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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"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000022#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
23#include "webrtc/system_wrappers/interface/event_wrapper.h"
24#include "webrtc/system_wrappers/interface/scoped_ptr.h"
25#include "webrtc/system_wrappers/interface/sleep.h"
26#include "webrtc/test/call_test.h"
27#include "webrtc/test/direct_transport.h"
28#include "webrtc/test/encoder_settings.h"
29#include "webrtc/test/fake_audio_device.h"
30#include "webrtc/test/fake_decoder.h"
31#include "webrtc/test/fake_encoder.h"
32#include "webrtc/test/frame_generator.h"
33#include "webrtc/test/frame_generator_capturer.h"
34#include "webrtc/test/null_transport.h"
35#include "webrtc/test/rtp_rtcp_observer.h"
36#include "webrtc/test/testsupport/fileutils.h"
andresp@webrtc.orgab071da2014-09-18 08:58:15 +000037#include "webrtc/test/testsupport/gtest_disable.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000038#include "webrtc/test/testsupport/perf_test.h"
39#include "webrtc/video/transport_adapter.h"
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +000040#include "webrtc/video_encoder.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000041
42namespace webrtc {
43
pbos@webrtc.org26c0c412014-09-03 16:17:12 +000044static const unsigned long kSilenceTimeoutMs = 2000;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000045
46class EndToEndTest : public test::CallTest {
47 public:
48 EndToEndTest() {}
49
50 virtual ~EndToEndTest() {
51 EXPECT_EQ(NULL, send_stream_);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +000052 EXPECT_TRUE(receive_streams_.empty());
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000053 }
54
55 protected:
pbos@webrtc.org26c0c412014-09-03 16:17:12 +000056 class UnusedTransport : public newapi::Transport {
57 private:
58 virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE {
59 ADD_FAILURE() << "Unexpected RTP sent.";
60 return false;
61 }
62
63 virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
64 ADD_FAILURE() << "Unexpected RTCP sent.";
65 return false;
66 }
67 };
68
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000069 void DecodesRetransmittedFrame(bool retransmit_over_rtx);
70 void ReceivesPliAndRecovers(int rtp_history_ms);
71 void RespectsRtcpMode(newapi::RtcpMode rtcp_mode);
72 void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +000073 void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000074 void TestRtpStatePreservation(bool use_rtx);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000075};
76
77TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
78 test::NullTransport transport;
79 CreateCalls(Call::Config(&transport), Call::Config(&transport));
80
81 CreateSendConfig(1);
82 CreateMatchingReceiveConfigs();
83
84 CreateStreams();
85
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +000086 receive_streams_[0]->Start();
87 receive_streams_[0]->Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000088
89 DestroyStreams();
90}
91
92TEST_F(EndToEndTest, ReceiverCanBeStoppedTwice) {
93 test::NullTransport transport;
94 CreateCalls(Call::Config(&transport), Call::Config(&transport));
95
96 CreateSendConfig(1);
97 CreateMatchingReceiveConfigs();
98
99 CreateStreams();
100
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000101 receive_streams_[0]->Stop();
102 receive_streams_[0]->Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000103
104 DestroyStreams();
105}
106
107TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
108 static const int kWidth = 320;
109 static const int kHeight = 240;
110 // This constant is chosen to be higher than the timeout in the video_render
111 // module. This makes sure that frames aren't dropped if there are no other
112 // frames in the queue.
113 static const int kDelayRenderCallbackMs = 1000;
114
115 class Renderer : public VideoRenderer {
116 public:
117 Renderer() : event_(EventWrapper::Create()) {}
118
119 virtual void RenderFrame(const I420VideoFrame& video_frame,
120 int /*time_to_render_ms*/) OVERRIDE {
121 event_->Set();
122 }
123
124 EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
125
126 scoped_ptr<EventWrapper> event_;
127 } renderer;
128
129 class TestFrameCallback : public I420FrameCallback {
130 public:
131 TestFrameCallback() : event_(EventWrapper::Create()) {}
132
133 EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
134
135 private:
136 virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE {
137 SleepMs(kDelayRenderCallbackMs);
138 event_->Set();
139 }
140
141 scoped_ptr<EventWrapper> event_;
142 };
143
144 test::DirectTransport sender_transport, receiver_transport;
145
146 CreateCalls(Call::Config(&sender_transport),
147 Call::Config(&receiver_transport));
148
149 sender_transport.SetReceiver(receiver_call_->Receiver());
150 receiver_transport.SetReceiver(sender_call_->Receiver());
151
152 CreateSendConfig(1);
153 CreateMatchingReceiveConfigs();
154
155 TestFrameCallback pre_render_callback;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000156 receive_configs_[0].pre_render_callback = &pre_render_callback;
157 receive_configs_[0].renderer = &renderer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000158
159 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000160 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000161
162 // Create frames that are smaller than the send width/height, this is done to
163 // check that the callbacks are done after processing video.
164 scoped_ptr<test::FrameGenerator> frame_generator(
165 test::FrameGenerator::Create(kWidth, kHeight));
166 send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
167 EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
168 << "Timed out while waiting for pre-render callback.";
169 EXPECT_EQ(kEventSignaled, renderer.Wait())
170 << "Timed out while waiting for the frame to render.";
171
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000172 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000173
174 sender_transport.StopSending();
175 receiver_transport.StopSending();
176
177 DestroyStreams();
178}
179
180TEST_F(EndToEndTest, TransmitsFirstFrame) {
181 class Renderer : public VideoRenderer {
182 public:
183 Renderer() : event_(EventWrapper::Create()) {}
184
185 virtual void RenderFrame(const I420VideoFrame& video_frame,
186 int /*time_to_render_ms*/) OVERRIDE {
187 event_->Set();
188 }
189
190 EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
191
192 scoped_ptr<EventWrapper> event_;
193 } renderer;
194
195 test::DirectTransport sender_transport, receiver_transport;
196
197 CreateCalls(Call::Config(&sender_transport),
198 Call::Config(&receiver_transport));
199
200 sender_transport.SetReceiver(receiver_call_->Receiver());
201 receiver_transport.SetReceiver(sender_call_->Receiver());
202
203 CreateSendConfig(1);
204 CreateMatchingReceiveConfigs();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000205 receive_configs_[0].renderer = &renderer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000206
207 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000208 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000209
210 scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create(
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000211 encoder_config_.streams[0].width, encoder_config_.streams[0].height));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000212 send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
213
214 EXPECT_EQ(kEventSignaled, renderer.Wait())
215 << "Timed out while waiting for the frame to render.";
216
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000217 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000218
219 sender_transport.StopSending();
220 receiver_transport.StopSending();
221
222 DestroyStreams();
223}
224
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000225TEST_F(EndToEndTest, SendsAndReceivesH264) {
226 class H264Observer : public test::EndToEndTest, public VideoRenderer {
227 public:
228 H264Observer()
229 : EndToEndTest(2 * kDefaultTimeoutMs),
230 fake_encoder_(Clock::GetRealTimeClock()),
231 frame_counter_(0) {}
232
233 virtual void PerformTest() OVERRIDE {
234 EXPECT_EQ(kEventSignaled, Wait())
235 << "Timed out while waiting for enough frames to be decoded.";
236 }
237
238 virtual void ModifyConfigs(
239 VideoSendStream::Config* send_config,
240 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000241 VideoEncoderConfig* encoder_config) OVERRIDE {
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000242 send_config->encoder_settings.encoder = &fake_encoder_;
243 send_config->encoder_settings.payload_name = "H264";
244 send_config->encoder_settings.payload_type = kFakeSendPayloadType;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000245 encoder_config->streams[0].min_bitrate_bps = 50000;
246 encoder_config->streams[0].target_bitrate_bps =
247 encoder_config->streams[0].max_bitrate_bps = 2000000;
stefan@webrtc.org79c33592014-08-06 09:24:53 +0000248
249 (*receive_configs)[0].renderer = this;
250 VideoCodec codec =
251 test::CreateDecoderVideoCodec(send_config->encoder_settings);
252 (*receive_configs)[0].codecs.resize(1);
253 (*receive_configs)[0].codecs[0] = codec;
254 (*receive_configs)[0].external_decoders.resize(1);
255 (*receive_configs)[0].external_decoders[0].payload_type =
256 send_config->encoder_settings.payload_type;
257 (*receive_configs)[0].external_decoders[0].decoder = &fake_decoder_;
258 }
259
260 virtual void RenderFrame(const I420VideoFrame& video_frame,
261 int time_to_render_ms) OVERRIDE {
262 const int kRequiredFrames = 500;
263 if (++frame_counter_ == kRequiredFrames)
264 observation_complete_->Set();
265 }
266
267 private:
268 test::FakeH264Decoder fake_decoder_;
269 test::FakeH264Encoder fake_encoder_;
270 int frame_counter_;
271 } test;
272
273 RunBaseTest(&test);
274}
275
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000276TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
277 class SyncRtcpObserver : public test::EndToEndTest {
278 public:
279 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
280
281 virtual Action OnReceiveRtcp(const uint8_t* packet,
282 size_t length) OVERRIDE {
283 RTCPUtility::RTCPParserV2 parser(packet, length, true);
284 EXPECT_TRUE(parser.IsValid());
285 uint32_t ssrc = 0;
286 ssrc |= static_cast<uint32_t>(packet[4]) << 24;
287 ssrc |= static_cast<uint32_t>(packet[5]) << 16;
288 ssrc |= static_cast<uint32_t>(packet[6]) << 8;
289 ssrc |= static_cast<uint32_t>(packet[7]) << 0;
290 EXPECT_EQ(kReceiverLocalSsrc, ssrc);
291 observation_complete_->Set();
292
293 return SEND_PACKET;
294 }
295
296 virtual void PerformTest() OVERRIDE {
297 EXPECT_EQ(kEventSignaled, Wait())
298 << "Timed out while waiting for a receiver RTCP packet to be sent.";
299 }
300 } test;
301
302 RunBaseTest(&test);
303}
304
305TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
306 static const int kNumberOfNacksToObserve = 2;
307 static const int kLossBurstSize = 2;
308 static const int kPacketsBetweenLossBursts = 9;
309 class NackObserver : public test::EndToEndTest {
310 public:
311 NackObserver()
312 : EndToEndTest(kLongTimeoutMs),
313 rtp_parser_(RtpHeaderParser::Create()),
314 sent_rtp_packets_(0),
315 packets_left_to_drop_(0),
316 nacks_left_(kNumberOfNacksToObserve) {}
317
318 private:
319 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
320 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000321 EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000322
323 // Never drop retransmitted packets.
