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pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001/*
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.org5b883172014-11-01 06:10:48 +000022#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.org994d0b72014-06-27 08:47:52 +000025#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.orgab071da2014-09-18 08:58:15 +000040#include "webrtc/test/testsupport/gtest_disable.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000041#include "webrtc/test/testsupport/perf_test.h"
42#include "webrtc/video/transport_adapter.h"
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +000043#include "webrtc/video_encoder.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000044
45namespace webrtc {
46
pbos@webrtc.org26c0c412014-09-03 16:17:12 +000047static const unsigned long kSilenceTimeoutMs = 2000;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000048
49class EndToEndTest : public test::CallTest {
50 public:
51 EndToEndTest() {}
52
53 virtual ~EndToEndTest() {
54 EXPECT_EQ(NULL, send_stream_);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +000055 EXPECT_TRUE(receive_streams_.empty());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000056 }
57
58 protected:
pbos@webrtc.org26c0c412014-09-03 16:17:12 +000059 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.org994d0b72014-06-27 08:47:52 +000072 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.orgbe9d2a42014-06-30 13:19:09 +000076 void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000077 void TestRtpStatePreservation(bool use_rtx);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000078};
79
80TEST_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.orgbe9d2a42014-06-30 13:19:09 +000089 receive_streams_[0]->Start();
90 receive_streams_[0]->Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000091
92 DestroyStreams();
93}
94
95TEST_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.orgbe9d2a42014-06-30 13:19:09 +0000104 receive_streams_[0]->Stop();
105 receive_streams_[0]->Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000106
107 DestroyStreams();
108}
109
110TEST_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.orgbe9d2a42014-06-30 13:19:09 +0000159 receive_configs_[0].pre_render_callback = &pre_render_callback;
160 receive_configs_[0].renderer = &renderer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000161
162 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000163 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000164
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.orgbe9d2a42014-06-30 13:19:09 +0000175 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000176
177 sender_transport.StopSending();
178 receiver_transport.StopSending();
179
180 DestroyStreams();
181}
182
183TEST_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.orgbe9d2a42014-06-30 13:19:09 +0000208 receive_configs_[0].renderer = &renderer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000209
210 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000211 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000212
213 scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create(
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000214 encoder_config_.streams[0].width, encoder_config_.streams[0].height));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000215 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.orgbe9d2a42014-06-30 13:19:09 +0000220 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000221
222 sender_transport.StopSending();
223 receiver_transport.StopSending();
224
225 DestroyStreams();
226}
227
marpan@webrtc.org5b883172014-11-01 06:10:48 +0000228// TODO(marpan): Re-enable this test on the next libvpx roll.
229TEST_F(EndToEndTest, DISABLED_SendsAndReceivesVP9) {
230 class VP9Observer : public test::EndToEndTest, public VideoRenderer {
231 public:
232 VP9Observer()
233 : EndToEndTest(2 * kDefaultTimeoutMs),
234 encoder_(VideoEncoder::Create(VideoEncoder::kVp9)),
235 decoder_(VP9Decoder::Create()),
236 frame_counter_(0) {}
237
238 virtual void PerformTest() OVERRIDE {
239 EXPECT_EQ(kEventSignaled, Wait())
240 << "Timed out while waiting for enough frames to be decoded.";
241 }
242
243 virtual void ModifyConfigs(
244 VideoSendStream::Config* send_config,
245 std::vector<VideoReceiveStream::Config>* receive_configs,
246 VideoEncoderConfig* encoder_config) OVERRIDE {
247 send_config->encoder_settings.encoder = encoder_.get();
248 send_config->encoder_settings.payload_name = "VP9";
249 send_config->encoder_settings.payload_type = VCM_VP9_PAYLOAD_TYPE;
250 encoder_config->streams[0].min_bitrate_bps = 50000;
251 encoder_config->streams[0].target_bitrate_bps =
252 encoder_config->streams[0].max_bitrate_bps = 2000000;
253
254 (*receive_configs)[0].renderer = this;
255 (*receive_configs)[0].decoders.resize(1);
256 (*receive_configs)[0].decoders[0].payload_type =
257 send_config->encoder_settings.payload_type;
258 (*receive_configs)[0].decoders[0].payload_name =
259 send_config->encoder_settings.payload_name;
260 (*receive_configs)[0].decoders[0].decoder = decoder_.get();
261 }
262
263 virtual void RenderFrame(const I420VideoFrame& video_frame,
264 int time_to_render_ms) OVERRIDE {
265 const int kRequiredFrames = 500;
266 if (++frame_counter_ == kRequiredFrames)
267 observation_complete_->Set();
268 }
269
270 private:
271 scoped_ptr<webrtc::VideoEncoder> encoder_;
272 scoped_ptr<webrtc::VideoDecoder> decoder_;
273 int frame_counter_;
274 } test;
275
276 RunBaseTest(&test);
277}
278
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000279TEST_F(EndToEndTest, SendsAndReceivesH264) {
280 class H264Observer : public test::EndToEndTest, public VideoRenderer {
281 public:
282 H264Observer()
283 : EndToEndTest(2 * kDefaultTimeoutMs),
284 fake_encoder_(Clock::GetRealTimeClock()),
285 frame_counter_(0) {}
286
287 virtual void PerformTest() OVERRIDE {
288 EXPECT_EQ(kEventSignaled, Wait())
289 << "Timed out while waiting for enough frames to be decoded.";
290 }
291
292 virtual void ModifyConfigs(
293 VideoSendStream::Config* send_config,
294 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000295 VideoEncoderConfig* encoder_config) OVERRIDE {
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000296 send_config->encoder_settings.encoder = &fake_encoder_;
297 send_config->encoder_settings.payload_name = "H264";
298 send_config->encoder_settings.payload_type = kFakeSendPayloadType;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000299 encoder_config->streams[0].min_bitrate_bps = 50000;
300 encoder_config->streams[0].target_bitrate_bps =
301 encoder_config->streams[0].max_bitrate_bps = 2000000;
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000302
303 (*receive_configs)[0].renderer = this;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000304 (*receive_configs)[0].decoders.resize(1);
305 (*receive_configs)[0].decoders[0].payload_type =
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000306 send_config->encoder_settings.payload_type;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +0000307 (*receive_configs)[0].decoders[0].payload_name =
308 send_config->encoder_settings.payload_name;
309 (*receive_configs)[0].decoders[0].decoder = &fake_decoder_;
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000310 }
311
312 virtual void RenderFrame(const I420VideoFrame& video_frame,
313 int time_to_render_ms) OVERRIDE {
314 const int kRequiredFrames = 500;
315 if (++frame_counter_ == kRequiredFrames)
316 observation_complete_->Set();
317 }
318
319 private:
320 test::FakeH264Decoder fake_decoder_;
321 test::FakeH264Encoder fake_encoder_;
322 int frame_counter_;
323 } test;
324
325 RunBaseTest(&test);
326}
327
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000328TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
329 class SyncRtcpObserver : public test::EndToEndTest {
330 public:
331 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
332
333 virtual Action OnReceiveRtcp(const uint8_t* packet,
334 size_t length) OVERRIDE {
335 RTCPUtility::RTCPParserV2 parser(packet, length, true);
336 EXPECT_TRUE(parser.IsValid());
337 uint32_t ssrc = 0;
338 ssrc |= static_cast<uint32_t>(packet[4]) << 24;
339 ssrc |= static_cast<uint32_t>(packet[5]) << 16;
340 ssrc |= static_cast<uint32_t>(packet[6]) << 8;
341 ssrc |= static_cast<uint32_t>(packet[7]) << 0;
342 EXPECT_EQ(kReceiverLocalSsrc, ssrc);
343 observation_complete_->Set();
344
345 return SEND_PACKET;
346 }
347
348 virtual void PerformTest() OVERRIDE {
349 EXPECT_EQ(kEventSignaled, Wait())
350 << "Timed out while waiting for a receiver RTCP packet to be sent.";
351 }
352 } test;
353
354 RunBaseTest(&test);
355}
356
357TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
358 static const int kNumberOfNacksToObserve = 2;
359 static const int kLossBurstSize = 2;
360 static const int kPacketsBetweenLossBursts = 9;
361 class NackObserver : public test::EndToEndTest {
362 public:
363 NackObserver()
364 : EndToEndTest(kLongTimeoutMs),
365 rtp_parser_(RtpHeaderParser::Create()),
366 sent_rtp_packets_(0),
367 packets_left_to_drop_(0),
368 nacks_left_(kNumberOfNacksToObserve) {}
369
370 private:
371 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
372 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000373 EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000374
375 // Never drop retransmitted packets.
376 if (dropped_packets_.find(header.sequenceNumber) !=
377 dropped_packets_.end()) {
378 retransmitted_packets_.insert(header.sequenceNumber);
379 if (nacks_left_ == 0 &&
380 retransmitted_packets_.size() == dropped_packets_.size()) {
381 observation_complete_->Set();
382 }
383 return SEND_PACKET;
384 }
385
386 ++sent_rtp_packets_;
387
388 // Enough NACKs received, stop dropping packets.
389 if (nacks_left_ == 0)
390 return SEND_PACKET;
391
392 // Check if it's time for a new loss burst.
