to inject delay. |
| 1573 | send_config->rtp.c_name = "SomeCName"; |
| 1574 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1575 | expected_receive_ssrc_ = (*receive_configs)[0].rtp.local_ssrc; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1576 | const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs; |
| 1577 | for (size_t i = 0; i < ssrcs.size(); ++i) |
| 1578 | expected_send_ssrcs_.insert(ssrcs[i]); |
| 1579 | |
| 1580 | expected_cname_ = send_config->rtp.c_name; |
| 1581 | } |
| 1582 | |
pbos@webrtc.org | ece3890 | 2014-11-14 11:52:04 +0000 | [diff] [blame] | 1583 | virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; } |
| 1584 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1585 | virtual void OnStreamsCreated( |
| 1586 | VideoSendStream* send_stream, |
| 1587 | const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1588 | send_stream_ = send_stream; |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1589 | receive_stream_ = receive_streams[0]; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1590 | } |
| 1591 | |
| 1592 | virtual void PerformTest() OVERRIDE { |
| 1593 | Clock* clock = Clock::GetRealTimeClock(); |
| 1594 | int64_t now = clock->TimeInMilliseconds(); |
| 1595 | int64_t stop_time = now + test::CallTest::kLongTimeoutMs; |
| 1596 | bool receive_ok = false; |
| 1597 | bool send_ok = false; |
| 1598 | |
| 1599 | while (now < stop_time) { |
| 1600 | if (!receive_ok) |
| 1601 | receive_ok = CheckReceiveStats(); |
| 1602 | if (!send_ok) |
| 1603 | send_ok = CheckSendStats(); |
| 1604 | |
| 1605 | if (receive_ok && send_ok) |
| 1606 | return; |
| 1607 | |
| 1608 | int64_t time_until_timout_ = stop_time - now; |
| 1609 | if (time_until_timout_ > 0) |
| 1610 | check_stats_event_->Wait(time_until_timout_); |
| 1611 | now = clock->TimeInMilliseconds(); |
| 1612 | } |
| 1613 | |
| 1614 | ADD_FAILURE() << "Timed out waiting for filled stats."; |
| 1615 | for (std::map<std::string, bool>::const_iterator it = |
| 1616 | receive_stats_filled_.begin(); |
| 1617 | it != receive_stats_filled_.end(); |
| 1618 | ++it) { |
| 1619 | if (!it->second) { |
| 1620 | ADD_FAILURE() << "Missing receive stats: " << it->first; |
| 1621 | } |
| 1622 | } |
| 1623 | |
| 1624 | for (std::map<std::string, bool>::const_iterator it = |
| 1625 | send_stats_filled_.begin(); |
| 1626 | it != send_stats_filled_.end(); |
| 1627 | ++it) { |
| 1628 | if (!it->second) { |
| 1629 | ADD_FAILURE() << "Missing send stats: " << it->first; |
| 1630 | } |
| 1631 | } |
| 1632 | } |
| 1633 | |
| 1634 | VideoReceiveStream* receive_stream_; |
| 1635 | std::map<std::string, bool> receive_stats_filled_; |
| 1636 | |
| 1637 | VideoSendStream* send_stream_; |
| 1638 | std::map<std::string, bool> send_stats_filled_; |
| 1639 | |
| 1640 | uint32_t expected_receive_ssrc_; |
| 1641 | std::set<uint32_t> expected_send_ssrcs_; |
| 1642 | std::string expected_cname_; |
| 1643 | |
| 1644 | scoped_ptr<EventWrapper> check_stats_event_; |
| 1645 | } test; |
| 1646 | |
| 1647 | RunBaseTest(&test); |
| 1648 | } |
| 1649 | |
| 1650 | TEST_F(EndToEndTest, ReceiverReferenceTimeReportEnabled) { |
| 1651 | TestXrReceiverReferenceTimeReport(true); |
| 1652 | } |
| 1653 | |
| 1654 | TEST_F(EndToEndTest, ReceiverReferenceTimeReportDisabled) { |
| 1655 | TestXrReceiverReferenceTimeReport(false); |
| 1656 | } |
| 1657 | |
| 1658 | TEST_F(EndToEndTest, TestReceivedRtpPacketStats) { |
| 1659 | static const size_t kNumRtpPacketsToSend = 5; |
| 1660 | class ReceivedRtpStatsObserver : public test::EndToEndTest { |
| 1661 | public: |
| 1662 | ReceivedRtpStatsObserver() |
| 1663 | : EndToEndTest(kDefaultTimeoutMs), |
| 1664 | receive_stream_(NULL), |
| 1665 | sent_rtp_(0) {} |
| 1666 | |