324 if (dropped_packets_.find(header.sequenceNumber) !=
325 dropped_packets_.end()) {
326 retransmitted_packets_.insert(header.sequenceNumber);
327 if (nacks_left_ == 0 &&
328 retransmitted_packets_.size() == dropped_packets_.size()) {
329 observation_complete_->Set();
330 }
331 return SEND_PACKET;
332 }
333
334 ++sent_rtp_packets_;
335
336 // Enough NACKs received, stop dropping packets.
337 if (nacks_left_ == 0)
338 return SEND_PACKET;
339
340 // Check if it's time for a new loss burst.
341 if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
342 packets_left_to_drop_ = kLossBurstSize;
343
344 if (packets_left_to_drop_ > 0) {
345 --packets_left_to_drop_;
346 dropped_packets_.insert(header.sequenceNumber);
347 return DROP_PACKET;
348 }
349
350 return SEND_PACKET;
351 }
352
353 virtual Action OnReceiveRtcp(const uint8_t* packet,
354 size_t length) OVERRIDE {
355 RTCPUtility::RTCPParserV2 parser(packet, length, true);
356 EXPECT_TRUE(parser.IsValid());
357
358 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
359 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
360 if (packet_type == RTCPUtility::kRtcpRtpfbNackCode) {
361 --nacks_left_;
362 break;
363 }
364 packet_type = parser.Iterate();
365 }
366 return SEND_PACKET;
367 }
368
369 virtual void ModifyConfigs(
370 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000371 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000372 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000373 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000374 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000375 }
376
377 virtual void PerformTest() OVERRIDE {
378 EXPECT_EQ(kEventSignaled, Wait())
379 << "Timed out waiting for packets to be NACKed, retransmitted and "
380 "rendered.";
381 }
382
383 scoped_ptr<RtpHeaderParser> rtp_parser_;
384 std::set<uint16_t> dropped_packets_;
385 std::set<uint16_t> retransmitted_packets_;
386 uint64_t sent_rtp_packets_;
387 int packets_left_to_drop_;
388 int nacks_left_;
389 } test;
390
391 RunBaseTest(&test);
392}
393
394// TODO(pbos): Flaky, webrtc:3269
395TEST_F(EndToEndTest, DISABLED_CanReceiveFec) {
396 class FecRenderObserver : public test::EndToEndTest, public VideoRenderer {
397 public:
398 FecRenderObserver()
399 : EndToEndTest(kDefaultTimeoutMs),
400 state_(kFirstPacket),
401 protected_sequence_number_(0),
402 protected_frame_timestamp_(0) {}
403
404 private:
405 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE
406 EXCLUSIVE_LOCKS_REQUIRED(crit_) {
407 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000408 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000409
410 EXPECT_EQ(kRedPayloadType, header.payloadType);
411 int encapsulated_payload_type =
412 static_cast<int>(packet[header.headerLength]);
413 if (encapsulated_payload_type != kFakeSendPayloadType)
414 EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
415
416 switch (state_) {
417 case kFirstPacket:
418 state_ = kDropEveryOtherPacketUntilFec;
419 break;
420 case kDropEveryOtherPacketUntilFec:
421 if (encapsulated_payload_type == kUlpfecPayloadType) {
422 state_ = kDropNextMediaPacket;
423 return SEND_PACKET;
424 }
425 if (header.sequenceNumber % 2 == 0)
426 return DROP_PACKET;
427 break;
428 case kDropNextMediaPacket:
429 if (encapsulated_payload_type == kFakeSendPayloadType) {
430 protected_sequence_number_ = header.sequenceNumber;
431 protected_frame_timestamp_ = header.timestamp;
432 state_ = kProtectedPacketDropped;
433 return DROP_PACKET;
434 }
435 break;
436 case kProtectedPacketDropped:
437 EXPECT_NE(header.sequenceNumber, protected_sequence_number_)
438 << "Protected packet retransmitted. Should not happen with FEC.";
439 break;
440 }
441
442 return SEND_PACKET;
443 }
444
445 virtual void RenderFrame(const I420VideoFrame& video_frame,
446 int time_to_render_ms) OVERRIDE {
447 CriticalSectionScoped lock(crit_.get());
448 // Rendering frame with timestamp associated with dropped packet -> FEC
449 // protection worked.
450 if (state_ == kProtectedPacketDropped &&
451 video_frame.timestamp() == protected_frame_timestamp_) {
452 observation_complete_->Set();
453 }
454 }
455
456 enum {
457 kFirstPacket,
458 kDropEveryOtherPacketUntilFec,
459 kDropNextMediaPacket,
460 kProtectedPacketDropped,
461 } state_;
462
463 virtual void ModifyConfigs(
464 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000465 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000466 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000467 // TODO(pbos): Run this test with combined NACK/FEC enabled as well.
468 // int rtp_history_ms = 1000;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000469 // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000470 // send_config->rtp.nack.rtp_history_ms = rtp_history_ms;
471 send_config->rtp.fec.red_payload_type = kRedPayloadType;
472 send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
473
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000474 (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
475 (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
476 (*receive_configs)[0].renderer = this;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000477 }
478
479 virtual void PerformTest() OVERRIDE {
480 EXPECT_EQ(kEventSignaled, Wait())
481 << "Timed out while waiting for retransmitted NACKed frames to be "
482 "rendered again.";
483 }
484
485 uint32_t protected_sequence_number_ GUARDED_BY(crit_);
486 uint32_t protected_frame_timestamp_ GUARDED_BY(crit_);
487 } test;
488
489 RunBaseTest(&test);
490}
491
492// This test drops second RTP packet with a marker bit set, makes sure it's
493// retransmitted and renders. Retransmission SSRCs are also checked.
494void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
495 static const int kDroppedFrameNumber = 2;
496 class RetransmissionObserver : public test::EndToEndTest,
497 public I420FrameCallback {
498 public:
499 explicit RetransmissionObserver(bool expect_rtx)
500 : EndToEndTest(kDefaultTimeoutMs),
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000501 retransmission_ssrc_(expect_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000502 retransmission_payload_type_(expect_rtx ? kSendRtxPayloadType
503 : kFakeSendPayloadType),
504 marker_bits_observed_(0),
505 retransmitted_timestamp_(0),
506 frame_retransmitted_(false) {}
507
508 private:
509 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
510 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000511 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000512
513 if (header.timestamp == retransmitted_timestamp_) {
514 EXPECT_EQ(retransmission_ssrc_, header.ssrc);
515 EXPECT_EQ(retransmission_payload_type_, header.payloadType);
516 frame_retransmitted_ = true;
517 return SEND_PACKET;
518 }
519
520 EXPECT_EQ(kSendSsrcs[0], header.ssrc);
521 EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
522
523 // Found the second frame's final packet, drop this and expect a
524 // retransmission.