393 if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
394 packets_left_to_drop_ = kLossBurstSize;
395
396 if (packets_left_to_drop_ > 0) {
397 --packets_left_to_drop_;
398 dropped_packets_.insert(header.sequenceNumber);
399 return DROP_PACKET;
400 }
401
402 return SEND_PACKET;
403 }
404
405 virtual Action OnReceiveRtcp(const uint8_t* packet,
406 size_t length) OVERRIDE {
407 RTCPUtility::RTCPParserV2 parser(packet, length, true);
408 EXPECT_TRUE(parser.IsValid());
409
410 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
411 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
412 if (packet_type == RTCPUtility::kRtcpRtpfbNackCode) {
413 --nacks_left_;
414 break;
415 }
416 packet_type = parser.Iterate();
417 }
418 return SEND_PACKET;
419 }
420
421 virtual void ModifyConfigs(
422 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000423 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000424 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000425 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000426 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000427 }
428
429 virtual void PerformTest() OVERRIDE {
430 EXPECT_EQ(kEventSignaled, Wait())
431 << "Timed out waiting for packets to be NACKed, retransmitted and "
432 "rendered.";
433 }
434
435 scoped_ptr<RtpHeaderParser> rtp_parser_;
436 std::set<uint16_t> dropped_packets_;
437 std::set<uint16_t> retransmitted_packets_;
438 uint64_t sent_rtp_packets_;
439 int packets_left_to_drop_;
440 int nacks_left_;
441 } test;
442
443 RunBaseTest(&test);
444}
445
446// TODO(pbos): Flaky, webrtc:3269
447TEST_F(EndToEndTest, DISABLED_CanReceiveFec) {
448 class FecRenderObserver : public test::EndToEndTest, public VideoRenderer {
449 public:
450 FecRenderObserver()
451 : EndToEndTest(kDefaultTimeoutMs),
452 state_(kFirstPacket),
453 protected_sequence_number_(0),
454 protected_frame_timestamp_(0) {}
455
456 private:
457 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE
458 EXCLUSIVE_LOCKS_REQUIRED(crit_) {
459 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000460 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000461
462 EXPECT_EQ(kRedPayloadType, header.payloadType);
463 int encapsulated_payload_type =
464 static_cast<int>(packet[header.headerLength]);
465 if (encapsulated_payload_type != kFakeSendPayloadType)
466 EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
467
468 switch (state_) {
469 case kFirstPacket:
470 state_ = kDropEveryOtherPacketUntilFec;
471 break;
472 case kDropEveryOtherPacketUntilFec:
473 if (encapsulated_payload_type == kUlpfecPayloadType) {
474 state_ = kDropNextMediaPacket;
475 return SEND_PACKET;
476 }
477 if (header.sequenceNumber % 2 == 0)
478 return DROP_PACKET;
479 break;
480 case kDropNextMediaPacket:
481 if (encapsulated_payload_type == kFakeSendPayloadType) {
482 protected_sequence_number_ = header.sequenceNumber;
483 protected_frame_timestamp_ = header.timestamp;
484 state_ = kProtectedPacketDropped;
485 return DROP_PACKET;
486 }
487 break;
488 case kProtectedPacketDropped:
489 EXPECT_NE(header.sequenceNumber, protected_sequence_number_)
490 << "Protected packet retransmitted. Should not happen with FEC.";
491 break;
492 }
493
494 return SEND_PACKET;
495 }
496
497 virtual void RenderFrame(const I420VideoFrame& video_frame,
498 int time_to_render_ms) OVERRIDE {
499 CriticalSectionScoped lock(crit_.get());
500 // Rendering frame with timestamp associated with dropped packet -> FEC
501 // protection worked.
502 if (state_ == kProtectedPacketDropped &&
503 video_frame.timestamp() == protected_frame_timestamp_) {
504 observation_complete_->Set();
505 }
506 }
507
508 enum {
509 kFirstPacket,
510 kDropEveryOtherPacketUntilFec,
511 kDropNextMediaPacket,
512 kProtectedPacketDropped,
513 } state_;
514
515 virtual void ModifyConfigs(
516 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000517 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000518 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000519 // TODO(pbos): Run this test with combined NACK/FEC enabled as well.
520 // int rtp_history_ms = 1000;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000521 // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000522 // send_config->rtp.nack.rtp_history_ms = rtp_history_ms;
523 send_config->rtp.fec.red_payload_type = kRedPayloadType;
524 send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
525
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000526 (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
527 (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
528 (*receive_configs)[0].renderer = this;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000529 }
530
531 virtual void PerformTest() OVERRIDE {
532 EXPECT_EQ(kEventSignaled, Wait())
533 << "Timed out while waiting for retransmitted NACKed frames to be "
534 "rendered again.";
535 }
536
537 uint32_t protected_sequence_number_ GUARDED_BY(crit_);
538 uint32_t protected_frame_timestamp_ GUARDED_BY(crit_);
539 } test;
540
541 RunBaseTest(&test);
542}
543
544// This test drops second RTP packet with a marker bit set, makes sure it's
545// retransmitted and renders. Retransmission SSRCs are also checked.
546void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
547 static const int kDroppedFrameNumber = 2;
548 class RetransmissionObserver : public test::EndToEndTest,
549 public I420FrameCallback {
550 public:
551 explicit RetransmissionObserver(bool expect_rtx)
552 : EndToEndTest(kDefaultTimeoutMs),
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000553 retransmission_ssrc_(expect_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000554 retransmission_payload_type_(expect_rtx ? kSendRtxPayloadType
555 : kFakeSendPayloadType),
556 marker_bits_observed_(0),
557 retransmitted_timestamp_(0),
558 frame_retransmitted_(false) {}
559
560 private:
561 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
562 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000563 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000564
565 if (header.timestamp == retransmitted_timestamp_) {
566 EXPECT_EQ(retransmission_ssrc_, header.ssrc);
567 EXPECT_EQ(retransmission_payload_type_, header.payloadType);
568 frame_retransmitted_ = true;
569 return SEND_PACKET;
570 }
571
572 EXPECT_EQ(kSendSsrcs[0], header.ssrc);
573 EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
574
575 // Found the second frame's final packet, drop this and expect a
576 // retransmission.
577 if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) {
578 retransmitted_timestamp_ = header.timestamp;
579 return DROP_PACKET;
580 }
581
582 return SEND_PACKET;
583 }
584
585 virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE {
586 CriticalSectionScoped lock(crit_.get());
587 if (frame->timestamp() == retransmitted_timestamp_) {
588 EXPECT_TRUE(frame_retransmitted_);
589 observation_complete_->Set();
590 }
591 }
592
593 virtual void ModifyConfigs(
594 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000595 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000596 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000597 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000598 (*receive_configs)[0].pre_render_callback = this;
599 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000600 if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
601 send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000602 send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000603 (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc =
604 kSendRtxSsrcs[0];
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000605 (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].payload_type =
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000606 kSendRtxPayloadType;
607 }
608 }
609
610 virtual void PerformTest() OVERRIDE {
611 EXPECT_EQ(kEventSignaled, Wait())
612 << "Timed out while waiting for retransmission to render.";
613 }
614
615 const uint32_t retransmission_ssrc_;
616 const int retransmission_payload_type_;
617 int marker_bits_observed_;
618 uint32_t retransmitted_timestamp_;
619 bool frame_retransmitted_;
620 } test(retransmit_over_rtx);
621
622 RunBaseTest(&test);
623}
624
625TEST_F(EndToEndTest, DecodesRetransmittedFrame) {
626 DecodesRetransmittedFrame(false);
627}
628
629TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
630 DecodesRetransmittedFrame(true);
631}
632
andresp@webrtc.org02686112014-09-19 08:24:19 +0000633TEST_F(EndToEndTest, UsesFrameCallbacks) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000634 static const int kWidth = 320;
635 static const int kHeight = 240;
636
637 class Renderer : public VideoRenderer {
638 public:
639 Renderer() : event_(EventWrapper::Create()) {}
640
641 virtual void RenderFrame(const I420VideoFrame& video_frame,
642 int /*time_to_render_ms*/) OVERRIDE {
643 EXPECT_EQ(0, *video_frame.buffer(kYPlane))
644 << "Rendered frame should have zero luma which is applied by the "
645 "pre-render callback.";
646 event_->Set();
647 }
648
649 EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
650 scoped_ptr<EventWrapper> event_;
651 } renderer;
652
653 class TestFrameCallback : public I420FrameCallback {
654 public:
655 TestFrameCallback(int expected_luma_byte, int next_luma_byte)
656 : event_(EventWrapper::Create()),
657 expected_luma_byte_(expected_luma_byte),
658 next_luma_byte_(next_luma_byte) {}
659
660 EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
661
662 private:
663 virtual void FrameCallback(I420VideoFrame* frame) {
664 EXPECT_EQ(kWidth, frame->width())
665 << "Width not as expected, callback done before resize?";
666 EXPECT_EQ(kHeight, frame->height())
667 << "Height not as expected, callback done before resize?";
668
669 // Previous luma specified, observed luma should be fairly close.
670 if (expected_luma_byte_ != -1) {
671 EXPECT_NEAR(expected_luma_byte_, *frame->buffer(kYPlane), 10);
672 }
673
674 memset(frame->buffer(kYPlane),
675 next_luma_byte_,
676 frame->allocated_size(kYPlane));
677
678 event_->Set();
679 }
680
681 scoped_ptr<EventWrapper> event_;
682 int expected_luma_byte_;
683 int next_luma_byte_;
684 };
685
686 TestFrameCallback pre_encode_callback(-1, 255); // Changes luma to 255.
687 TestFrameCallback pre_render_callback(255, 0); // Changes luma from 255 to 0.
688
689 test::DirectTransport sender_transport, receiver_transport;
690
691 CreateCalls(Call::Config(&sender_transport),
692 Call::Config(&receiver_transport));
693
694 sender_transport.SetReceiver(receiver_call_->Receiver());
695 receiver_transport.SetReceiver(sender_call_->Receiver());
696
697 CreateSendConfig(1);
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +0000698 scoped_ptr<VideoEncoder> encoder(
699 VideoEncoder::Create(VideoEncoder::kVp8));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000700 send_config_.encoder_settings.encoder = encoder.get();
701 send_config_.encoder_settings.payload_name = "VP8";
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000702 ASSERT_EQ(1u, encoder_config_.streams.size()) << "Test setup error.";
703 encoder_config_.streams[0].width = kWidth;
704 encoder_config_.streams[0].height = kHeight;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000705 send_config_.pre_encode_callback = &pre_encode_callback;
706
707 CreateMatchingReceiveConfigs();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000708 receive_configs_[0].pre_render_callback = &pre_render_callback;
709 receive_configs_[0].renderer = &renderer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000710
711 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000712 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000713
714 // Create frames that are smaller than the send width/height, this is done to
715 // check that the callbacks are done after processing video.