| 1667 | private: |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1668 | virtual void OnStreamsCreated( |
| 1669 | VideoSendStream* send_stream, |
| 1670 | const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { |
| 1671 | receive_stream_ = receive_streams[0]; |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 1672 | } |
| 1673 | |
| 1674 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1675 | if (sent_rtp_ >= kNumRtpPacketsToSend) { |
| 1676 | VideoReceiveStream::Stats stats = receive_stream_->GetStats(); |
| 1677 | if (kNumRtpPacketsToSend == stats.rtp_stats.packets) { |
| 1678 | observation_complete_->Set(); |
| 1679 | } |
| 1680 | return DROP_PACKET; |
| 1681 | } |
| 1682 | ++sent_rtp_; |
| 1683 | return SEND_PACKET; |
| 1684 | } |
| 1685 | |
| 1686 | virtual void PerformTest() OVERRIDE { |
| 1687 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1688 | << "Timed out while verifying number of received RTP packets."; |
| 1689 | } |
| 1690 | |
| 1691 | VideoReceiveStream* receive_stream_; |
| 1692 | uint32_t sent_rtp_; |
| 1693 | } test; |
| 1694 | |
| 1695 | RunBaseTest(&test); |
| 1696 | } |
| 1697 | |
pbos@webrtc.org | be9d2a4 | 2014-06-30 13:19:09 +0000 | [diff] [blame] | 1698 | TEST_F(EndToEndTest, SendsSetSsrc) { TestSendsSetSsrcs(1, false); } |
| 1699 | |
| 1700 | TEST_F(EndToEndTest, SendsSetSimulcastSsrcs) { |
| 1701 | TestSendsSetSsrcs(kNumSsrcs, false); |
| 1702 | } |
| 1703 | |
| 1704 | TEST_F(EndToEndTest, CanSwitchToUseAllSsrcs) { |
| 1705 | TestSendsSetSsrcs(kNumSsrcs, true); |
| 1706 | } |
| 1707 | |
mflodman@webrtc.org | f946068 | 2014-07-24 16:41:25 +0000 | [diff] [blame] | 1708 | TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) { |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1709 | class ObserveRedundantPayloads: public test::EndToEndTest { |
| 1710 | public: |
| 1711 | ObserveRedundantPayloads() |
| 1712 | : EndToEndTest(kDefaultTimeoutMs), ssrcs_to_observe_(kNumSsrcs) { |
pbos@webrtc.org | dde16f1 | 2014-08-05 23:35:43 +0000 | [diff] [blame] | 1713 | for (size_t i = 0; i < kNumSsrcs; ++i) { |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1714 | registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true; |
| 1715 | } |
| 1716 | } |
| 1717 | |
| 1718 | private: |
| 1719 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1720 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 1721 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1722 | |
| 1723 | if (!registered_rtx_ssrc_[header.ssrc]) |
| 1724 | return SEND_PACKET; |
| 1725 | |
pkasting@chromium.org | 4591fbd | 2014-11-20 22:28:14 +0000 | [diff] [blame^] | 1726 | EXPECT_LE(header.headerLength + header.paddingLength, length); |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1727 | const bool packet_is_redundant_payload = |
pkasting@chromium.org | 4591fbd | 2014-11-20 22:28:14 +0000 | [diff] [blame^] | 1728 | header.headerLength + header.paddingLength < length; |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1729 | |
| 1730 | if (!packet_is_redundant_payload) |
| 1731 | return SEND_PACKET; |
| 1732 | |
| 1733 | if (!observed_redundant_retransmission_[header.ssrc]) { |
| 1734 | observed_redundant_retransmission_[header.ssrc] = true; |
| 1735 | if (--ssrcs_to_observe_ == 0) |
| 1736 | observation_complete_->Set(); |
| 1737 | } |
| 1738 | |
| 1739 | return SEND_PACKET; |
| 1740 | } |
| 1741 | |
| 1742 | virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; } |
| 1743 | |
| 1744 | virtual void ModifyConfigs( |
| 1745 | VideoSendStream::Config* send_config, |
| 1746 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1747 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1748 | // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1749 | for (size_t i = 0; i < encoder_config->streams.size(); ++i) { |