525 if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) {
526 retransmitted_timestamp_ = header.timestamp;
527 return DROP_PACKET;
528 }
529
530 return SEND_PACKET;
531 }
532
533 virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE {
534 CriticalSectionScoped lock(crit_.get());
535 if (frame->timestamp() == retransmitted_timestamp_) {
536 EXPECT_TRUE(frame_retransmitted_);
537 observation_complete_->Set();
538 }
539 }
540
541 virtual void ModifyConfigs(
542 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000543 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000544 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000545 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000546 (*receive_configs)[0].pre_render_callback = this;
547 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000548 if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
549 send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000550 send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000551 (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc =
552 kSendRtxSsrcs[0];
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000553 (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].payload_type =
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000554 kSendRtxPayloadType;
555 }
556 }
557
558 virtual void PerformTest() OVERRIDE {
559 EXPECT_EQ(kEventSignaled, Wait())
560 << "Timed out while waiting for retransmission to render.";
561 }
562
563 const uint32_t retransmission_ssrc_;
564 const int retransmission_payload_type_;
565 int marker_bits_observed_;
566 uint32_t retransmitted_timestamp_;
567 bool frame_retransmitted_;
568 } test(retransmit_over_rtx);
569
570 RunBaseTest(&test);
571}
572
573TEST_F(EndToEndTest, DecodesRetransmittedFrame) {
574 DecodesRetransmittedFrame(false);
575}
576
577TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
578 DecodesRetransmittedFrame(true);
579}
580
andresp@webrtc.org02686112014-09-19 08:24:19 +0000581TEST_F(EndToEndTest, UsesFrameCallbacks) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000582 static const int kWidth = 320;
583 static const int kHeight = 240;
584
585 class Renderer : public VideoRenderer {
586 public:
587 Renderer() : event_(EventWrapper::Create()) {}
588
589 virtual void RenderFrame(const I420VideoFrame& video_frame,
590 int /*time_to_render_ms*/) OVERRIDE {
591 EXPECT_EQ(0, *video_frame.buffer(kYPlane))
592 << "Rendered frame should have zero luma which is applied by the "
593 "pre-render callback.";
594 event_->Set();
595 }
596
597 EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
598 scoped_ptr<EventWrapper> event_;
599 } renderer;
600
601 class TestFrameCallback : public I420FrameCallback {
602 public:
603 TestFrameCallback(int expected_luma_byte, int next_luma_byte)
604 : event_(EventWrapper::Create()),
605 expected_luma_byte_(expected_luma_byte),
606 next_luma_byte_(next_luma_byte) {}
607
608 EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
609
610 private:
611 virtual void FrameCallback(I420VideoFrame* frame) {
612 EXPECT_EQ(kWidth, frame->width())
613 << "Width not as expected, callback done before resize?";
614 EXPECT_EQ(kHeight, frame->height())
615 << "Height not as expected, callback done before resize?";
616
617 // Previous luma specified, observed luma should be fairly close.
618 if (expected_luma_byte_ != -1) {
619 EXPECT_NEAR(expected_luma_byte_, *frame->buffer(kYPlane), 10);
620 }
621
622 memset(frame->buffer(kYPlane),
623 next_luma_byte_,
624 frame->allocated_size(kYPlane));
625
626 event_->Set();
627 }
628
629 scoped_ptr<EventWrapper> event_;
630 int expected_luma_byte_;
631 int next_luma_byte_;
632 };
633
634 TestFrameCallback pre_encode_callback(-1, 255); // Changes luma to 255.
635 TestFrameCallback pre_render_callback(255, 0); // Changes luma from 255 to 0.
636
637 test::DirectTransport sender_transport, receiver_transport;
638
639 CreateCalls(Call::Config(&sender_transport),
640 Call::Config(&receiver_transport));
641
642 sender_transport.SetReceiver(receiver_call_->Receiver());
643 receiver_transport.SetReceiver(sender_call_->Receiver());
644
645 CreateSendConfig(1);
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +0000646 scoped_ptr<VideoEncoder> encoder(
647 VideoEncoder::Create(VideoEncoder::kVp8));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000648 send_config_.encoder_settings.encoder = encoder.get();
649 send_config_.encoder_settings.payload_name = "VP8";
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000650 ASSERT_EQ(1u, encoder_config_.streams.size()) << "Test setup error.";
651 encoder_config_.streams[0].width = kWidth;
652 encoder_config_.streams[0].height = kHeight;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000653 send_config_.pre_encode_callback = &pre_encode_callback;
654
655 CreateMatchingReceiveConfigs();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000656 receive_configs_[0].pre_render_callback = &pre_render_callback;
657 receive_configs_[0].renderer = &renderer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000658
659 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000660 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000661
662 // Create frames that are smaller than the send width/height, this is done to
663 // check that the callbacks are done after processing video.
664 scoped_ptr<test::FrameGenerator> frame_generator(
665 test::FrameGenerator::Create(kWidth / 2, kHeight / 2));
666 send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
667
668 EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait())
669 << "Timed out while waiting for pre-encode callback.";
670 EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
671 << "Timed out while waiting for pre-render callback.";
672 EXPECT_EQ(kEventSignaled, renderer.Wait())
673 << "Timed out while waiting for the frame to render.";
674
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000675 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000676
677 sender_transport.StopSending();
678 receiver_transport.StopSending();
679
680 DestroyStreams();
681}
682
683void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
684 static const int kPacketsToDrop = 1;
685
686 class PliObserver : public test::EndToEndTest, public VideoRenderer {
687 public:
688 explicit PliObserver(int rtp_history_ms)
689 : EndToEndTest(kLongTimeoutMs),
690 rtp_history_ms_(rtp_history_ms),
691 nack_enabled_(rtp_history_ms > 0),
692 highest_dropped_timestamp_(0),
693 frames_to_drop_(0),
694 received_pli_(false) {}
695
696 private:
697 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
698 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000699 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000700
701 // Drop all retransmitted packets to force a PLI.
702 if (header.timestamp <= highest_dropped_timestamp_)
703 return DROP_PACKET;
704
705 if (frames_to_drop_ > 0) {
706 highest_dropped_timestamp_ = header.timestamp;
707 --frames_to_drop_;
708 return DROP_PACKET;
709 }
710
711 return SEND_PACKET;
712 }
713
714 virtual Action OnReceiveRtcp(const uint8_t* packet,
715 size_t length) OVERRIDE {
716 RTCPUtility::RTCPParserV2 parser(packet, length, true);
717 EXPECT_TRUE(parser.IsValid());
718
719 for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
720 packet_type != RTCPUtility::kRtcpNotValidCode;
721 packet_type = parser.Iterate()) {
722 if (!nack_enabled_)
723 EXPECT_NE(packet_type, RTCPUtility::kRtcpRtpfbNackCode);
724
725 if (packet_type == RTCPUtility::kRtcpPsfbPliCode) {
726 received_pli_ = true;
727 break;
728 }
729 }
730 return SEND_PACKET;
731 }
732
733 virtual void RenderFrame(const I420VideoFrame& video_frame,
734 int time_to_render_ms) OVERRIDE {
735 CriticalSectionScoped lock(crit_.get());
736 if (received_pli_ &&
737 video_frame.timestamp() > highest_dropped_timestamp_) {
738 observation_complete_->Set();
739 }
740 if (!received_pli_)
741 frames_to_drop_ = kPacketsToDrop;
742 }
743
744 virtual void ModifyConfigs(
745 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000746 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000747 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000748 send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000749 (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
750 (*receive_configs)[0].renderer = this;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000751 }
752
753 virtual void PerformTest() OVERRIDE {
754 EXPECT_EQ(kEventSignaled, Wait()) << "Timed out waiting for PLI to be "
755 "received and a frame to be "
756 "rendered afterwards.";
757 }
758
759 int rtp_history_ms_;
760 bool nack_enabled_;
761 uint32_t highest_dropped_timestamp_;
762 int frames_to_drop_;
763 bool received_pli_;
764 } test(rtp_history_ms);
765
766 RunBaseTest(&test);
767}
768
769TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) {
770 ReceivesPliAndRecovers(1000);
771}
772
773// TODO(pbos): Enable this when 2250 is resolved.