716 scoped_ptr<test::FrameGenerator> frame_generator(
717 test::FrameGenerator::Create(kWidth / 2, kHeight / 2));
718 send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
719
720 EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait())
721 << "Timed out while waiting for pre-encode callback.";
722 EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
723 << "Timed out while waiting for pre-render callback.";
724 EXPECT_EQ(kEventSignaled, renderer.Wait())
725 << "Timed out while waiting for the frame to render.";
726
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000727 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000728
729 sender_transport.StopSending();
730 receiver_transport.StopSending();
731
732 DestroyStreams();
733}
734
735void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
736 static const int kPacketsToDrop = 1;
737
738 class PliObserver : public test::EndToEndTest, public VideoRenderer {
739 public:
740 explicit PliObserver(int rtp_history_ms)
741 : EndToEndTest(kLongTimeoutMs),
742 rtp_history_ms_(rtp_history_ms),
743 nack_enabled_(rtp_history_ms > 0),
744 highest_dropped_timestamp_(0),
745 frames_to_drop_(0),
746 received_pli_(false) {}
747
748 private:
749 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
750 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000751 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000752
753 // Drop all retransmitted packets to force a PLI.
754 if (header.timestamp <= highest_dropped_timestamp_)
755 return DROP_PACKET;
756
757 if (frames_to_drop_ > 0) {
758 highest_dropped_timestamp_ = header.timestamp;
759 --frames_to_drop_;
760 return DROP_PACKET;
761 }
762
763 return SEND_PACKET;
764 }
765
766 virtual Action OnReceiveRtcp(const uint8_t* packet,
767 size_t length) OVERRIDE {
768 RTCPUtility::RTCPParserV2 parser(packet, length, true);
769 EXPECT_TRUE(parser.IsValid());
770
771 for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
772 packet_type != RTCPUtility::kRtcpNotValidCode;
773 packet_type = parser.Iterate()) {
774 if (!nack_enabled_)
775 EXPECT_NE(packet_type, RTCPUtility::kRtcpRtpfbNackCode);
776
777 if (packet_type == RTCPUtility::kRtcpPsfbPliCode) {
778 received_pli_ = true;
779 break;
780 }
781 }
782 return SEND_PACKET;
783 }
784
785 virtual void RenderFrame(const I420VideoFrame& video_frame,
786 int time_to_render_ms) OVERRIDE {
787 CriticalSectionScoped lock(crit_.get());
788 if (received_pli_ &&
789 video_frame.timestamp() > highest_dropped_timestamp_) {
790 observation_complete_->Set();
791 }
792 if (!received_pli_)
793 frames_to_drop_ = kPacketsToDrop;
794 }
795
796 virtual void ModifyConfigs(
797 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000798 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000799 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000800 send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000801 (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
802 (*receive_configs)[0].renderer = this;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000803 }
804
805 virtual void PerformTest() OVERRIDE {
806 EXPECT_EQ(kEventSignaled, Wait()) << "Timed out waiting for PLI to be "
807 "received and a frame to be "
808 "rendered afterwards.";
809 }
810
811 int rtp_history_ms_;
812 bool nack_enabled_;
813 uint32_t highest_dropped_timestamp_;
814 int frames_to_drop_;
815 bool received_pli_;
816 } test(rtp_history_ms);
817
818 RunBaseTest(&test);
819}
820
821TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) {
822 ReceivesPliAndRecovers(1000);
823}
824
825// TODO(pbos): Enable this when 2250 is resolved.
826TEST_F(EndToEndTest, DISABLED_ReceivesPliAndRecoversWithoutNack) {
827 ReceivesPliAndRecovers(0);
828}
829
830TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
831 class PacketInputObserver : public PacketReceiver {
832 public:
833 explicit PacketInputObserver(PacketReceiver* receiver)
834 : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {}
835
836 EventTypeWrapper Wait() {
837 return delivered_packet_->Wait(kDefaultTimeoutMs);
838 }
839
840 private:
841 virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
842 size_t length) OVERRIDE {
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000843 if (RtpHeaderParser::IsRtcp(packet, length)) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000844 return receiver_->DeliverPacket(packet, length);
845 } else {
846 DeliveryStatus delivery_status =
847 receiver_->DeliverPacket(packet, length);
848 EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
849 delivered_packet_->Set();
850 return delivery_status;
851 }
852 }
853
854 PacketReceiver* receiver_;
855 scoped_ptr<EventWrapper> delivered_packet_;
856 };
857
858 test::DirectTransport send_transport, receive_transport;
859
860 CreateCalls(Call::Config(&send_transport), Call::Config(&receive_transport));
861 PacketInputObserver input_observer(receiver_call_->Receiver());
862
863 send_transport.SetReceiver(&input_observer);
864 receive_transport.SetReceiver(sender_call_->Receiver());
865
866 CreateSendConfig(1);
867 CreateMatchingReceiveConfigs();
868
869 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000870 CreateFrameGeneratorCapturer();
871 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000872
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000873 receiver_call_->DestroyVideoReceiveStream(receive_streams_[0]);
874 receive_streams_.clear();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000875
876 // Wait() waits for a received packet.
877 EXPECT_EQ(kEventSignaled, input_observer.Wait());
878
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000879 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000880
881 DestroyStreams();
882
883 send_transport.StopSending();
884 receive_transport.StopSending();
885}
886
887void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
888 static const int kNumCompoundRtcpPacketsToObserve = 10;
889 class RtcpModeObserver : public test::EndToEndTest {
890 public:
891 explicit RtcpModeObserver(newapi::RtcpMode rtcp_mode)
892 : EndToEndTest(kDefaultTimeoutMs),
893 rtcp_mode_(rtcp_mode),
894 sent_rtp_(0),
895 sent_rtcp_(0) {}
896
897 private:
898 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
899 if (++sent_rtp_ % 3 == 0)
900 return DROP_PACKET;
901
902 return SEND_PACKET;
903 }
904
905 virtual Action OnReceiveRtcp(const uint8_t* packet,
906 size_t length) OVERRIDE {
907 ++sent_rtcp_;
908 RTCPUtility::RTCPParserV2 parser(packet, length, true);
909 EXPECT_TRUE(parser.IsValid());
910
911 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
912 bool has_report_block = false;
913 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
914 EXPECT_NE(RTCPUtility::kRtcpSrCode, packet_type);
915 if (packet_type == RTCPUtility::kRtcpRrCode) {
916 has_report_block = true;
917 break;
918 }
919 packet_type = parser.Iterate();
920 }
921
922 switch (rtcp_mode_) {
923 case newapi::kRtcpCompound:
924 if (!has_report_block) {
925 ADD_FAILURE() << "Received RTCP packet without receiver report for "
926 "kRtcpCompound.";
927 observation_complete_->Set();
928 }
929
930 if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
931 observation_complete_->Set();
932
933 break;
934 case newapi::kRtcpReducedSize:
935 if (!has_report_block)
936 observation_complete_->Set();
937 break;
938 }
939
940 return SEND_PACKET;
941 }
942
943 virtual void ModifyConfigs(
944 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000945 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000946 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000947 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000948 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
949 (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000950 }
951
952 virtual void PerformTest() OVERRIDE {
953 EXPECT_EQ(kEventSignaled, Wait())
954 << (rtcp_mode_ == newapi::kRtcpCompound
955 ? "Timed out before observing enough compound packets."
956 : "Timed out before receiving a non-compound RTCP packet.");
957 }
958
959 newapi::RtcpMode rtcp_mode_;
960 int sent_rtp_;
961 int sent_rtcp_;
962 } test(rtcp_mode);
963
964 RunBaseTest(&test);
965}
966
967TEST_F(EndToEndTest, UsesRtcpCompoundMode) {
968 RespectsRtcpMode(newapi::kRtcpCompound);
969}
970
971TEST_F(EndToEndTest, UsesRtcpReducedSizeMode) {
972 RespectsRtcpMode(newapi::kRtcpReducedSize);
973}
974
975// Test sets up a Call multiple senders with different resolutions and SSRCs.
976// Another is set up to receive all three of these with different renderers.