| 1750 | encoder_config->streams[i].min_bitrate_bps = 10000; |
| 1751 | encoder_config->streams[i].target_bitrate_bps = 15000; |
| 1752 | encoder_config->streams[i].max_bitrate_bps = 20000; |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1753 | } |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1754 | |
| 1755 | send_config->rtp.rtx.payload_type = kSendRtxPayloadType; |
| 1756 | send_config->rtp.rtx.pad_with_redundant_payloads = true; |
| 1757 | |
| 1758 | for (size_t i = 0; i < kNumSsrcs; ++i) |
| 1759 | send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); |
pbos@webrtc.org | ad3b5a5 | 2014-10-24 09:23:21 +0000 | [diff] [blame] | 1760 | |
| 1761 | // Significantly higher than max bitrates for all video streams -> forcing |
| 1762 | // padding to trigger redundant padding on all RTX SSRCs. |
| 1763 | encoder_config->min_transmit_bitrate_bps = 100000; |
pbos@webrtc.org | 20c1f56 | 2014-07-04 10:58:12 +0000 | [diff] [blame] | 1764 | } |
| 1765 | |
| 1766 | virtual void PerformTest() OVERRIDE { |
| 1767 | EXPECT_EQ(kEventSignaled, Wait()) |
| 1768 | << "Timed out while waiting for redundant payloads on all SSRCs."; |
| 1769 | } |
| 1770 | |
| 1771 | private: |
| 1772 | size_t ssrcs_to_observe_; |
| 1773 | std::map<uint32_t, bool> observed_redundant_retransmission_; |
| 1774 | std::map<uint32_t, bool> registered_rtx_ssrc_; |
| 1775 | } test; |
| 1776 | |
| 1777 | RunBaseTest(&test); |
| 1778 | } |
| 1779 | |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1780 | void EndToEndTest::TestRtpStatePreservation(bool use_rtx) { |
| 1781 | static const uint32_t kMaxSequenceNumberGap = 100; |
| 1782 | static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90; |
| 1783 | class RtpSequenceObserver : public test::RtpRtcpObserver { |
| 1784 | public: |
pbos@webrtc.org | dde16f1 | 2014-08-05 23:35:43 +0000 | [diff] [blame] | 1785 | explicit RtpSequenceObserver(bool use_rtx) |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1786 | : test::RtpRtcpObserver(kDefaultTimeoutMs), |
| 1787 | crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| 1788 | ssrcs_to_observe_(kNumSsrcs) { |
| 1789 | for (size_t i = 0; i < kNumSsrcs; ++i) { |
| 1790 | configured_ssrcs_[kSendSsrcs[i]] = true; |
| 1791 | if (use_rtx) |
| 1792 | configured_ssrcs_[kSendRtxSsrcs[i]] = true; |
| 1793 | } |
| 1794 | } |
| 1795 | |
| 1796 | void ResetExpectedSsrcs(size_t num_expected_ssrcs) { |
| 1797 | CriticalSectionScoped lock(crit_.get()); |
| 1798 | ssrc_observed_.clear(); |
| 1799 | ssrcs_to_observe_ = num_expected_ssrcs; |
| 1800 | } |
| 1801 | |
| 1802 | private: |
| 1803 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1804 | RTPHeader header; |
pbos@webrtc.org | 62bafae | 2014-07-08 12:10:51 +0000 | [diff] [blame] | 1805 | EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1806 | const uint32_t ssrc = header.ssrc; |
| 1807 | const uint16_t sequence_number = header.sequenceNumber; |
| 1808 | const uint32_t timestamp = header.timestamp; |
| 1809 | const bool only_padding = |
pkasting@chromium.org | 4591fbd | 2014-11-20 22:28:14 +0000 | [diff] [blame^] | 1810 | header.headerLength + header.paddingLength == length; |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1811 | |
| 1812 | EXPECT_TRUE(configured_ssrcs_[ssrc]) |
| 1813 | << "Received SSRC that wasn't configured: " << ssrc; |
| 1814 | |
| 1815 | std::map<uint32_t, uint16_t>::iterator it = |
| 1816 | last_observed_sequence_number_.find(header.ssrc); |
| 1817 | if (it == last_observed_sequence_number_.end()) { |
| 1818 | last_observed_sequence_number_[ssrc] = sequence_number; |
| 1819 | last_observed_timestamp_[ssrc] = timestamp; |