774TEST_F(EndToEndTest, DISABLED_ReceivesPliAndRecoversWithoutNack) {
775 ReceivesPliAndRecovers(0);
776}
777
778TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
779 class PacketInputObserver : public PacketReceiver {
780 public:
781 explicit PacketInputObserver(PacketReceiver* receiver)
782 : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {}
783
784 EventTypeWrapper Wait() {
785 return delivered_packet_->Wait(kDefaultTimeoutMs);
786 }
787
788 private:
789 virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
790 size_t length) OVERRIDE {
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000791 if (RtpHeaderParser::IsRtcp(packet, length)) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000792 return receiver_->DeliverPacket(packet, length);
793 } else {
794 DeliveryStatus delivery_status =
795 receiver_->DeliverPacket(packet, length);
796 EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
797 delivered_packet_->Set();
798 return delivery_status;
799 }
800 }
801
802 PacketReceiver* receiver_;
803 scoped_ptr<EventWrapper> delivered_packet_;
804 };
805
806 test::DirectTransport send_transport, receive_transport;
807
808 CreateCalls(Call::Config(&send_transport), Call::Config(&receive_transport));
809 PacketInputObserver input_observer(receiver_call_->Receiver());
810
811 send_transport.SetReceiver(&input_observer);
812 receive_transport.SetReceiver(sender_call_->Receiver());
813
814 CreateSendConfig(1);
815 CreateMatchingReceiveConfigs();
816
817 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000818 CreateFrameGeneratorCapturer();
819 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000820
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000821 receiver_call_->DestroyVideoReceiveStream(receive_streams_[0]);
822 receive_streams_.clear();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000823
824 // Wait() waits for a received packet.
825 EXPECT_EQ(kEventSignaled, input_observer.Wait());
826
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000827 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000828
829 DestroyStreams();
830
831 send_transport.StopSending();
832 receive_transport.StopSending();
833}
834
835void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
836 static const int kNumCompoundRtcpPacketsToObserve = 10;
837 class RtcpModeObserver : public test::EndToEndTest {
838 public:
839 explicit RtcpModeObserver(newapi::RtcpMode rtcp_mode)
840 : EndToEndTest(kDefaultTimeoutMs),
841 rtcp_mode_(rtcp_mode),
842 sent_rtp_(0),
843 sent_rtcp_(0) {}
844
845 private:
846 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
847 if (++sent_rtp_ % 3 == 0)
848 return DROP_PACKET;
849
850 return SEND_PACKET;
851 }
852
853 virtual Action OnReceiveRtcp(const uint8_t* packet,
854 size_t length) OVERRIDE {
855 ++sent_rtcp_;
856 RTCPUtility::RTCPParserV2 parser(packet, length, true);
857 EXPECT_TRUE(parser.IsValid());
858
859 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
860 bool has_report_block = false;
861 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
862 EXPECT_NE(RTCPUtility::kRtcpSrCode, packet_type);
863 if (packet_type == RTCPUtility::kRtcpRrCode) {
864 has_report_block = true;
865 break;
866 }
867 packet_type = parser.Iterate();
868 }
869
870 switch (rtcp_mode_) {
871 case newapi::kRtcpCompound:
872 if (!has_report_block) {
873 ADD_FAILURE() << "Received RTCP packet without receiver report for "
874 "kRtcpCompound.";
875 observation_complete_->Set();
876 }
877
878 if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
879 observation_complete_->Set();
880
881 break;
882 case newapi::kRtcpReducedSize:
883 if (!has_report_block)
884 observation_complete_->Set();
885 break;
886 }
887
888 return SEND_PACKET;
889 }
890
891 virtual void ModifyConfigs(
892 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000893 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000894 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000895 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000896 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
897 (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000898 }
899
900 virtual void PerformTest() OVERRIDE {
901 EXPECT_EQ(kEventSignaled, Wait())
902 << (rtcp_mode_ == newapi::kRtcpCompound
903 ? "Timed out before observing enough compound packets."
904 : "Timed out before receiving a non-compound RTCP packet.");
905 }
906
907 newapi::RtcpMode rtcp_mode_;
908 int sent_rtp_;
909 int sent_rtcp_;
910 } test(rtcp_mode);
911
912 RunBaseTest(&test);
913}
914
915TEST_F(EndToEndTest, UsesRtcpCompoundMode) {
916 RespectsRtcpMode(newapi::kRtcpCompound);
917}
918
919TEST_F(EndToEndTest, UsesRtcpReducedSizeMode) {
920 RespectsRtcpMode(newapi::kRtcpReducedSize);
921}
922
923// Test sets up a Call multiple senders with different resolutions and SSRCs.
924// Another is set up to receive all three of these with different renderers.
925// Each renderer verifies that it receives the expected resolution, and as soon
926// as every renderer has received a frame, the test finishes.
andresp@webrtc.org02686112014-09-19 08:24:19 +0000927TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000928 static const size_t kNumStreams = 3;
929
930 class VideoOutputObserver : public VideoRenderer {
931 public:
932 VideoOutputObserver(test::FrameGeneratorCapturer** capturer,
933 int width,
934 int height)
935 : capturer_(capturer),
936 width_(width),
937 height_(height),
938 done_(EventWrapper::Create()) {}
939
940 virtual void RenderFrame(const I420VideoFrame& video_frame,
941 int time_to_render_ms) OVERRIDE {
942 EXPECT_EQ(width_, video_frame.width());
943 EXPECT_EQ(height_, video_frame.height());
944 (*capturer_)->Stop();
945 done_->Set();
946 }
947
948 EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
949
950 private:
951 test::FrameGeneratorCapturer** capturer_;
952 int width_;
953 int height_;
954 scoped_ptr<EventWrapper> done_;
955 };
956
957 struct {
958 uint32_t ssrc;
959 int width;
960 int height;
961 } codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}};
962
963 test::DirectTransport sender_transport, receiver_transport;
964 scoped_ptr<Call> sender_call(Call::Create(Call::Config(&sender_transport)));
965 scoped_ptr<Call> receiver_call(
966 Call::Create(Call::Config(&receiver_transport)));
967 sender_transport.SetReceiver(receiver_call->Receiver());
968 receiver_transport.SetReceiver(sender_call->Receiver());
969
970 VideoSendStream* send_streams[kNumStreams];
971 VideoReceiveStream* receive_streams[kNumStreams];
972
973 VideoOutputObserver* observers[kNumStreams];
974 test::FrameGeneratorCapturer* frame_generators[kNumStreams];
975
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +0000976 scoped_ptr<VideoEncoder> encoders[kNumStreams];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000977 for (size_t i = 0; i < kNumStreams; ++i)
pbos@webrtc.orgab990ae2014-09-17 09:02:25 +0000978 encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000979
980 for (size_t i = 0; i < kNumStreams; ++i) {
981 uint32_t ssrc = codec_settings[i].ssrc;
982 int width = codec_settings[i].width;
983 int height = codec_settings[i].height;
984 observers[i] = new VideoOutputObserver(&frame_generators[i], width, height);
985
pbos@webrtc.orgbd249bc2014-07-07 04:45:15 +0000986 VideoSendStream::Config send_config;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000987 send_config.rtp.ssrcs.push_back(ssrc);
988 send_config.encoder_settings.encoder = encoders[i].get();
989 send_config.encoder_settings.payload_name = "VP8";
990 send_config.encoder_settings.payload_type = 124;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000991 VideoEncoderConfig encoder_config;
992 encoder_config.streams = test::CreateVideoStreams(1);
993 VideoStream* stream = &encoder_config.streams[0];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000994 stream->width = width;
995 stream->height = height;
996 stream->max_framerate = 5;
997 stream->min_bitrate_bps = stream->target_bitrate_bps =
998 stream->max_bitrate_bps = 100000;
999 send_streams[i] =
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001000 sender_call->CreateVideoSendStream(send_config, encoder_config);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001001 send_streams[i]->Start();
1002
pbos@webrtc.orgbd249bc2014-07-07 04:45:15 +00001003 VideoReceiveStream::Config receive_config;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001004 receive_config.renderer = observers[i];
1005 receive_config.rtp.remote_ssrc = ssrc;
1006 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
1007 VideoCodec codec =