977// Each renderer verifies that it receives the expected resolution, and as soon
978// as every renderer has received a frame, the test finishes.
andresp@webrtc.org02686112014-09-19 08:24:19 +0000979TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000980 static const size_t kNumStreams = 3;
981
982 class VideoOutputObserver : public VideoRenderer {
983 public:
984 VideoOutputObserver(test::FrameGeneratorCapturer** capturer,
985 int width,
986 int height)
987 : capturer_(capturer),
988 width_(width),
989 height_(height),
990 done_(EventWrapper::Create()) {}
991
992 virtual void RenderFrame(const I420VideoFrame& video_frame,
993 int time_to_render_ms) OVERRIDE {
994 EXPECT_EQ(width_, video_frame.width());
995 EXPECT_EQ(height_, video_frame.height());
996 (*capturer_)->Stop();
997 done_->Set();
998 }
999
1000 EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
1001
1002 private:
1003 test::FrameGeneratorCapturer** capturer_;
1004 int width_;
1005 int height_;
1006 scoped_ptr<EventWrapper> done_;
1007 };
1008
1009 struct {
1010 uint32_t ssrc;
1011 int width;
1012 int height;
1013 } codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}};
1014
1015 test::DirectTransport sender_transport, receiver_transport;
1016 scoped_ptr<Call> sender_call(Call::Create(Call::Config(&sender_transport)));
1017 scoped_ptr<Call> receiver_call(
1018 Call::Create(Call::Config(&receiver_transport)));
1019 sender_transport.SetReceiver(receiver_call->Receiver());
1020 receiver_transport.SetReceiver(sender_call->Receiver());
1021
1022 VideoSendStream* send_streams[kNumStreams];
1023 VideoReceiveStream* receive_streams[kNumStreams];
1024
1025 VideoOutputObserver* observers[kNumStreams];
1026 test::FrameGeneratorCapturer* frame_generators[kNumStreams];
1027
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +00001028 scoped_ptr<VideoEncoder> encoders[kNumStreams];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001029 for (size_t i = 0; i < kNumStreams; ++i)
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +00001030 encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001031
pbos@webrtc.org776e6f22014-10-29 15:28:39 +00001032 ScopedVector<VideoDecoder> allocated_decoders;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001033 for (size_t i = 0; i < kNumStreams; ++i) {
1034 uint32_t ssrc = codec_settings[i].ssrc;
1035 int width = codec_settings[i].width;
1036 int height = codec_settings[i].height;
1037 observers[i] = new VideoOutputObserver(&frame_generators[i], width, height);
1038
pbos@webrtc.orgbd249bc2014-07-07 04:45:15 +00001039 VideoSendStream::Config send_config;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001040 send_config.rtp.ssrcs.push_back(ssrc);
1041 send_config.encoder_settings.encoder = encoders[i].get();
1042 send_config.encoder_settings.payload_name = "VP8";
1043 send_config.encoder_settings.payload_type = 124;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001044 VideoEncoderConfig encoder_config;
1045 encoder_config.streams = test::CreateVideoStreams(1);
1046 VideoStream* stream = &encoder_config.streams[0];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001047 stream->width = width;
1048 stream->height = height;
1049 stream->max_framerate = 5;
1050 stream->min_bitrate_bps = stream->target_bitrate_bps =
1051 stream->max_bitrate_bps = 100000;
1052 send_streams[i] =
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001053 sender_call->CreateVideoSendStream(send_config, encoder_config);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001054 send_streams[i]->Start();
1055
pbos@webrtc.orgbd249bc2014-07-07 04:45:15 +00001056 VideoReceiveStream::Config receive_config;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001057 receive_config.renderer = observers[i];
1058 receive_config.rtp.remote_ssrc = ssrc;
1059 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
pbos@webrtc.org776e6f22014-10-29 15:28:39 +00001060 VideoReceiveStream::Decoder decoder =
1061 test::CreateMatchingDecoder(send_config.encoder_settings);
1062 allocated_decoders.push_back(decoder.decoder);
1063 receive_config.decoders.push_back(decoder);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001064 receive_streams[i] =
1065 receiver_call->CreateVideoReceiveStream(receive_config);
1066 receive_streams[i]->Start();
1067
1068 frame_generators[i] = test::FrameGeneratorCapturer::Create(
1069 send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock());
1070 frame_generators[i]->Start();
1071 }
1072
1073 for (size_t i = 0; i < kNumStreams; ++i) {
1074 EXPECT_EQ(kEventSignaled, observers[i]->Wait())
1075 << "Timed out while waiting for observer " << i << " to render.";
1076 }
1077
1078 for (size_t i = 0; i < kNumStreams; ++i) {
1079 frame_generators[i]->Stop();
1080 sender_call->DestroyVideoSendStream(send_streams[i]);
1081 receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
1082 delete frame_generators[i];
1083 delete observers[i];
1084 }
1085
1086 sender_transport.StopSending();
1087 receiver_transport.StopSending();
1088}
1089
1090TEST_F(EndToEndTest, ObserversEncodedFrames) {
1091 class EncodedFrameTestObserver : public EncodedFrameObserver {
1092 public:
1093 EncodedFrameTestObserver()
1094 : length_(0),
1095 frame_type_(kFrameEmpty),
1096 called_(EventWrapper::Create()) {}
1097 virtual ~EncodedFrameTestObserver() {}
1098
1099 virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
1100 frame_type_ = encoded_frame.frame_type_;
1101 length_ = encoded_frame.length_;
1102 buffer_.reset(new uint8_t[length_]);
1103 memcpy(buffer_.get(), encoded_frame.data_, length_);
1104 called_->Set();
1105 }
1106
1107 EventTypeWrapper Wait() { return called_->Wait(kDefaultTimeoutMs); }
1108
1109 void ExpectEqualFrames(const EncodedFrameTestObserver& observer) {
1110 ASSERT_EQ(length_, observer.length_)
1111 << "Observed frames are of different lengths.";
1112 EXPECT_EQ(frame_type_, observer.frame_type_)
1113 << "Observed frames have different frame types.";
1114 EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_))
1115 << "Observed encoded frames have different content.";
1116 }
1117
1118 private:
1119 scoped_ptr<uint8_t[]> buffer_;
1120 size_t length_;
1121 FrameType frame_type_;
1122 scoped_ptr<EventWrapper> called_;
1123 };
1124
1125 EncodedFrameTestObserver post_encode_observer;
1126 EncodedFrameTestObserver pre_decode_observer;
1127
1128 test::DirectTransport sender_transport, receiver_transport;
1129
1130 CreateCalls(Call::Config(&sender_transport),
1131 Call::Config(&receiver_transport));
1132
1133 sender_transport.SetReceiver(receiver_call_->Receiver());
1134 receiver_transport.SetReceiver(sender_call_->Receiver());
1135
1136 CreateSendConfig(1);
1137 CreateMatchingReceiveConfigs();
1138 send_config_.post_encode_callback = &post_encode_observer;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001139 receive_configs_[0].pre_decode_callback = &pre_decode_observer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001140
1141 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001142 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001143
1144 scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create(
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001145 encoder_config_.streams[0].width, encoder_config_.streams[0].height));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001146 send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
1147
1148 EXPECT_EQ(kEventSignaled, post_encode_observer.Wait())
1149 << "Timed out while waiting for send-side encoded-frame callback.";
1150
1151 EXPECT_EQ(kEventSignaled, pre_decode_observer.Wait())
1152 << "Timed out while waiting for pre-decode encoded-frame callback.";
1153
1154 post_encode_observer.ExpectEqualFrames(pre_decode_observer);
1155
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001156 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001157
1158 sender_transport.StopSending();
1159 receiver_transport.StopSending();
1160
1161 DestroyStreams();
1162}
1163
1164TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
1165 class RembObserver : public test::EndToEndTest {
1166 public:
1167 RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
1168
1169 virtual Action OnReceiveRtcp(const uint8_t* packet,
1170 size_t length) OVERRIDE {
1171 RTCPUtility::RTCPParserV2 parser(packet, length, true);
1172 EXPECT_TRUE(parser.IsValid());
1173
1174 bool received_psfb = false;
1175 bool received_remb = false;
1176 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
1177 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
1178 if (packet_type == RTCPUtility::kRtcpPsfbRembCode) {
1179 const RTCPUtility::RTCPPacket& packet = parser.Packet();
1180 EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc);
1181 received_psfb = true;
1182 } else if (packet_type == RTCPUtility::kRtcpPsfbRembItemCode) {
1183 const RTCPUtility::RTCPPacket& packet = parser.Packet();
1184 EXPECT_GT(packet.REMBItem.BitRate, 0u);
1185 EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u);
1186 EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]);
1187 received_remb = true;
1188 }
1189 packet_type = parser.Iterate();
1190 }
1191 if (received_psfb && received_remb)
1192 observation_complete_->Set();
1193 return SEND_PACKET;
1194 }
1195 virtual void PerformTest() OVERRIDE {
1196 EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for a "
1197 "receiver RTCP REMB packet to be "
1198 "sent.";
1199 }
1200 } test;
1201
1202 RunBaseTest(&test);
1203}
1204
stefan@webrtc.org0bae1fa2014-11-05 14:05:29 +00001205TEST_F(EndToEndTest, VerifyBandwidthStats) {
1206 class RtcpObserver : public test::EndToEndTest, public PacketReceiver {
1207 public:
1208 RtcpObserver()
1209 : EndToEndTest(kDefaultTimeoutMs),
1210 sender_call_(NULL),
1211 receiver_call_(NULL),
1212 has_seen_pacer_delay_(false) {}
1213
1214 virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
1215 size_t length) OVERRIDE {
1216 Call::Stats sender_stats = sender_call_->GetStats();
1217 Call::Stats receiver_stats = receiver_call_->GetStats();
1218 if (!has_seen_pacer_delay_)
1219 has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
1220 if (sender_stats.send_bandwidth_bps > 0 &&
1221 receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_)
1222 observation_complete_->Set();
1223 return receiver_call_->Receiver()->DeliverPacket(packet, length);
1224 }
1225
1226 virtual void OnCallsCreated(Call* sender_call,
1227 Call* receiver_call) OVERRIDE {
1228 sender_call_ = sender_call;
1229 receiver_call_ = receiver_call;
1230 }
1231
1232 virtual void PerformTest() OVERRIDE {
1233 EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
1234 "non-zero bandwidth stats.";
1235 }
1236
1237 virtual void SetReceivers(
1238 PacketReceiver* send_transport_receiver,
1239 PacketReceiver* receive_transport_receiver) OVERRIDE {
1240 test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
1241 }
1242
1243 private:
1244 Call* sender_call_;
1245 Call* receiver_call_;
1246 bool has_seen_pacer_delay_;
1247 } test;
1248
1249 RunBaseTest(&test);
1250}
1251
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001252void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
1253 static const int kNumRtcpReportPacketsToObserve = 5;
1254 class RtcpXrObserver : public test::EndToEndTest {
1255 public:
1256 explicit RtcpXrObserver(bool enable_rrtr)
1257 : EndToEndTest(kDefaultTimeoutMs),
1258 enable_rrtr_(enable_rrtr),
1259 sent_rtcp_sr_(0),
1260 sent_rtcp_rr_(0),
1261 sent_rtcp_rrtr_(0),
1262 sent_rtcp_dlrr_(0) {}
1263
1264 private:
1265 // Receive stream should send RR packets (and RRTR packets if enabled).