| 1820 | } else { |
| 1821 | // Verify sequence numbers are reasonably close. |
| 1822 | uint32_t extended_sequence_number = sequence_number; |
| 1823 | // Check for roll-over. |
| 1824 | if (sequence_number < last_observed_sequence_number_[ssrc]) |
| 1825 | extended_sequence_number += 0xFFFFu + 1; |
| 1826 | EXPECT_LE( |
| 1827 | extended_sequence_number - last_observed_sequence_number_[ssrc], |
| 1828 | kMaxSequenceNumberGap) |
| 1829 | << "Gap in sequence numbers (" |
| 1830 | << last_observed_sequence_number_[ssrc] << " -> " << sequence_number |
| 1831 | << ") too large for SSRC: " << ssrc << "."; |
| 1832 | last_observed_sequence_number_[ssrc] = sequence_number; |
| 1833 | |
| 1834 | // TODO(pbos): Remove this check if we ever have monotonically |
| 1835 | // increasing timestamps. Right now padding packets add a delta which |
| 1836 | // can cause reordering between padding packets and regular packets, |
| 1837 | // hence we drop padding-only packets to not flake. |
| 1838 | if (only_padding) { |
| 1839 | // Verify that timestamps are reasonably close. |
| 1840 | uint64_t extended_timestamp = timestamp; |
| 1841 | // Check for roll-over. |
| 1842 | if (timestamp < last_observed_timestamp_[ssrc]) |
| 1843 | extended_timestamp += static_cast<uint64_t>(0xFFFFFFFFu) + 1; |
| 1844 | EXPECT_LE(extended_timestamp - last_observed_timestamp_[ssrc], |
| 1845 | kMaxTimestampGap) |
| 1846 | << "Gap in timestamps (" << last_observed_timestamp_[ssrc] |
| 1847 | << " -> " << timestamp << ") too large for SSRC: " << ssrc << "."; |
| 1848 | } |
| 1849 | last_observed_timestamp_[ssrc] = timestamp; |
| 1850 | } |
| 1851 | |
| 1852 | CriticalSectionScoped lock(crit_.get()); |
| 1853 | // Wait for media packets on all ssrcs. |
| 1854 | if (!ssrc_observed_[ssrc] && !only_padding) { |
| 1855 | ssrc_observed_[ssrc] = true; |
| 1856 | if (--ssrcs_to_observe_ == 0) |
| 1857 | observation_complete_->Set(); |
| 1858 | } |
| 1859 | |
| 1860 | return SEND_PACKET; |
| 1861 | } |
| 1862 | |
| 1863 | std::map<uint32_t, uint16_t> last_observed_sequence_number_; |
| 1864 | std::map<uint32_t, uint32_t> last_observed_timestamp_; |
| 1865 | std::map<uint32_t, bool> configured_ssrcs_; |
| 1866 | |
| 1867 | scoped_ptr<CriticalSectionWrapper> crit_; |
| 1868 | size_t ssrcs_to_observe_ GUARDED_BY(crit_); |
| 1869 | std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_); |
| 1870 | } observer(use_rtx); |
| 1871 | |
| 1872 | CreateCalls(Call::Config(observer.SendTransport()), |
| 1873 | Call::Config(observer.ReceiveTransport())); |
| 1874 | observer.SetReceivers(sender_call_->Receiver(), NULL); |
| 1875 | |
| 1876 | CreateSendConfig(kNumSsrcs); |
| 1877 | |
| 1878 | if (use_rtx) { |
| 1879 | for (size_t i = 0; i < kNumSsrcs; ++i) { |
| 1880 | send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); |
| 1881 | } |
| 1882 | send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; |
| 1883 | } |
| 1884 | |
| 1885 | // Lower bitrates so that all streams send initially. |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1886 | for (size_t i = 0; i < encoder_config_.streams.size(); ++i) { |
| 1887 | encoder_config_.streams[i].min_bitrate_bps = 10000; |
| 1888 | encoder_config_.streams[i].target_bitrate_bps = 15000; |
| 1889 | encoder_config_.streams[i].max_bitrate_bps = 20000; |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1890 | } |
| 1891 | |
pbos@webrtc.org | 32452b2 | 2014-10-22 12:15:24 +0000 | [diff] [blame] | 1892 | // Use the same total bitrates when sending a single stream to avoid lowering |
| 1893 | // the bitrate estimate and requiring a subsequent rampup. |
| 1894 | VideoEncoderConfig one_stream = encoder_config_; |
| 1895 | one_stream.streams.resize(1); |