1008 test::CreateDecoderVideoCodec(send_config.encoder_settings);
1009 receive_config.codecs.push_back(codec);
1010 receive_streams[i] =
1011 receiver_call->CreateVideoReceiveStream(receive_config);
1012 receive_streams[i]->Start();
1013
1014 frame_generators[i] = test::FrameGeneratorCapturer::Create(
1015 send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock());
1016 frame_generators[i]->Start();
1017 }
1018
1019 for (size_t i = 0; i < kNumStreams; ++i) {
1020 EXPECT_EQ(kEventSignaled, observers[i]->Wait())
1021 << "Timed out while waiting for observer " << i << " to render.";
1022 }
1023
1024 for (size_t i = 0; i < kNumStreams; ++i) {
1025 frame_generators[i]->Stop();
1026 sender_call->DestroyVideoSendStream(send_streams[i]);
1027 receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
1028 delete frame_generators[i];
1029 delete observers[i];
1030 }
1031
1032 sender_transport.StopSending();
1033 receiver_transport.StopSending();
1034}
1035
1036TEST_F(EndToEndTest, ObserversEncodedFrames) {
1037 class EncodedFrameTestObserver : public EncodedFrameObserver {
1038 public:
1039 EncodedFrameTestObserver()
1040 : length_(0),
1041 frame_type_(kFrameEmpty),
1042 called_(EventWrapper::Create()) {}
1043 virtual ~EncodedFrameTestObserver() {}
1044
1045 virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
1046 frame_type_ = encoded_frame.frame_type_;
1047 length_ = encoded_frame.length_;
1048 buffer_.reset(new uint8_t[length_]);
1049 memcpy(buffer_.get(), encoded_frame.data_, length_);
1050 called_->Set();
1051 }
1052
1053 EventTypeWrapper Wait() { return called_->Wait(kDefaultTimeoutMs); }
1054
1055 void ExpectEqualFrames(const EncodedFrameTestObserver& observer) {
1056 ASSERT_EQ(length_, observer.length_)
1057 << "Observed frames are of different lengths.";
1058 EXPECT_EQ(frame_type_, observer.frame_type_)
1059 << "Observed frames have different frame types.";
1060 EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_))
1061 << "Observed encoded frames have different content.";
1062 }
1063
1064 private:
1065 scoped_ptr<uint8_t[]> buffer_;
1066 size_t length_;
1067 FrameType frame_type_;
1068 scoped_ptr<EventWrapper> called_;
1069 };
1070
1071 EncodedFrameTestObserver post_encode_observer;
1072 EncodedFrameTestObserver pre_decode_observer;
1073
1074 test::DirectTransport sender_transport, receiver_transport;
1075
1076 CreateCalls(Call::Config(&sender_transport),
1077 Call::Config(&receiver_transport));
1078
1079 sender_transport.SetReceiver(receiver_call_->Receiver());
1080 receiver_transport.SetReceiver(sender_call_->Receiver());
1081
1082 CreateSendConfig(1);
1083 CreateMatchingReceiveConfigs();
1084 send_config_.post_encode_callback = &post_encode_observer;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001085 receive_configs_[0].pre_decode_callback = &pre_decode_observer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001086
1087 CreateStreams();
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001088 Start();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001089
1090 scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create(
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001091 encoder_config_.streams[0].width, encoder_config_.streams[0].height));
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001092 send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
1093
1094 EXPECT_EQ(kEventSignaled, post_encode_observer.Wait())
1095 << "Timed out while waiting for send-side encoded-frame callback.";
1096
1097 EXPECT_EQ(kEventSignaled, pre_decode_observer.Wait())
1098 << "Timed out while waiting for pre-decode encoded-frame callback.";
1099
1100 post_encode_observer.ExpectEqualFrames(pre_decode_observer);
1101
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001102 Stop();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001103
1104 sender_transport.StopSending();
1105 receiver_transport.StopSending();
1106
1107 DestroyStreams();
1108}
1109
1110TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
1111 class RembObserver : public test::EndToEndTest {
1112 public:
1113 RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
1114
1115 virtual Action OnReceiveRtcp(const uint8_t* packet,
1116 size_t length) OVERRIDE {
1117 RTCPUtility::RTCPParserV2 parser(packet, length, true);
1118 EXPECT_TRUE(parser.IsValid());
1119
1120 bool received_psfb = false;
1121 bool received_remb = false;
1122 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
1123 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
1124 if (packet_type == RTCPUtility::kRtcpPsfbRembCode) {
1125 const RTCPUtility::RTCPPacket& packet = parser.Packet();
1126 EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc);
1127 received_psfb = true;
1128 } else if (packet_type == RTCPUtility::kRtcpPsfbRembItemCode) {
1129 const RTCPUtility::RTCPPacket& packet = parser.Packet();
1130 EXPECT_GT(packet.REMBItem.BitRate, 0u);
1131 EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u);
1132 EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]);
1133 received_remb = true;
1134 }
1135 packet_type = parser.Iterate();
1136 }
1137 if (received_psfb && received_remb)
1138 observation_complete_->Set();
1139 return SEND_PACKET;
1140 }
1141 virtual void PerformTest() OVERRIDE {
1142 EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for a "
1143 "receiver RTCP REMB packet to be "
1144 "sent.";
1145 }
1146 } test;
1147
1148 RunBaseTest(&test);
1149}
1150
1151void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
1152 static const int kNumRtcpReportPacketsToObserve = 5;
1153 class RtcpXrObserver : public test::EndToEndTest {
1154 public:
1155 explicit RtcpXrObserver(bool enable_rrtr)
1156 : EndToEndTest(kDefaultTimeoutMs),
1157 enable_rrtr_(enable_rrtr),
1158 sent_rtcp_sr_(0),
1159 sent_rtcp_rr_(0),
1160 sent_rtcp_rrtr_(0),
1161 sent_rtcp_dlrr_(0) {}
1162
1163 private:
1164 // Receive stream should send RR packets (and RRTR packets if enabled).
1165 virtual Action OnReceiveRtcp(const uint8_t* packet,
1166 size_t length) OVERRIDE {
1167 RTCPUtility::RTCPParserV2 parser(packet, length, true);
1168 EXPECT_TRUE(parser.IsValid());
1169
1170 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
1171 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
1172 if (packet_type == RTCPUtility::kRtcpRrCode) {
1173 ++sent_rtcp_rr_;
1174 } else if (packet_type ==
1175 RTCPUtility::kRtcpXrReceiverReferenceTimeCode) {
1176 ++sent_rtcp_rrtr_;
1177 }
1178 EXPECT_NE(packet_type, RTCPUtility::kRtcpSrCode);
1179 EXPECT_NE(packet_type, RTCPUtility::kRtcpXrDlrrReportBlockItemCode);
1180 packet_type = parser.Iterate();
1181 }
1182 return SEND_PACKET;
1183 }
1184 // Send stream should send SR packets (and DLRR packets if enabled).
1185 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
1186 RTCPUtility::RTCPParserV2 parser(packet, length, true);
1187 EXPECT_TRUE(parser.IsValid());
1188
1189 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
1190 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
1191 if (packet_type == RTCPUtility::kRtcpSrCode) {
1192 ++sent_rtcp_sr_;
1193 } else if (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) {
1194 ++sent_rtcp_dlrr_;
1195 }
1196 EXPECT_NE(packet_type, RTCPUtility::kRtcpXrReceiverReferenceTimeCode);
1197 packet_type = parser.Iterate();
1198 }
1199 if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
1200 sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) {
1201 if (enable_rrtr_) {
1202 EXPECT_GT(sent_rtcp_rrtr_, 0);
1203 EXPECT_GT(sent_rtcp_dlrr_, 0);
1204 } else {
1205 EXPECT_EQ(0, sent_rtcp_rrtr_);
1206 EXPECT_EQ(0, sent_rtcp_dlrr_);
1207 }
1208 observation_complete_->Set();
1209 }
1210 return SEND_PACKET;
1211 }
1212
1213 virtual void ModifyConfigs(
1214 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001215 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001216 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001217 (*receive_configs)[0].rtp.rtcp_mode = newapi::kRtcpReducedSize;
1218 (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
1219 enable_rrtr_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001220 }
1221
1222 virtual void PerformTest() OVERRIDE {
1223 EXPECT_EQ(kEventSignaled, Wait())
1224 << "Timed out while waiting for RTCP SR/RR packets to be sent.";
1225 }
1226
1227 bool enable_rrtr_;
1228 int sent_rtcp_sr_;
1229 int sent_rtcp_rr_;
1230 int sent_rtcp_rrtr_;
1231 int sent_rtcp_dlrr_;
1232 } test(enable_rrtr);
1233
1234 RunBaseTest(&test);
1235}
1236
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001237void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
1238 bool send_single_ssrc_first) {