1266 virtual Action OnReceiveRtcp(const uint8_t* packet,
1267 size_t length) OVERRIDE {
1268 RTCPUtility::RTCPParserV2 parser(packet, length, true);
1269 EXPECT_TRUE(parser.IsValid());
1270
1271 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
1272 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
1273 if (packet_type == RTCPUtility::kRtcpRrCode) {
1274 ++sent_rtcp_rr_;
1275 } else if (packet_type ==
1276 RTCPUtility::kRtcpXrReceiverReferenceTimeCode) {
1277 ++sent_rtcp_rrtr_;
1278 }
1279 EXPECT_NE(packet_type, RTCPUtility::kRtcpSrCode);
1280 EXPECT_NE(packet_type, RTCPUtility::kRtcpXrDlrrReportBlockItemCode);
1281 packet_type = parser.Iterate();
1282 }
1283 return SEND_PACKET;
1284 }
1285 // Send stream should send SR packets (and DLRR packets if enabled).
1286 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
1287 RTCPUtility::RTCPParserV2 parser(packet, length, true);
1288 EXPECT_TRUE(parser.IsValid());
1289
1290 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
1291 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
1292 if (packet_type == RTCPUtility::kRtcpSrCode) {
1293 ++sent_rtcp_sr_;
1294 } else if (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) {
1295 ++sent_rtcp_dlrr_;
1296 }
1297 EXPECT_NE(packet_type, RTCPUtility::kRtcpXrReceiverReferenceTimeCode);
1298 packet_type = parser.Iterate();
1299 }
1300 if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
1301 sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) {
1302 if (enable_rrtr_) {
1303 EXPECT_GT(sent_rtcp_rrtr_, 0);
1304 EXPECT_GT(sent_rtcp_dlrr_, 0);
1305 } else {
1306 EXPECT_EQ(0, sent_rtcp_rrtr_);
1307 EXPECT_EQ(0, sent_rtcp_dlrr_);
1308 }
1309 observation_complete_->Set();
1310 }
1311 return SEND_PACKET;
1312 }
1313
1314 virtual void ModifyConfigs(
1315 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001316 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001317 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001318 (*receive_configs)[0].rtp.rtcp_mode = newapi::kRtcpReducedSize;
1319 (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
1320 enable_rrtr_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001321 }
1322
1323 virtual void PerformTest() OVERRIDE {
1324 EXPECT_EQ(kEventSignaled, Wait())
1325 << "Timed out while waiting for RTCP SR/RR packets to be sent.";
1326 }
1327
1328 bool enable_rrtr_;
1329 int sent_rtcp_sr_;
1330 int sent_rtcp_rr_;
1331 int sent_rtcp_rrtr_;
1332 int sent_rtcp_dlrr_;
1333 } test(enable_rrtr);
1334
1335 RunBaseTest(&test);
1336}
1337
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001338void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
1339 bool send_single_ssrc_first) {
1340 class SendsSetSsrcs : public test::EndToEndTest {
1341 public:
1342 SendsSetSsrcs(const uint32_t* ssrcs,
1343 size_t num_ssrcs,
1344 bool send_single_ssrc_first)
1345 : EndToEndTest(kDefaultTimeoutMs),
1346 num_ssrcs_(num_ssrcs),
1347 send_single_ssrc_first_(send_single_ssrc_first),
1348 ssrcs_to_observe_(num_ssrcs),
1349 expect_single_ssrc_(send_single_ssrc_first) {
1350 for (size_t i = 0; i < num_ssrcs; ++i)
1351 valid_ssrcs_[ssrcs[i]] = true;
1352 }
1353
1354 private:
1355 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1356 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001357 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001358
1359 EXPECT_TRUE(valid_ssrcs_[header.ssrc])
1360 << "Received unknown SSRC: " << header.ssrc;
1361
1362 if (!valid_ssrcs_[header.ssrc])
1363 observation_complete_->Set();
1364
1365 if (!is_observed_[header.ssrc]) {
1366 is_observed_[header.ssrc] = true;
1367 --ssrcs_to_observe_;
1368 if (expect_single_ssrc_) {
1369 expect_single_ssrc_ = false;
1370 observation_complete_->Set();
1371 }
1372 }
1373
1374 if (ssrcs_to_observe_ == 0)
1375 observation_complete_->Set();
1376
1377 return SEND_PACKET;
1378 }
1379
1380 virtual size_t GetNumStreams() const OVERRIDE { return num_ssrcs_; }
1381
1382 virtual void ModifyConfigs(
1383 VideoSendStream::Config* send_config,
1384 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001385 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001386 if (num_ssrcs_ > 1) {
1387 // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001388 for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
1389 encoder_config->streams[i].min_bitrate_bps = 10000;
1390 encoder_config->streams[i].target_bitrate_bps = 15000;
1391 encoder_config->streams[i].max_bitrate_bps = 20000;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001392 }
1393 }
1394
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001395 encoder_config_all_streams_ = *encoder_config;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001396 if (send_single_ssrc_first_)
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001397 encoder_config->streams.resize(1);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001398 }
1399
1400 virtual void OnStreamsCreated(
1401 VideoSendStream* send_stream,
1402 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
1403 send_stream_ = send_stream;
1404 }
1405
1406 virtual void PerformTest() OVERRIDE {
1407 EXPECT_EQ(kEventSignaled, Wait())
1408 << "Timed out while waiting for "
1409 << (send_single_ssrc_first_ ? "first SSRC." : "SSRCs.");
1410
1411 if (send_single_ssrc_first_) {
1412 // Set full simulcast and continue with the rest of the SSRCs.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001413 send_stream_->ReconfigureVideoEncoder(encoder_config_all_streams_);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001414 EXPECT_EQ(kEventSignaled, Wait())
1415 << "Timed out while waiting on additional SSRCs.";
1416 }
1417 }
1418
1419 private:
1420 std::map<uint32_t, bool> valid_ssrcs_;
1421 std::map<uint32_t, bool> is_observed_;
1422
1423 const size_t num_ssrcs_;
1424 const bool send_single_ssrc_first_;
1425
1426 size_t ssrcs_to_observe_;
1427 bool expect_single_ssrc_;
1428
1429 VideoSendStream* send_stream_;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001430 VideoEncoderConfig encoder_config_all_streams_;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001431 } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first);
1432
1433 RunBaseTest(&test);
1434}
1435
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001436TEST_F(EndToEndTest, GetStats) {
1437 class StatsObserver : public test::EndToEndTest, public I420FrameCallback {
1438 public:
1439 StatsObserver()
1440 : EndToEndTest(kLongTimeoutMs),
1441 receive_stream_(NULL),
1442 send_stream_(NULL),
1443 expected_receive_ssrc_(),
1444 expected_send_ssrcs_(),
1445 check_stats_event_(EventWrapper::Create()) {}
1446
1447 private:
1448 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1449 check_stats_event_->Set();
1450 return SEND_PACKET;
1451 }
1452
1453 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
1454 check_stats_event_->Set();
1455 return SEND_PACKET;
1456 }
1457
1458 virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE {
1459 check_stats_event_->Set();
1460 return SEND_PACKET;
1461 }
1462
1463 virtual Action OnReceiveRtcp(const uint8_t* packet,
1464 size_t length) OVERRIDE {
1465 check_stats_event_->Set();
1466 return SEND_PACKET;
1467 }
1468
1469 virtual void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
1470 // Ensure that we have at least 5ms send side delay.
1471 int64_t render_time = video_frame->render_time_ms();
1472 if (render_time > 0)
1473 video_frame->set_render_time_ms(render_time - 5);
1474 }
1475
1476 bool CheckReceiveStats() {
1477 assert(receive_stream_ != NULL);
1478 VideoReceiveStream::Stats stats = receive_stream_->GetStats();
1479 EXPECT_EQ(expected_receive_ssrc_, stats.ssrc);
1480
1481 // Make sure all fields have been populated.