| 1896 | for (size_t i = 1; i < encoder_config_.streams.size(); ++i) { |
| 1897 | one_stream.streams.front().min_bitrate_bps += |
| 1898 | encoder_config_.streams[i].min_bitrate_bps; |
| 1899 | one_stream.streams.front().target_bitrate_bps += |
| 1900 | encoder_config_.streams[i].target_bitrate_bps; |
| 1901 | one_stream.streams.front().max_bitrate_bps += |
| 1902 | encoder_config_.streams[i].max_bitrate_bps; |
| 1903 | } |
| 1904 | |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1905 | CreateMatchingReceiveConfigs(); |
| 1906 | |
| 1907 | CreateStreams(); |
| 1908 | CreateFrameGeneratorCapturer(); |
| 1909 | |
| 1910 | Start(); |
| 1911 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1912 | << "Timed out waiting for all SSRCs to send packets."; |
| 1913 | |
| 1914 | // Test stream resetting more than once to make sure that the state doesn't |
| 1915 | // get set once (this could be due to using std::map::insert for instance). |
| 1916 | for (size_t i = 0; i < 3; ++i) { |
| 1917 | frame_generator_capturer_->Stop(); |
| 1918 | sender_call_->DestroyVideoSendStream(send_stream_); |
| 1919 | |
| 1920 | // Re-create VideoSendStream with only one stream. |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1921 | send_stream_ = |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1922 | sender_call_->CreateVideoSendStream(send_config_, one_stream); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1923 | send_stream_->Start(); |
| 1924 | CreateFrameGeneratorCapturer(); |
| 1925 | frame_generator_capturer_->Start(); |
| 1926 | |
| 1927 | observer.ResetExpectedSsrcs(1); |
| 1928 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1929 | << "Timed out waiting for single RTP packet."; |
| 1930 | |
| 1931 | // Reconfigure back to use all streams. |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1932 | send_stream_->ReconfigureVideoEncoder(encoder_config_); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1933 | observer.ResetExpectedSsrcs(kNumSsrcs); |
| 1934 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1935 | << "Timed out waiting for all SSRCs to send packets."; |
| 1936 | |
| 1937 | // Reconfigure down to one stream. |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1938 | send_stream_->ReconfigureVideoEncoder(one_stream); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1939 | observer.ResetExpectedSsrcs(1); |
| 1940 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1941 | << "Timed out waiting for single RTP packet."; |
| 1942 | |
| 1943 | // Reconfigure back to use all streams. |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 1944 | send_stream_->ReconfigureVideoEncoder(encoder_config_); |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1945 | observer.ResetExpectedSsrcs(kNumSsrcs); |
| 1946 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1947 | << "Timed out waiting for all SSRCs to send packets."; |
| 1948 | } |
| 1949 | |
| 1950 | observer.StopSending(); |
| 1951 | |
| 1952 | Stop(); |
| 1953 | DestroyStreams(); |
| 1954 | } |
| 1955 | |
aluebs@webrtc.org | b623c5c | 2014-08-26 14:22:51 +0000 | [diff] [blame] | 1956 | TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpState) { |
pbos@webrtc.org | 2bb1bda | 2014-07-07 13:06:48 +0000 | [diff] [blame] | 1957 | TestRtpStatePreservation(false); |
| 1958 | } |
| 1959 | |
| 1960 | TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) { |
| 1961 | TestRtpStatePreservation(true); |
| 1962 | } |
| 1963 | |
pbos@webrtc.org | 26c0c41 | 2014-09-03 16:17:12 +0000 | [diff] [blame] | 1964 | TEST_F(EndToEndTest, RespectsNetworkState) { |
| 1965 | // TODO(pbos): Remove accepted downtime packets etc. when signaling network |
| 1966 | // down blocks until no more packets will be sent. |