1239 class SendsSetSsrcs : public test::EndToEndTest {
1240 public:
1241 SendsSetSsrcs(const uint32_t* ssrcs,
1242 size_t num_ssrcs,
1243 bool send_single_ssrc_first)
1244 : EndToEndTest(kDefaultTimeoutMs),
1245 num_ssrcs_(num_ssrcs),
1246 send_single_ssrc_first_(send_single_ssrc_first),
1247 ssrcs_to_observe_(num_ssrcs),
1248 expect_single_ssrc_(send_single_ssrc_first) {
1249 for (size_t i = 0; i < num_ssrcs; ++i)
1250 valid_ssrcs_[ssrcs[i]] = true;
1251 }
1252
1253 private:
1254 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1255 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001256 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001257
1258 EXPECT_TRUE(valid_ssrcs_[header.ssrc])
1259 << "Received unknown SSRC: " << header.ssrc;
1260
1261 if (!valid_ssrcs_[header.ssrc])
1262 observation_complete_->Set();
1263
1264 if (!is_observed_[header.ssrc]) {
1265 is_observed_[header.ssrc] = true;
1266 --ssrcs_to_observe_;
1267 if (expect_single_ssrc_) {
1268 expect_single_ssrc_ = false;
1269 observation_complete_->Set();
1270 }
1271 }
1272
1273 if (ssrcs_to_observe_ == 0)
1274 observation_complete_->Set();
1275
1276 return SEND_PACKET;
1277 }
1278
1279 virtual size_t GetNumStreams() const OVERRIDE { return num_ssrcs_; }
1280
1281 virtual void ModifyConfigs(
1282 VideoSendStream::Config* send_config,
1283 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001284 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001285 if (num_ssrcs_ > 1) {
1286 // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001287 for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
1288 encoder_config->streams[i].min_bitrate_bps = 10000;
1289 encoder_config->streams[i].target_bitrate_bps = 15000;
1290 encoder_config->streams[i].max_bitrate_bps = 20000;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001291 }
1292 }
1293
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001294 encoder_config_all_streams_ = *encoder_config;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001295 if (send_single_ssrc_first_)
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001296 encoder_config->streams.resize(1);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001297 }
1298
1299 virtual void OnStreamsCreated(
1300 VideoSendStream* send_stream,
1301 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
1302 send_stream_ = send_stream;
1303 }
1304
1305 virtual void PerformTest() OVERRIDE {
1306 EXPECT_EQ(kEventSignaled, Wait())
1307 << "Timed out while waiting for "
1308 << (send_single_ssrc_first_ ? "first SSRC." : "SSRCs.");
1309
1310 if (send_single_ssrc_first_) {
1311 // Set full simulcast and continue with the rest of the SSRCs.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001312 send_stream_->ReconfigureVideoEncoder(encoder_config_all_streams_);
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001313 EXPECT_EQ(kEventSignaled, Wait())
1314 << "Timed out while waiting on additional SSRCs.";
1315 }
1316 }
1317
1318 private:
1319 std::map<uint32_t, bool> valid_ssrcs_;
1320 std::map<uint32_t, bool> is_observed_;
1321
1322 const size_t num_ssrcs_;
1323 const bool send_single_ssrc_first_;
1324
1325 size_t ssrcs_to_observe_;
1326 bool expect_single_ssrc_;
1327
1328 VideoSendStream* send_stream_;
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001329 VideoEncoderConfig encoder_config_all_streams_;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001330 } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first);
1331
1332 RunBaseTest(&test);
1333}
1334
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001335TEST_F(EndToEndTest, GetStats) {
1336 class StatsObserver : public test::EndToEndTest, public I420FrameCallback {
1337 public:
1338 StatsObserver()
1339 : EndToEndTest(kLongTimeoutMs),
1340 receive_stream_(NULL),
1341 send_stream_(NULL),
1342 expected_receive_ssrc_(),
1343 expected_send_ssrcs_(),
1344 check_stats_event_(EventWrapper::Create()) {}
1345
1346 private:
1347 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1348 check_stats_event_->Set();
1349 return SEND_PACKET;
1350 }
1351
1352 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
1353 check_stats_event_->Set();
1354 return SEND_PACKET;
1355 }
1356
1357 virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE {
1358 check_stats_event_->Set();
1359 return SEND_PACKET;
1360 }
1361
1362 virtual Action OnReceiveRtcp(const uint8_t* packet,
1363 size_t length) OVERRIDE {
1364 check_stats_event_->Set();
1365 return SEND_PACKET;
1366 }
1367
1368 virtual void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
1369 // Ensure that we have at least 5ms send side delay.
1370 int64_t render_time = video_frame->render_time_ms();
1371 if (render_time > 0)
1372 video_frame->set_render_time_ms(render_time - 5);
1373 }
1374
1375 bool CheckReceiveStats() {
1376 assert(receive_stream_ != NULL);
1377 VideoReceiveStream::Stats stats = receive_stream_->GetStats();
1378 EXPECT_EQ(expected_receive_ssrc_, stats.ssrc);
1379
1380 // Make sure all fields have been populated.
1381
1382 receive_stats_filled_["IncomingRate"] |=
1383 stats.network_frame_rate != 0 || stats.bitrate_bps != 0;
1384
1385 receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
1386
1387 receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
1388
1389 receive_stats_filled_["StatisticsUpdated"] |=
1390 stats.rtcp_stats.cumulative_lost != 0 ||
1391 stats.rtcp_stats.extended_max_sequence_number != 0 ||
1392 stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
1393
1394 receive_stats_filled_["DataCountersUpdated"] |=
1395 stats.rtp_stats.bytes != 0 || stats.rtp_stats.fec_packets != 0 ||
1396 stats.rtp_stats.header_bytes != 0 || stats.rtp_stats.packets != 0 ||
1397 stats.rtp_stats.padding_bytes != 0 ||
1398 stats.rtp_stats.retransmitted_packets != 0;
1399
1400 receive_stats_filled_["CodecStats"] |=
1401 stats.avg_delay_ms != 0 || stats.discarded_packets != 0 ||
1402 stats.key_frames != 0 || stats.delta_frames != 0;
1403
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001404 return AllStatsFilled(receive_stats_filled_);
1405 }
1406
1407 bool CheckSendStats() {
1408 assert(send_stream_ != NULL);
1409 VideoSendStream::Stats stats = send_stream_->GetStats();
1410
1411 send_stats_filled_["NumStreams"] |=
1412 stats.substreams.size() == expected_send_ssrcs_.size();
1413
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001414 for (std::map<uint32_t, StreamStats>::const_iterator it =
1415 stats.substreams.begin();
1416 it != stats.substreams.end();
1417 ++it) {
1418 EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
1419 expected_send_ssrcs_.end());
1420
1421 send_stats_filled_[CompoundKey("IncomingRate", it->first)] |=
1422 stats.input_frame_rate != 0;
1423
1424 const StreamStats& stream_stats = it->second;
1425
1426 send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
1427 stream_stats.rtcp_stats.cumulative_lost != 0 ||
1428 stream_stats.rtcp_stats.extended_max_sequence_number != 0 ||
1429 stream_stats.rtcp_stats.fraction_lost != 0;
1430
1431 send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |=
1432 stream_stats.rtp_stats.fec_packets != 0 ||
1433 stream_stats.rtp_stats.padding_bytes != 0 ||
1434 stream_stats.rtp_stats.retransmitted_packets != 0 ||
1435 stream_stats.rtp_stats.packets != 0;
1436
1437 send_stats_filled_[CompoundKey("BitrateStatisticsObserver",
1438 it->first)] |=
1439 stream_stats.bitrate_bps != 0;
1440
1441 send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
1442 stream_stats.delta_frames != 0 || stream_stats.key_frames != 0;
1443
1444 send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
1445 stats.encode_frame_rate != 0;
stefan@webrtc.org168f23f2014-07-11 13:44:02 +00001446
1447 send_stats_filled_[CompoundKey("Delay", it->first)] |=
1448 stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001449 }
1450
1451 return AllStatsFilled(send_stats_filled_);
1452 }
1453
1454 std::string CompoundKey(const char* name, uint32_t ssrc) {
1455 std::ostringstream oss;
1456 oss << name << "_" << ssrc;
1457 return oss.str();
1458 }
1459
1460 bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
1461 for (std::map<std::string, bool>::const_iterator it = stats_map.begin();
1462 it != stats_map.end();
1463 ++it) {
1464 if (!it->second)
1465 return false;
1466 }
1467 return true;
1468 }
1469
1470 virtual void ModifyConfigs(
1471 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001472 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001473 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001474 send_config->pre_encode_callback = this; // Used to inject delay.