1482
1483 receive_stats_filled_["IncomingRate"] |=
stefan@webrtc.org0bae1fa2014-11-05 14:05:29 +00001484 stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001485
1486 receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
1487
1488 receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
1489
1490 receive_stats_filled_["StatisticsUpdated"] |=
1491 stats.rtcp_stats.cumulative_lost != 0 ||
1492 stats.rtcp_stats.extended_max_sequence_number != 0 ||
1493 stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
1494
1495 receive_stats_filled_["DataCountersUpdated"] |=
1496 stats.rtp_stats.bytes != 0 || stats.rtp_stats.fec_packets != 0 ||
1497 stats.rtp_stats.header_bytes != 0 || stats.rtp_stats.packets != 0 ||
1498 stats.rtp_stats.padding_bytes != 0 ||
1499 stats.rtp_stats.retransmitted_packets != 0;
1500
1501 receive_stats_filled_["CodecStats"] |=
1502 stats.avg_delay_ms != 0 || stats.discarded_packets != 0 ||
1503 stats.key_frames != 0 || stats.delta_frames != 0;
1504
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001505 return AllStatsFilled(receive_stats_filled_);
1506 }
1507
1508 bool CheckSendStats() {
1509 assert(send_stream_ != NULL);
1510 VideoSendStream::Stats stats = send_stream_->GetStats();
1511
1512 send_stats_filled_["NumStreams"] |=
1513 stats.substreams.size() == expected_send_ssrcs_.size();
1514
stefan@webrtc.org0bae1fa2014-11-05 14:05:29 +00001515 for (std::map<uint32_t, SsrcStats>::const_iterator it =
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001516 stats.substreams.begin();
1517 it != stats.substreams.end();
1518 ++it) {
1519 EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
1520 expected_send_ssrcs_.end());
1521
1522 send_stats_filled_[CompoundKey("IncomingRate", it->first)] |=
1523 stats.input_frame_rate != 0;
1524
stefan@webrtc.org0bae1fa2014-11-05 14:05:29 +00001525 const SsrcStats& stream_stats = it->second;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001526
1527 send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
1528 stream_stats.rtcp_stats.cumulative_lost != 0 ||
1529 stream_stats.rtcp_stats.extended_max_sequence_number != 0 ||
1530 stream_stats.rtcp_stats.fraction_lost != 0;
1531
1532 send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |=
1533 stream_stats.rtp_stats.fec_packets != 0 ||
1534 stream_stats.rtp_stats.padding_bytes != 0 ||
1535 stream_stats.rtp_stats.retransmitted_packets != 0 ||
1536 stream_stats.rtp_stats.packets != 0;
1537
1538 send_stats_filled_[CompoundKey("BitrateStatisticsObserver",
1539 it->first)] |=
stefan@webrtc.org0bae1fa2014-11-05 14:05:29 +00001540 stream_stats.total_bitrate_bps != 0;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001541
1542 send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
1543 stream_stats.delta_frames != 0 || stream_stats.key_frames != 0;
1544
1545 send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
1546 stats.encode_frame_rate != 0;
stefan@webrtc.org168f23f2014-07-11 13:44:02 +00001547
1548 send_stats_filled_[CompoundKey("Delay", it->first)] |=
1549 stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001550 }
1551
1552 return AllStatsFilled(send_stats_filled_);
1553 }
1554
1555 std::string CompoundKey(const char* name, uint32_t ssrc) {
1556 std::ostringstream oss;
1557 oss << name << "_" << ssrc;
1558 return oss.str();
1559 }
1560
1561 bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
1562 for (std::map<std::string, bool>::const_iterator it = stats_map.begin();
1563 it != stats_map.end();
1564 ++it) {
1565 if (!it->second)
1566 return false;
1567 }
1568 return true;
1569 }
1570
1571 virtual void ModifyConfigs(
1572 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001573 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001574 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001575 send_config->pre_encode_callback = this; // Used to inject delay.
1576 send_config->rtp.c_name = "SomeCName";
1577
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001578 expected_receive_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001579 const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
1580 for (size_t i = 0; i < ssrcs.size(); ++i)
1581 expected_send_ssrcs_.insert(ssrcs[i]);
1582
1583 expected_cname_ = send_config->rtp.c_name;
1584 }
1585
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001586 virtual void OnStreamsCreated(
1587 VideoSendStream* send_stream,
1588 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001589 send_stream_ = send_stream;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001590 receive_stream_ = receive_streams[0];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001591 }
1592
1593 virtual void PerformTest() OVERRIDE {
1594 Clock* clock = Clock::GetRealTimeClock();
1595 int64_t now = clock->TimeInMilliseconds();
1596 int64_t stop_time = now + test::CallTest::kLongTimeoutMs;
1597 bool receive_ok = false;
1598 bool send_ok = false;
1599
1600 while (now < stop_time) {
1601 if (!receive_ok)
1602 receive_ok = CheckReceiveStats();
1603 if (!send_ok)
1604 send_ok = CheckSendStats();
1605
1606 if (receive_ok && send_ok)
1607 return;
1608
1609 int64_t time_until_timout_ = stop_time - now;
1610 if (time_until_timout_ > 0)
1611 check_stats_event_->Wait(time_until_timout_);
1612 now = clock->TimeInMilliseconds();
1613 }
1614
1615 ADD_FAILURE() << "Timed out waiting for filled stats.";
1616 for (std::map<std::string, bool>::const_iterator it =
1617 receive_stats_filled_.begin();
1618 it != receive_stats_filled_.end();
1619 ++it) {
1620 if (!it->second) {
1621 ADD_FAILURE() << "Missing receive stats: " << it->first;
1622 }
1623 }
1624
1625 for (std::map<std::string, bool>::const_iterator it =
1626 send_stats_filled_.begin();
1627 it != send_stats_filled_.end();
1628 ++it) {
1629 if (!it->second) {
1630 ADD_FAILURE() << "Missing send stats: " << it->first;
1631 }
1632 }
1633 }
1634
1635 VideoReceiveStream* receive_stream_;
1636 std::map<std::string, bool> receive_stats_filled_;
1637
1638 VideoSendStream* send_stream_;
1639 std::map<std::string, bool> send_stats_filled_;
1640
1641 uint32_t expected_receive_ssrc_;
1642 std::set<uint32_t> expected_send_ssrcs_;
1643 std::string expected_cname_;
1644
1645 scoped_ptr<EventWrapper> check_stats_event_;
1646 } test;
1647
1648 RunBaseTest(&test);
1649}
1650
1651TEST_F(EndToEndTest, ReceiverReferenceTimeReportEnabled) {
1652 TestXrReceiverReferenceTimeReport(true);
1653}
1654
1655TEST_F(EndToEndTest, ReceiverReferenceTimeReportDisabled) {
1656 TestXrReceiverReferenceTimeReport(false);
1657}
1658
1659TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
1660 static const size_t kNumRtpPacketsToSend = 5;
1661 class ReceivedRtpStatsObserver : public test::EndToEndTest {
1662 public:
1663 ReceivedRtpStatsObserver()
1664 : EndToEndTest(kDefaultTimeoutMs),
1665 receive_stream_(NULL),
1666 sent_rtp_(0) {}
1667
1668 private:
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001669 virtual void OnStreamsCreated(
1670 VideoSendStream* send_stream,
1671 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
1672 receive_stream_ = receive_streams[0];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001673 }
1674
1675 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1676 if (sent_rtp_ >= kNumRtpPacketsToSend) {
1677 VideoReceiveStream::Stats stats = receive_stream_->GetStats();
1678 if (kNumRtpPacketsToSend == stats.rtp_stats.packets) {
1679 observation_complete_->Set();
1680 }
1681 return DROP_PACKET;
1682 }
1683 ++sent_rtp_;
1684 return SEND_PACKET;
1685 }
1686
1687 virtual void PerformTest() OVERRIDE {
1688 EXPECT_EQ(kEventSignaled, Wait())
1689 << "Timed out while verifying number of received RTP packets.";
1690 }
1691
1692 VideoReceiveStream* receive_stream_;
1693 uint32_t sent_rtp_;
1694 } test;
1695
1696 RunBaseTest(&test);
1697}
1698
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001699TEST_F(EndToEndTest, SendsSetSsrc) { TestSendsSetSsrcs(1, false); }
1700
1701TEST_F(EndToEndTest, SendsSetSimulcastSsrcs) {
1702 TestSendsSetSsrcs(kNumSsrcs, false);
1703}
1704
1705TEST_F(EndToEndTest, CanSwitchToUseAllSsrcs) {
1706 TestSendsSetSsrcs(kNumSsrcs, true);
1707}
1708
mflodman@webrtc.orgf9460682014-07-24 16:41:25 +00001709TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001710 class ObserveRedundantPayloads: public test::EndToEndTest {
1711 public:
1712 ObserveRedundantPayloads()
1713 : EndToEndTest(kDefaultTimeoutMs), ssrcs_to_observe_(kNumSsrcs) {
pbos@webrtc.orgdde16f12014-08-05 23:35:43 +00001714 for (size_t i = 0; i < kNumSsrcs; ++i) {
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001715 registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
1716 }
1717 }
1718
1719 private:
1720 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1721 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001722 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001723
1724 if (!registered_rtx_ssrc_[header.ssrc])
1725 return SEND_PACKET;
1726
1727 EXPECT_LE(static_cast<size_t>(header.headerLength + header.paddingLength),
1728 length);
1729 const bool packet_is_redundant_payload =
1730 static_cast<size_t>(header.headerLength + header.paddingLength) <
1731 length;
1732
1733 if (!packet_is_redundant_payload)
1734 return SEND_PACKET;
1735
1736 if (!observed_redundant_retransmission_[header.ssrc]) {
1737 observed_redundant_retransmission_[header.ssrc] = true;
1738 if (--ssrcs_to_observe_ == 0)
1739 observation_complete_->Set();
1740 }
1741
1742 return SEND_PACKET;
1743 }
1744
1745 virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; }
1746
1747 virtual void ModifyConfigs(
1748 VideoSendStream::Config* send_config,
1749 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001750 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001751 // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001752 for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
1753 encoder_config->streams[i].min_bitrate_bps = 10000;
1754 encoder_config->streams[i].target_bitrate_bps = 15000;
1755 encoder_config->streams[i].max_bitrate_bps = 20000;
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001756 }
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001757
1758 send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
1759 send_config->rtp.rtx.pad_with_redundant_payloads = true;
1760
1761 for (size_t i = 0; i < kNumSsrcs; ++i)
1762 send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
pbos@webrtc.orgad3b5a52014-10-24 09:23:21 +00001763
1764 // Significantly higher than max bitrates for all video streams -> forcing
1765 // padding to trigger redundant padding on all RTX SSRCs.