| 1967 | |
| 1968 | // Pacer will send from its packet list and then send required padding before |
| 1969 | // checking paused_ again. This should be enough for one round of pacing, |
| 1970 | // otherwise increase. |
| 1971 | static const int kNumAcceptedDowntimeRtp = 5; |
| 1972 | // A single RTCP may be in the pipeline. |
| 1973 | static const int kNumAcceptedDowntimeRtcp = 1; |
| 1974 | class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder { |
| 1975 | public: |
| 1976 | NetworkStateTest() |
| 1977 | : EndToEndTest(kDefaultTimeoutMs), |
| 1978 | FakeEncoder(Clock::GetRealTimeClock()), |
| 1979 | test_crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| 1980 | encoded_frames_(EventWrapper::Create()), |
| 1981 | sender_packets_(EventWrapper::Create()), |
| 1982 | receiver_packets_(EventWrapper::Create()), |
| 1983 | sender_state_(Call::kNetworkUp), |
| 1984 | down_sender_rtp_(0), |
| 1985 | down_sender_rtcp_(0), |
| 1986 | receiver_state_(Call::kNetworkUp), |
| 1987 | down_receiver_rtcp_(0), |
| 1988 | down_frames_(0) {} |
| 1989 | |
| 1990 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 1991 | CriticalSectionScoped lock(test_crit_.get()); |
| 1992 | if (sender_state_ == Call::kNetworkDown) { |
| 1993 | ++down_sender_rtp_; |
| 1994 | EXPECT_LE(down_sender_rtp_, kNumAcceptedDowntimeRtp) |
| 1995 | << "RTP sent during sender-side downtime."; |
| 1996 | if (down_sender_rtp_> kNumAcceptedDowntimeRtp) |
| 1997 | sender_packets_->Set(); |
| 1998 | } else { |
| 1999 | sender_packets_->Set(); |
| 2000 | } |
| 2001 | return SEND_PACKET; |
| 2002 | } |
| 2003 | |
| 2004 | virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 2005 | CriticalSectionScoped lock(test_crit_.get()); |
| 2006 | if (sender_state_ == Call::kNetworkDown) { |
| 2007 | ++down_sender_rtcp_; |
| 2008 | EXPECT_LE(down_sender_rtcp_, kNumAcceptedDowntimeRtcp) |
| 2009 | << "RTCP sent during sender-side downtime."; |
| 2010 | if (down_sender_rtcp_ > kNumAcceptedDowntimeRtcp) |
| 2011 | sender_packets_->Set(); |
| 2012 | } else { |
| 2013 | sender_packets_->Set(); |
| 2014 | } |
| 2015 | return SEND_PACKET; |
| 2016 | } |
| 2017 | |
| 2018 | virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 2019 | ADD_FAILURE() << "Unexpected receiver RTP, should not be sending."; |
| 2020 | return SEND_PACKET; |
| 2021 | } |
| 2022 | |
| 2023 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 2024 | size_t length) OVERRIDE { |
| 2025 | CriticalSectionScoped lock(test_crit_.get()); |
| 2026 | if (receiver_state_ == Call::kNetworkDown) { |
| 2027 | ++down_receiver_rtcp_; |
| 2028 | EXPECT_LE(down_receiver_rtcp_, kNumAcceptedDowntimeRtcp) |
| 2029 | << "RTCP sent during receiver-side downtime."; |
| 2030 | if (down_receiver_rtcp_ > kNumAcceptedDowntimeRtcp) |
| 2031 | receiver_packets_->Set(); |
| 2032 | } else { |
| 2033 | receiver_packets_->Set(); |
| 2034 | } |
| 2035 | return SEND_PACKET; |
| 2036 | } |
| 2037 | |
| 2038 | virtual void OnCallsCreated(Call* sender_call, |
| 2039 | Call* receiver_call) OVERRIDE { |
| 2040 | sender_call_ = sender_call; |
| 2041 | receiver_call_ = receiver_call; |
| 2042 | } |
| 2043 | |
| 2044 | virtual void ModifyConfigs( |
| 2045 | VideoSendStream::Config* send_config, |
| 2046 | std::vector<VideoReceiveStream::Config>* receive_configs, |
pbos@webrtc.org | bbe0a85 | 2014-09-19 12:30:25 +0000 | [diff] [blame] | 2047 | VideoEncoderConfig* encoder_config) OVERRIDE { |
pbos@webrtc.org | 26c0c41 | 2014-09-03 16:17:12 +0000 | [diff] [blame] | 2048 | send_config->encoder_settings.encoder = this; |
| 2049 | } |
| 2050 | |
| 2051 | virtual void PerformTest() OVERRIDE { |
| 2052 | EXPECT_EQ(kEventSignaled, encoded_frames_->Wait(kDefaultTimeoutMs)) |