1475 send_config->rtp.c_name = "SomeCName";
1476
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001477 expected_receive_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001478 const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
1479 for (size_t i = 0; i < ssrcs.size(); ++i)
1480 expected_send_ssrcs_.insert(ssrcs[i]);
1481
1482 expected_cname_ = send_config->rtp.c_name;
1483 }
1484
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001485 virtual void OnStreamsCreated(
1486 VideoSendStream* send_stream,
1487 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001488 send_stream_ = send_stream;
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001489 receive_stream_ = receive_streams[0];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001490 }
1491
1492 virtual void PerformTest() OVERRIDE {
1493 Clock* clock = Clock::GetRealTimeClock();
1494 int64_t now = clock->TimeInMilliseconds();
1495 int64_t stop_time = now + test::CallTest::kLongTimeoutMs;
1496 bool receive_ok = false;
1497 bool send_ok = false;
1498
1499 while (now < stop_time) {
1500 if (!receive_ok)
1501 receive_ok = CheckReceiveStats();
1502 if (!send_ok)
1503 send_ok = CheckSendStats();
1504
1505 if (receive_ok && send_ok)
1506 return;
1507
1508 int64_t time_until_timout_ = stop_time - now;
1509 if (time_until_timout_ > 0)
1510 check_stats_event_->Wait(time_until_timout_);
1511 now = clock->TimeInMilliseconds();
1512 }
1513
1514 ADD_FAILURE() << "Timed out waiting for filled stats.";
1515 for (std::map<std::string, bool>::const_iterator it =
1516 receive_stats_filled_.begin();
1517 it != receive_stats_filled_.end();
1518 ++it) {
1519 if (!it->second) {
1520 ADD_FAILURE() << "Missing receive stats: " << it->first;
1521 }
1522 }
1523
1524 for (std::map<std::string, bool>::const_iterator it =
1525 send_stats_filled_.begin();
1526 it != send_stats_filled_.end();
1527 ++it) {
1528 if (!it->second) {
1529 ADD_FAILURE() << "Missing send stats: " << it->first;
1530 }
1531 }
1532 }
1533
1534 VideoReceiveStream* receive_stream_;
1535 std::map<std::string, bool> receive_stats_filled_;
1536
1537 VideoSendStream* send_stream_;
1538 std::map<std::string, bool> send_stats_filled_;
1539
1540 uint32_t expected_receive_ssrc_;
1541 std::set<uint32_t> expected_send_ssrcs_;
1542 std::string expected_cname_;
1543
1544 scoped_ptr<EventWrapper> check_stats_event_;
1545 } test;
1546
1547 RunBaseTest(&test);
1548}
1549
1550TEST_F(EndToEndTest, ReceiverReferenceTimeReportEnabled) {
1551 TestXrReceiverReferenceTimeReport(true);
1552}
1553
1554TEST_F(EndToEndTest, ReceiverReferenceTimeReportDisabled) {
1555 TestXrReceiverReferenceTimeReport(false);
1556}
1557
1558TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
1559 static const size_t kNumRtpPacketsToSend = 5;
1560 class ReceivedRtpStatsObserver : public test::EndToEndTest {
1561 public:
1562 ReceivedRtpStatsObserver()
1563 : EndToEndTest(kDefaultTimeoutMs),
1564 receive_stream_(NULL),
1565 sent_rtp_(0) {}
1566
1567 private:
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001568 virtual void OnStreamsCreated(
1569 VideoSendStream* send_stream,
1570 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
1571 receive_stream_ = receive_streams[0];
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001572 }
1573
1574 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1575 if (sent_rtp_ >= kNumRtpPacketsToSend) {
1576 VideoReceiveStream::Stats stats = receive_stream_->GetStats();
1577 if (kNumRtpPacketsToSend == stats.rtp_stats.packets) {
1578 observation_complete_->Set();
1579 }
1580 return DROP_PACKET;
1581 }
1582 ++sent_rtp_;
1583 return SEND_PACKET;
1584 }
1585
1586 virtual void PerformTest() OVERRIDE {
1587 EXPECT_EQ(kEventSignaled, Wait())
1588 << "Timed out while verifying number of received RTP packets.";
1589 }
1590
1591 VideoReceiveStream* receive_stream_;
1592 uint32_t sent_rtp_;
1593 } test;
1594
1595 RunBaseTest(&test);
1596}
1597
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001598TEST_F(EndToEndTest, SendsSetSsrc) { TestSendsSetSsrcs(1, false); }
1599
1600TEST_F(EndToEndTest, SendsSetSimulcastSsrcs) {
1601 TestSendsSetSsrcs(kNumSsrcs, false);
1602}
1603
1604TEST_F(EndToEndTest, CanSwitchToUseAllSsrcs) {
1605 TestSendsSetSsrcs(kNumSsrcs, true);
1606}
1607
mflodman@webrtc.orgf9460682014-07-24 16:41:25 +00001608TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001609 class ObserveRedundantPayloads: public test::EndToEndTest {
1610 public:
1611 ObserveRedundantPayloads()
1612 : EndToEndTest(kDefaultTimeoutMs), ssrcs_to_observe_(kNumSsrcs) {
pbos@webrtc.orgdde16f12014-08-05 23:35:43 +00001613 for (size_t i = 0; i < kNumSsrcs; ++i) {
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001614 registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
1615 }
1616 }
1617
1618 private:
1619 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1620 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001621 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001622
1623 if (!registered_rtx_ssrc_[header.ssrc])
1624 return SEND_PACKET;
1625
1626 EXPECT_LE(static_cast<size_t>(header.headerLength + header.paddingLength),
1627 length);
1628 const bool packet_is_redundant_payload =
1629 static_cast<size_t>(header.headerLength + header.paddingLength) <
1630 length;
1631
1632 if (!packet_is_redundant_payload)
1633 return SEND_PACKET;
1634
1635 if (!observed_redundant_retransmission_[header.ssrc]) {
1636 observed_redundant_retransmission_[header.ssrc] = true;
1637 if (--ssrcs_to_observe_ == 0)
1638 observation_complete_->Set();
1639 }
1640
1641 return SEND_PACKET;
1642 }
1643
1644 virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; }
1645
1646 virtual void ModifyConfigs(
1647 VideoSendStream::Config* send_config,
1648 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001649 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001650 // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001651 for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
1652 encoder_config->streams[i].min_bitrate_bps = 10000;
1653 encoder_config->streams[i].target_bitrate_bps = 15000;
1654 encoder_config->streams[i].max_bitrate_bps = 20000;
pbos@webrtc.org20c1f562014-07-04 10:58:12 +00001655 }
1656 // Significantly higher than max bitrates for all video streams -> forcing
1657 // padding to trigger redundant padding on all RTX SSRCs.
1658 send_config->rtp.min_transmit_bitrate_bps = 100000;
1659
1660 send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
1661 send_config->rtp.rtx.pad_with_redundant_payloads = true;
1662
1663 for (size_t i = 0; i < kNumSsrcs; ++i)
1664 send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
1665 }
1666
1667 virtual void PerformTest() OVERRIDE {
1668 EXPECT_EQ(kEventSignaled, Wait())
1669 << "Timed out while waiting for redundant payloads on all SSRCs.";
1670 }
1671
1672 private:
1673 size_t ssrcs_to_observe_;
1674 std::map<uint32_t, bool> observed_redundant_retransmission_;
1675 std::map<uint32_t, bool> registered_rtx_ssrc_;
1676 } test;
1677
1678 RunBaseTest(&test);
1679}
1680
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001681void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
1682 static const uint32_t kMaxSequenceNumberGap = 100;
1683 static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
1684 class RtpSequenceObserver : public test::RtpRtcpObserver {
1685 public:
pbos@webrtc.orgdde16f12014-08-05 23:35:43 +00001686 explicit RtpSequenceObserver(bool use_rtx)
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001687 : test::RtpRtcpObserver(kDefaultTimeoutMs),
1688 crit_(CriticalSectionWrapper::CreateCriticalSection()),
1689 ssrcs_to_observe_(kNumSsrcs) {
1690 for (size_t i = 0; i < kNumSsrcs; ++i) {
1691 configured_ssrcs_[kSendSsrcs[i]] = true;
1692 if (use_rtx)
1693 configured_ssrcs_[kSendRtxSsrcs[i]] = true;
1694 }
1695 }
1696
1697 void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
1698 CriticalSectionScoped lock(crit_.get());
1699 ssrc_observed_.clear();
1700 ssrcs_to_observe_ = num_expected_ssrcs;
1701 }
1702
1703 private:
1704 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1705 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001706 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001707 const uint32_t ssrc = header.ssrc;
1708 const uint16_t sequence_number = header.sequenceNumber;
1709 const uint32_t timestamp = header.timestamp;
1710 const bool only_padding =
1711 static_cast<size_t>(header.headerLength + header.paddingLength) ==
1712 length;
1713
1714 EXPECT_TRUE(configured_ssrcs_[ssrc])
1715 << "Received SSRC that wasn't configured: " << ssrc;
1716
1717 std::map<uint32_t, uint16_t>::iterator it =
1718 last_observed_sequence_number_.find(header.ssrc);
1719 if (it == last_observed_sequence_number_.end()) {
1720 last_observed_sequence_number_[ssrc] = sequence_number;
1721 last_observed_timestamp_[ssrc] = timestamp;
1722 } else {
1723 // Verify sequence numbers are reasonably close.
1724 uint32_t extended_sequence_number = sequence_number;
1725 // Check for roll-over.