1766 encoder_config->min_transmit_bitrate_bps = 100000;
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001767 }
1768
1769 virtual void PerformTest() OVERRIDE {
1770 EXPECT_EQ(kEventSignaled, Wait())
1771 << "Timed out while waiting for redundant payloads on all SSRCs.";
1772 }
1773
1774 private:
1775 size_t ssrcs_to_observe_;
1776 std::map<uint32_t, bool> observed_redundant_retransmission_;
1777 std::map<uint32_t, bool> registered_rtx_ssrc_;
1778 } test;
1779
1780 RunBaseTest(&test);
1781}
1782
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001783void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
1784 static const uint32_t kMaxSequenceNumberGap = 100;
1785 static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
1786 class RtpSequenceObserver : public test::RtpRtcpObserver {
1787 public:
pbos@webrtc.orgdde16f12014-08-05 23:35:43 +00001788 explicit RtpSequenceObserver(bool use_rtx)
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001789 : test::RtpRtcpObserver(kDefaultTimeoutMs),
1790 crit_(CriticalSectionWrapper::CreateCriticalSection()),
1791 ssrcs_to_observe_(kNumSsrcs) {
1792 for (size_t i = 0; i < kNumSsrcs; ++i) {
1793 configured_ssrcs_[kSendSsrcs[i]] = true;
1794 if (use_rtx)
1795 configured_ssrcs_[kSendRtxSsrcs[i]] = true;
1796 }
1797 }
1798
1799 void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
1800 CriticalSectionScoped lock(crit_.get());
1801 ssrc_observed_.clear();
1802 ssrcs_to_observe_ = num_expected_ssrcs;
1803 }
1804
1805 private:
1806 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1807 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001808 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001809 const uint32_t ssrc = header.ssrc;
1810 const uint16_t sequence_number = header.sequenceNumber;
1811 const uint32_t timestamp = header.timestamp;
1812 const bool only_padding =
1813 static_cast<size_t>(header.headerLength + header.paddingLength) ==
1814 length;
1815
1816 EXPECT_TRUE(configured_ssrcs_[ssrc])
1817 << "Received SSRC that wasn't configured: " << ssrc;
1818
1819 std::map<uint32_t, uint16_t>::iterator it =
1820 last_observed_sequence_number_.find(header.ssrc);
1821 if (it == last_observed_sequence_number_.end()) {
1822 last_observed_sequence_number_[ssrc] = sequence_number;
1823 last_observed_timestamp_[ssrc] = timestamp;
1824 } else {
1825 // Verify sequence numbers are reasonably close.
1826 uint32_t extended_sequence_number = sequence_number;
1827 // Check for roll-over.
1828 if (sequence_number < last_observed_sequence_number_[ssrc])
1829 extended_sequence_number += 0xFFFFu + 1;
1830 EXPECT_LE(
1831 extended_sequence_number - last_observed_sequence_number_[ssrc],
1832 kMaxSequenceNumberGap)
1833 << "Gap in sequence numbers ("
1834 << last_observed_sequence_number_[ssrc] << " -> " << sequence_number
1835 << ") too large for SSRC: " << ssrc << ".";
1836 last_observed_sequence_number_[ssrc] = sequence_number;
1837
1838 // TODO(pbos): Remove this check if we ever have monotonically
1839 // increasing timestamps. Right now padding packets add a delta which
1840 // can cause reordering between padding packets and regular packets,
1841 // hence we drop padding-only packets to not flake.
1842 if (only_padding) {
1843 // Verify that timestamps are reasonably close.
1844 uint64_t extended_timestamp = timestamp;
1845 // Check for roll-over.
1846 if (timestamp < last_observed_timestamp_[ssrc])
1847 extended_timestamp += static_cast<uint64_t>(0xFFFFFFFFu) + 1;
1848 EXPECT_LE(extended_timestamp - last_observed_timestamp_[ssrc],
1849 kMaxTimestampGap)
1850 << "Gap in timestamps (" << last_observed_timestamp_[ssrc]
1851 << " -> " << timestamp << ") too large for SSRC: " << ssrc << ".";
1852 }
1853 last_observed_timestamp_[ssrc] = timestamp;
1854 }
1855
1856 CriticalSectionScoped lock(crit_.get());
1857 // Wait for media packets on all ssrcs.
1858 if (!ssrc_observed_[ssrc] && !only_padding) {
1859 ssrc_observed_[ssrc] = true;
1860 if (--ssrcs_to_observe_ == 0)
1861 observation_complete_->Set();
1862 }
1863
1864 return SEND_PACKET;
1865 }
1866
1867 std::map<uint32_t, uint16_t> last_observed_sequence_number_;
1868 std::map<uint32_t, uint32_t> last_observed_timestamp_;
1869 std::map<uint32_t, bool> configured_ssrcs_;
1870
1871 scoped_ptr<CriticalSectionWrapper> crit_;
1872 size_t ssrcs_to_observe_ GUARDED_BY(crit_);
1873 std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_);
1874 } observer(use_rtx);
1875
1876 CreateCalls(Call::Config(observer.SendTransport()),
1877 Call::Config(observer.ReceiveTransport()));
1878 observer.SetReceivers(sender_call_->Receiver(), NULL);
1879
1880 CreateSendConfig(kNumSsrcs);
1881
1882 if (use_rtx) {
1883 for (size_t i = 0; i < kNumSsrcs; ++i) {
1884 send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
1885 }
1886 send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
1887 }
1888
1889 // Lower bitrates so that all streams send initially.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001890 for (size_t i = 0; i < encoder_config_.streams.size(); ++i) {
1891 encoder_config_.streams[i].min_bitrate_bps = 10000;
1892 encoder_config_.streams[i].target_bitrate_bps = 15000;
1893 encoder_config_.streams[i].max_bitrate_bps = 20000;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001894 }
1895
pbos@webrtc.org32452b22014-10-22 12:15:24 +00001896 // Use the same total bitrates when sending a single stream to avoid lowering
1897 // the bitrate estimate and requiring a subsequent rampup.
1898 VideoEncoderConfig one_stream = encoder_config_;
1899 one_stream.streams.resize(1);
1900 for (size_t i = 1; i < encoder_config_.streams.size(); ++i) {
1901 one_stream.streams.front().min_bitrate_bps +=
1902 encoder_config_.streams[i].min_bitrate_bps;
1903 one_stream.streams.front().target_bitrate_bps +=
1904 encoder_config_.streams[i].target_bitrate_bps;
1905 one_stream.streams.front().max_bitrate_bps +=
1906 encoder_config_.streams[i].max_bitrate_bps;
1907 }
1908
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001909 CreateMatchingReceiveConfigs();
1910
1911 CreateStreams();
1912 CreateFrameGeneratorCapturer();
1913
1914 Start();
1915 EXPECT_EQ(kEventSignaled, observer.Wait())
1916 << "Timed out waiting for all SSRCs to send packets.";
1917
1918 // Test stream resetting more than once to make sure that the state doesn't
1919 // get set once (this could be due to using std::map::insert for instance).
1920 for (size_t i = 0; i < 3; ++i) {
1921 frame_generator_capturer_->Stop();
1922 sender_call_->DestroyVideoSendStream(send_stream_);
1923
1924 // Re-create VideoSendStream with only one stream.
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001925 send_stream_ =
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001926 sender_call_->CreateVideoSendStream(send_config_, one_stream);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001927 send_stream_->Start();
1928 CreateFrameGeneratorCapturer();
1929 frame_generator_capturer_->Start();
1930
1931 observer.ResetExpectedSsrcs(1);
1932 EXPECT_EQ(kEventSignaled, observer.Wait())
1933 << "Timed out waiting for single RTP packet.";
1934
1935 // Reconfigure back to use all streams.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001936 send_stream_->ReconfigureVideoEncoder(encoder_config_);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001937 observer.ResetExpectedSsrcs(kNumSsrcs);
1938 EXPECT_EQ(kEventSignaled, observer.Wait())
1939 << "Timed out waiting for all SSRCs to send packets.";
1940
1941 // Reconfigure down to one stream.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001942 send_stream_->ReconfigureVideoEncoder(one_stream);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001943 observer.ResetExpectedSsrcs(1);
1944 EXPECT_EQ(kEventSignaled, observer.Wait())
1945 << "Timed out waiting for single RTP packet.";
1946
1947 // Reconfigure back to use all streams.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001948 send_stream_->ReconfigureVideoEncoder(encoder_config_);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001949 observer.ResetExpectedSsrcs(kNumSsrcs);
1950 EXPECT_EQ(kEventSignaled, observer.Wait())
1951 << "Timed out waiting for all SSRCs to send packets.";
1952 }
1953
1954 observer.StopSending();
1955
1956 Stop();
1957 DestroyStreams();
1958}
1959
aluebs@webrtc.orgb623c5c2014-08-26 14:22:51 +00001960TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpState) {
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001961 TestRtpStatePreservation(false);
1962}
1963
1964TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
1965 TestRtpStatePreservation(true);
1966}
1967
pbos@webrtc.org26c0c412014-09-03 16:17:12 +00001968TEST_F(EndToEndTest, RespectsNetworkState) {
1969 // TODO(pbos): Remove accepted downtime packets etc. when signaling network
1970 // down blocks until no more packets will be sent.
1971
1972 // Pacer will send from its packet list and then send required padding before
1973 // checking paused_ again. This should be enough for one round of pacing,
1974 // otherwise increase.
1975 static const int kNumAcceptedDowntimeRtp = 5;
1976 // A single RTCP may be in the pipeline.