| 2053 | << "No frames received by the encoder."; |
| 2054 | EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs)) |
| 2055 | << "Timed out waiting for send-side packets."; |
| 2056 | EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs)) |
| 2057 | << "Timed out waiting for receiver-side packets."; |
| 2058 | |
| 2059 | // Sender-side network down. |
| 2060 | sender_call_->SignalNetworkState(Call::kNetworkDown); |
| 2061 | { |
| 2062 | CriticalSectionScoped lock(test_crit_.get()); |
| 2063 | sender_packets_->Reset(); // Earlier packets should not count. |
| 2064 | sender_state_ = Call::kNetworkDown; |
| 2065 | } |
| 2066 | EXPECT_EQ(kEventTimeout, sender_packets_->Wait(kSilenceTimeoutMs)) |
| 2067 | << "Packets sent during sender-network downtime."; |
| 2068 | EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs)) |
| 2069 | << "Timed out waiting for receiver-side packets."; |
| 2070 | // Receiver-side network down. |
| 2071 | receiver_call_->SignalNetworkState(Call::kNetworkDown); |
| 2072 | { |
| 2073 | CriticalSectionScoped lock(test_crit_.get()); |
| 2074 | receiver_packets_->Reset(); // Earlier packets should not count. |
| 2075 | receiver_state_ = Call::kNetworkDown; |
| 2076 | } |
| 2077 | EXPECT_EQ(kEventTimeout, receiver_packets_->Wait(kSilenceTimeoutMs)) |
| 2078 | << "Packets sent during receiver-network downtime."; |
| 2079 | |
| 2080 | // Network back up again for both. |
| 2081 | { |
| 2082 | CriticalSectionScoped lock(test_crit_.get()); |
| 2083 | sender_packets_->Reset(); // Earlier packets should not count. |
| 2084 | receiver_packets_->Reset(); // Earlier packets should not count. |
| 2085 | sender_state_ = receiver_state_ = Call::kNetworkUp; |
| 2086 | } |
| 2087 | sender_call_->SignalNetworkState(Call::kNetworkUp); |
| 2088 | receiver_call_->SignalNetworkState(Call::kNetworkUp); |
| 2089 | EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs)) |
| 2090 | << "Timed out waiting for send-side packets."; |
| 2091 | EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs)) |
| 2092 | << "Timed out waiting for receiver-side packets."; |
| 2093 | } |
| 2094 | |
| 2095 | virtual int32_t Encode(const I420VideoFrame& input_image, |
| 2096 | const CodecSpecificInfo* codec_specific_info, |
| 2097 | const std::vector<VideoFrameType>* frame_types) |
| 2098 | OVERRIDE { |
| 2099 | { |
| 2100 | CriticalSectionScoped lock(test_crit_.get()); |
| 2101 | if (sender_state_ == Call::kNetworkDown) { |
| 2102 | ++down_frames_; |
| 2103 | EXPECT_LE(down_frames_, 1) |
| 2104 | << "Encoding more than one frame while network is down."; |
| 2105 | if (down_frames_ > 1) |
| 2106 | encoded_frames_->Set(); |
| 2107 | } else { |
| 2108 | encoded_frames_->Set(); |
| 2109 | } |
| 2110 | } |
| 2111 | return test::FakeEncoder::Encode( |
| 2112 | input_image, codec_specific_info, frame_types); |
| 2113 | } |
| 2114 | |
| 2115 | private: |
| 2116 | const scoped_ptr<CriticalSectionWrapper> test_crit_; |
| 2117 | scoped_ptr<EventWrapper> encoded_frames_; |
| 2118 | scoped_ptr<EventWrapper> sender_packets_; |
| 2119 | scoped_ptr<EventWrapper> receiver_packets_; |
| 2120 | Call* sender_call_; |
| 2121 | Call* receiver_call_; |
| 2122 | Call::NetworkState sender_state_ GUARDED_BY(test_crit_); |
| 2123 | int down_sender_rtp_ GUARDED_BY(test_crit_); |
| 2124 | int down_sender_rtcp_ GUARDED_BY(test_crit_); |
| 2125 | Call::NetworkState receiver_state_ GUARDED_BY(test_crit_); |
| 2126 | int down_receiver_rtcp_ GUARDED_BY(test_crit_); |
| 2127 | int down_frames_ GUARDED_BY(test_crit_); |
| 2128 | } test; |
| 2129 | |
| 2130 | RunBaseTest(&test); |
| 2131 | } |
| 2132 | |
| 2133 | TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) { |