1977 static const int kNumAcceptedDowntimeRtcp = 1;
1978 class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
1979 public:
1980 NetworkStateTest()
1981 : EndToEndTest(kDefaultTimeoutMs),
1982 FakeEncoder(Clock::GetRealTimeClock()),
1983 test_crit_(CriticalSectionWrapper::CreateCriticalSection()),
1984 encoded_frames_(EventWrapper::Create()),
1985 sender_packets_(EventWrapper::Create()),
1986 receiver_packets_(EventWrapper::Create()),
1987 sender_state_(Call::kNetworkUp),
1988 down_sender_rtp_(0),
1989 down_sender_rtcp_(0),
1990 receiver_state_(Call::kNetworkUp),
1991 down_receiver_rtcp_(0),
1992 down_frames_(0) {}
1993
1994 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1995 CriticalSectionScoped lock(test_crit_.get());
1996 if (sender_state_ == Call::kNetworkDown) {
1997 ++down_sender_rtp_;
1998 EXPECT_LE(down_sender_rtp_, kNumAcceptedDowntimeRtp)
1999 << "RTP sent during sender-side downtime.";
2000 if (down_sender_rtp_> kNumAcceptedDowntimeRtp)
2001 sender_packets_->Set();
2002 } else {
2003 sender_packets_->Set();
2004 }
2005 return SEND_PACKET;
2006 }
2007
2008 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
2009 CriticalSectionScoped lock(test_crit_.get());
2010 if (sender_state_ == Call::kNetworkDown) {
2011 ++down_sender_rtcp_;
2012 EXPECT_LE(down_sender_rtcp_, kNumAcceptedDowntimeRtcp)
2013 << "RTCP sent during sender-side downtime.";
2014 if (down_sender_rtcp_ > kNumAcceptedDowntimeRtcp)
2015 sender_packets_->Set();
2016 } else {
2017 sender_packets_->Set();
2018 }
2019 return SEND_PACKET;
2020 }
2021
2022 virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE {
2023 ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
2024 return SEND_PACKET;
2025 }
2026
2027 virtual Action OnReceiveRtcp(const uint8_t* packet,
2028 size_t length) OVERRIDE {
2029 CriticalSectionScoped lock(test_crit_.get());
2030 if (receiver_state_ == Call::kNetworkDown) {
2031 ++down_receiver_rtcp_;
2032 EXPECT_LE(down_receiver_rtcp_, kNumAcceptedDowntimeRtcp)
2033 << "RTCP sent during receiver-side downtime.";
2034 if (down_receiver_rtcp_ > kNumAcceptedDowntimeRtcp)
2035 receiver_packets_->Set();
2036 } else {
2037 receiver_packets_->Set();
2038 }
2039 return SEND_PACKET;
2040 }
2041
2042 virtual void OnCallsCreated(Call* sender_call,
2043 Call* receiver_call) OVERRIDE {
2044 sender_call_ = sender_call;
2045 receiver_call_ = receiver_call;
2046 }
2047
2048 virtual void ModifyConfigs(
2049 VideoSendStream::Config* send_config,
2050 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00002051 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org26c0c412014-09-03 16:17:12 +00002052 send_config->encoder_settings.encoder = this;
2053 }
2054
2055 virtual void PerformTest() OVERRIDE {
2056 EXPECT_EQ(kEventSignaled, encoded_frames_->Wait(kDefaultTimeoutMs))
2057 << "No frames received by the encoder.";
2058 EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs))
2059 << "Timed out waiting for send-side packets.";
2060 EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
2061 << "Timed out waiting for receiver-side packets.";
2062
2063 // Sender-side network down.
2064 sender_call_->SignalNetworkState(Call::kNetworkDown);
2065 {
2066 CriticalSectionScoped lock(test_crit_.get());
2067 sender_packets_->Reset(); // Earlier packets should not count.
2068 sender_state_ = Call::kNetworkDown;
2069 }
2070 EXPECT_EQ(kEventTimeout, sender_packets_->Wait(kSilenceTimeoutMs))
2071 << "Packets sent during sender-network downtime.";
2072 EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
2073 << "Timed out waiting for receiver-side packets.";
2074 // Receiver-side network down.
2075 receiver_call_->SignalNetworkState(Call::kNetworkDown);
2076 {
2077 CriticalSectionScoped lock(test_crit_.get());
2078 receiver_packets_->Reset(); // Earlier packets should not count.
2079 receiver_state_ = Call::kNetworkDown;
2080 }
2081 EXPECT_EQ(kEventTimeout, receiver_packets_->Wait(kSilenceTimeoutMs))
2082 << "Packets sent during receiver-network downtime.";
2083
2084 // Network back up again for both.
2085 {
2086 CriticalSectionScoped lock(test_crit_.get());
2087 sender_packets_->Reset(); // Earlier packets should not count.
2088 receiver_packets_->Reset(); // Earlier packets should not count.
2089 sender_state_ = receiver_state_ = Call::kNetworkUp;
2090 }
2091 sender_call_->SignalNetworkState(Call::kNetworkUp);
2092 receiver_call_->SignalNetworkState(Call::kNetworkUp);
2093 EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs))
2094 << "Timed out waiting for send-side packets.";
2095 EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
2096 << "Timed out waiting for receiver-side packets.";
2097 }
2098
2099 virtual int32_t Encode(const I420VideoFrame& input_image,
2100 const CodecSpecificInfo* codec_specific_info,
2101 const std::vector<VideoFrameType>* frame_types)
2102 OVERRIDE {
2103 {
2104 CriticalSectionScoped lock(test_crit_.get());
2105 if (sender_state_ == Call::kNetworkDown) {
2106 ++down_frames_;
2107 EXPECT_LE(down_frames_, 1)
2108 << "Encoding more than one frame while network is down.";
2109 if (down_frames_ > 1)
2110 encoded_frames_->Set();
2111 } else {
2112 encoded_frames_->Set();
2113 }
2114 }
2115 return test::FakeEncoder::Encode(
2116 input_image, codec_specific_info, frame_types);
2117 }
2118
2119 private:
2120 const scoped_ptr<CriticalSectionWrapper> test_crit_;
2121 scoped_ptr<EventWrapper> encoded_frames_;
2122 scoped_ptr<EventWrapper> sender_packets_;
2123 scoped_ptr<EventWrapper> receiver_packets_;
2124 Call* sender_call_;
2125 Call* receiver_call_;
2126 Call::NetworkState sender_state_ GUARDED_BY(test_crit_);
2127 int down_sender_rtp_ GUARDED_BY(test_crit_);
2128 int down_sender_rtcp_ GUARDED_BY(test_crit_);
2129 Call::NetworkState receiver_state_ GUARDED_BY(test_crit_);
2130 int down_receiver_rtcp_ GUARDED_BY(test_crit_);
2131 int down_frames_ GUARDED_BY(test_crit_);
2132 } test;
2133
2134 RunBaseTest(&test);
2135}
2136
2137TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
2138 class UnusedEncoder : public test::FakeEncoder {
2139 public:
2140 UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
2141 virtual int32_t Encode(const I420VideoFrame& input_image,
2142 const CodecSpecificInfo* codec_specific_info,
2143 const std::vector<VideoFrameType>* frame_types)
2144 OVERRIDE {
2145 ADD_FAILURE() << "Unexpected frame encode.";
2146 return test::FakeEncoder::Encode(
2147 input_image, codec_specific_info, frame_types);
2148 }
2149 };
2150
2151 UnusedTransport transport;
2152 CreateSenderCall(Call::Config(&transport));
2153 sender_call_->SignalNetworkState(Call::kNetworkDown);
2154
2155 CreateSendConfig(1);
2156 UnusedEncoder unused_encoder;
2157 send_config_.encoder_settings.encoder = &unused_encoder;
2158 CreateStreams();
2159 CreateFrameGeneratorCapturer();
2160
2161 Start();
2162 SleepMs(kSilenceTimeoutMs);
2163 Stop();
2164
2165 DestroyStreams();
2166}
2167
2168TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {
2169 test::DirectTransport sender_transport;
2170 CreateSenderCall(Call::Config(&sender_transport));
2171 UnusedTransport transport;
2172 CreateReceiverCall(Call::Config(&transport));
2173 sender_transport.SetReceiver(receiver_call_->Receiver());
2174
2175 receiver_call_->SignalNetworkState(Call::kNetworkDown);
2176
2177 CreateSendConfig(1);
2178 CreateMatchingReceiveConfigs();
2179 CreateStreams();
2180 CreateFrameGeneratorCapturer();
2181
2182 Start();
2183 SleepMs(kSilenceTimeoutMs);
2184 Stop();
2185
2186 sender_transport.StopSending();
2187
2188 DestroyStreams();
2189}
pbos@webrtc.org09cc6862014-11-04 13:48:15 +00002190
2191// TODO(pbos): Remove this regression test when VideoEngine is no longer used as
2192// a backend. This is to test that we hand channels back properly.
2193TEST_F(EndToEndTest, CanCreateAndDestroyManyVideoStreams) {
2194 test::NullTransport transport;
2195 scoped_ptr<Call> call(Call::Create(Call::Config(&transport)));
2196 test::FakeDecoder fake_decoder;
2197 test::FakeEncoder fake_encoder(Clock::GetRealTimeClock());
2198 for (size_t i = 0; i < 100; ++i) {
2199 VideoSendStream::Config send_config;
2200 send_config.encoder_settings.encoder = &fake_encoder;
2201 send_config.encoder_settings.payload_name = "FAKE";
2202 send_config.encoder_settings.payload_type = 123;
2203
2204 VideoEncoderConfig encoder_config;
2205 encoder_config.streams = test::CreateVideoStreams(1);
2206 send_config.rtp.ssrcs.push_back(1);
2207 VideoSendStream* send_stream =
2208 call->CreateVideoSendStream(send_config, encoder_config);
2209 call->DestroyVideoSendStream(send_stream);
2210
2211 VideoReceiveStream::Config receive_config;
2212 receive_config.rtp.remote_ssrc = 1;
2213 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
2214 VideoReceiveStream::Decoder decoder;
2215 decoder.decoder = &fake_decoder;
2216 decoder.payload_type = 123;
2217 decoder.payload_name = "FAKE";
2218 receive_config.decoders.push_back(decoder);
2219 VideoReceiveStream* receive_stream =
2220 call->CreateVideoReceiveStream(receive_config);
2221 call->DestroyVideoReceiveStream(receive_stream);
2222 }
2223}
2224
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00002225} // namespace webrtc