| 2134 | class UnusedEncoder : public test::FakeEncoder { |
| 2135 | public: |
| 2136 | UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {} |
| 2137 | virtual int32_t Encode(const I420VideoFrame& input_image, |
| 2138 | const CodecSpecificInfo* codec_specific_info, |
| 2139 | const std::vector<VideoFrameType>* frame_types) |
| 2140 | OVERRIDE { |
| 2141 | ADD_FAILURE() << "Unexpected frame encode."; |
| 2142 | return test::FakeEncoder::Encode( |
| 2143 | input_image, codec_specific_info, frame_types); |
| 2144 | } |
| 2145 | }; |
| 2146 | |
| 2147 | UnusedTransport transport; |
| 2148 | CreateSenderCall(Call::Config(&transport)); |
| 2149 | sender_call_->SignalNetworkState(Call::kNetworkDown); |
| 2150 | |
| 2151 | CreateSendConfig(1); |
| 2152 | UnusedEncoder unused_encoder; |
| 2153 | send_config_.encoder_settings.encoder = &unused_encoder; |
| 2154 | CreateStreams(); |
| 2155 | CreateFrameGeneratorCapturer(); |
| 2156 | |
| 2157 | Start(); |
| 2158 | SleepMs(kSilenceTimeoutMs); |
| 2159 | Stop(); |
| 2160 | |
| 2161 | DestroyStreams(); |
| 2162 | } |
| 2163 | |
| 2164 | TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) { |
| 2165 | test::DirectTransport sender_transport; |
| 2166 | CreateSenderCall(Call::Config(&sender_transport)); |
| 2167 | UnusedTransport transport; |
| 2168 | CreateReceiverCall(Call::Config(&transport)); |
| 2169 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 2170 | |
| 2171 | receiver_call_->SignalNetworkState(Call::kNetworkDown); |
| 2172 | |
| 2173 | CreateSendConfig(1); |
| 2174 | CreateMatchingReceiveConfigs(); |
| 2175 | CreateStreams(); |
| 2176 | CreateFrameGeneratorCapturer(); |
| 2177 | |
| 2178 | Start(); |
| 2179 | SleepMs(kSilenceTimeoutMs); |
| 2180 | Stop(); |
| 2181 | |
| 2182 | sender_transport.StopSending(); |
| 2183 | |
| 2184 | DestroyStreams(); |
| 2185 | } |
pbos@webrtc.org | 09cc686 | 2014-11-04 13:48:15 +0000 | [diff] [blame] | 2186 | |
| 2187 | // TODO(pbos): Remove this regression test when VideoEngine is no longer used as |
| 2188 | // a backend. This is to test that we hand channels back properly. |
| 2189 | TEST_F(EndToEndTest, CanCreateAndDestroyManyVideoStreams) { |
| 2190 | test::NullTransport transport; |
| 2191 | scoped_ptr<Call> call(Call::Create(Call::Config(&transport))); |
| 2192 | test::FakeDecoder fake_decoder; |
| 2193 | test::FakeEncoder fake_encoder(Clock::GetRealTimeClock()); |
| 2194 | for (size_t i = 0; i < 100; ++i) { |
| 2195 | VideoSendStream::Config send_config; |
| 2196 | send_config.encoder_settings.encoder = &fake_encoder; |
| 2197 | send_config.encoder_settings.payload_name = "FAKE"; |
| 2198 | send_config.encoder_settings.payload_type = 123; |
| 2199 | |
| 2200 | VideoEncoderConfig encoder_config; |
| 2201 | encoder_config.streams = test::CreateVideoStreams(1); |
| 2202 | send_config.rtp.ssrcs.push_back(1); |
| 2203 | VideoSendStream* send_stream = |
| 2204 | call->CreateVideoSendStream(send_config, encoder_config); |
| 2205 | call->DestroyVideoSendStream(send_stream); |
| 2206 | |
| 2207 | VideoReceiveStream::Config receive_config; |
| 2208 | receive_config.rtp.remote_ssrc = 1; |
| 2209 | receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| 2210 | VideoReceiveStream::Decoder decoder; |
| 2211 | decoder.decoder = &fake_decoder; |
| 2212 | decoder.payload_type = 123; |
| 2213 | decoder.payload_name = "FAKE"; |
| 2214 | receive_config.decoders.push_back(decoder); |
| 2215 | VideoReceiveStream* receive_stream = |
| 2216 | call->CreateVideoReceiveStream(receive_config); |
| 2217 | call->DestroyVideoReceiveStream(receive_stream); |
| 2218 | } |
| 2219 | } |
| 2220 | |
pbos@webrtc.org | 994d0b7 | 2014-06-27 08:47:52 +0000 | [diff] [blame] | 2221 | } // namespace webrtc |