1726 if (sequence_number < last_observed_sequence_number_[ssrc])
1727 extended_sequence_number += 0xFFFFu + 1;
1728 EXPECT_LE(
1729 extended_sequence_number - last_observed_sequence_number_[ssrc],
1730 kMaxSequenceNumberGap)
1731 << "Gap in sequence numbers ("
1732 << last_observed_sequence_number_[ssrc] << " -> " << sequence_number
1733 << ") too large for SSRC: " << ssrc << ".";
1734 last_observed_sequence_number_[ssrc] = sequence_number;
1735
1736 // TODO(pbos): Remove this check if we ever have monotonically
1737 // increasing timestamps. Right now padding packets add a delta which
1738 // can cause reordering between padding packets and regular packets,
1739 // hence we drop padding-only packets to not flake.
1740 if (only_padding) {
1741 // Verify that timestamps are reasonably close.
1742 uint64_t extended_timestamp = timestamp;
1743 // Check for roll-over.
1744 if (timestamp < last_observed_timestamp_[ssrc])
1745 extended_timestamp += static_cast<uint64_t>(0xFFFFFFFFu) + 1;
1746 EXPECT_LE(extended_timestamp - last_observed_timestamp_[ssrc],
1747 kMaxTimestampGap)
1748 << "Gap in timestamps (" << last_observed_timestamp_[ssrc]
1749 << " -> " << timestamp << ") too large for SSRC: " << ssrc << ".";
1750 }
1751 last_observed_timestamp_[ssrc] = timestamp;
1752 }
1753
1754 CriticalSectionScoped lock(crit_.get());
1755 // Wait for media packets on all ssrcs.
1756 if (!ssrc_observed_[ssrc] && !only_padding) {
1757 ssrc_observed_[ssrc] = true;
1758 if (--ssrcs_to_observe_ == 0)
1759 observation_complete_->Set();
1760 }
1761
1762 return SEND_PACKET;
1763 }
1764
1765 std::map<uint32_t, uint16_t> last_observed_sequence_number_;
1766 std::map<uint32_t, uint32_t> last_observed_timestamp_;
1767 std::map<uint32_t, bool> configured_ssrcs_;
1768
1769 scoped_ptr<CriticalSectionWrapper> crit_;
1770 size_t ssrcs_to_observe_ GUARDED_BY(crit_);
1771 std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_);
1772 } observer(use_rtx);
1773
1774 CreateCalls(Call::Config(observer.SendTransport()),
1775 Call::Config(observer.ReceiveTransport()));
1776 observer.SetReceivers(sender_call_->Receiver(), NULL);
1777
1778 CreateSendConfig(kNumSsrcs);
1779
1780 if (use_rtx) {
1781 for (size_t i = 0; i < kNumSsrcs; ++i) {
1782 send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
1783 }
1784 send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
1785 }
1786
1787 // Lower bitrates so that all streams send initially.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001788 for (size_t i = 0; i < encoder_config_.streams.size(); ++i) {
1789 encoder_config_.streams[i].min_bitrate_bps = 10000;
1790 encoder_config_.streams[i].target_bitrate_bps = 15000;
1791 encoder_config_.streams[i].max_bitrate_bps = 20000;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001792 }
1793
1794 CreateMatchingReceiveConfigs();
1795
1796 CreateStreams();
1797 CreateFrameGeneratorCapturer();
1798
1799 Start();
1800 EXPECT_EQ(kEventSignaled, observer.Wait())
1801 << "Timed out waiting for all SSRCs to send packets.";
1802
1803 // Test stream resetting more than once to make sure that the state doesn't
1804 // get set once (this could be due to using std::map::insert for instance).
1805 for (size_t i = 0; i < 3; ++i) {
1806 frame_generator_capturer_->Stop();
1807 sender_call_->DestroyVideoSendStream(send_stream_);
1808
1809 // Re-create VideoSendStream with only one stream.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001810 VideoEncoderConfig one_stream = encoder_config_;
1811 one_stream.streams.resize(1);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001812 send_stream_ =
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001813 sender_call_->CreateVideoSendStream(send_config_, one_stream);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001814 send_stream_->Start();
1815 CreateFrameGeneratorCapturer();
1816 frame_generator_capturer_->Start();
1817
1818 observer.ResetExpectedSsrcs(1);
1819 EXPECT_EQ(kEventSignaled, observer.Wait())
1820 << "Timed out waiting for single RTP packet.";
1821
1822 // Reconfigure back to use all streams.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001823 send_stream_->ReconfigureVideoEncoder(encoder_config_);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001824 observer.ResetExpectedSsrcs(kNumSsrcs);
1825 EXPECT_EQ(kEventSignaled, observer.Wait())
1826 << "Timed out waiting for all SSRCs to send packets.";
1827
1828 // Reconfigure down to one stream.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001829 send_stream_->ReconfigureVideoEncoder(one_stream);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001830 observer.ResetExpectedSsrcs(1);
1831 EXPECT_EQ(kEventSignaled, observer.Wait())
1832 << "Timed out waiting for single RTP packet.";
1833
1834 // Reconfigure back to use all streams.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001835 send_stream_->ReconfigureVideoEncoder(encoder_config_);
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001836 observer.ResetExpectedSsrcs(kNumSsrcs);
1837 EXPECT_EQ(kEventSignaled, observer.Wait())
1838 << "Timed out waiting for all SSRCs to send packets.";
1839 }
1840
1841 observer.StopSending();
1842
1843 Stop();
1844 DestroyStreams();
1845}
1846
aluebs@webrtc.orgb623c5c2014-08-26 14:22:51 +00001847TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpState) {
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +00001848 TestRtpStatePreservation(false);
1849}
1850
1851TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
1852 TestRtpStatePreservation(true);
1853}
1854
pbos@webrtc.org26c0c412014-09-03 16:17:12 +00001855TEST_F(EndToEndTest, RespectsNetworkState) {
1856 // TODO(pbos): Remove accepted downtime packets etc. when signaling network
1857 // down blocks until no more packets will be sent.
1858
1859 // Pacer will send from its packet list and then send required padding before
1860 // checking paused_ again. This should be enough for one round of pacing,
1861 // otherwise increase.
1862 static const int kNumAcceptedDowntimeRtp = 5;
1863 // A single RTCP may be in the pipeline.
1864 static const int kNumAcceptedDowntimeRtcp = 1;
1865 class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
1866 public:
1867 NetworkStateTest()
1868 : EndToEndTest(kDefaultTimeoutMs),
1869 FakeEncoder(Clock::GetRealTimeClock()),
1870 test_crit_(CriticalSectionWrapper::CreateCriticalSection()),
1871 encoded_frames_(EventWrapper::Create()),
1872 sender_packets_(EventWrapper::Create()),
1873 receiver_packets_(EventWrapper::Create()),
1874 sender_state_(Call::kNetworkUp),
1875 down_sender_rtp_(0),
1876 down_sender_rtcp_(0),
1877 receiver_state_(Call::kNetworkUp),
1878 down_receiver_rtcp_(0),
1879 down_frames_(0) {}
1880
1881 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1882 CriticalSectionScoped lock(test_crit_.get());
1883 if (sender_state_ == Call::kNetworkDown) {
1884 ++down_sender_rtp_;
1885 EXPECT_LE(down_sender_rtp_, kNumAcceptedDowntimeRtp)
1886 << "RTP sent during sender-side downtime.";
1887 if (down_sender_rtp_> kNumAcceptedDowntimeRtp)
1888 sender_packets_->Set();
1889 } else {
1890 sender_packets_->Set();
1891 }
1892 return SEND_PACKET;
1893 }
1894
1895 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
1896 CriticalSectionScoped lock(test_crit_.get());
1897 if (sender_state_ == Call::kNetworkDown) {
1898 ++down_sender_rtcp_;
1899 EXPECT_LE(down_sender_rtcp_, kNumAcceptedDowntimeRtcp)
1900 << "RTCP sent during sender-side downtime.";
1901 if (down_sender_rtcp_ > kNumAcceptedDowntimeRtcp)
1902 sender_packets_->Set();
1903 } else {
1904 sender_packets_->Set();
1905 }
1906 return SEND_PACKET;
1907 }
1908
1909 virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE {
1910 ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
1911 return SEND_PACKET;
1912 }
1913
1914 virtual Action OnReceiveRtcp(const uint8_t* packet,
1915 size_t length) OVERRIDE {
1916 CriticalSectionScoped lock(test_crit_.get());
1917 if (receiver_state_ == Call::kNetworkDown) {
1918 ++down_receiver_rtcp_;
1919 EXPECT_LE(down_receiver_rtcp_, kNumAcceptedDowntimeRtcp)
1920 << "RTCP sent during receiver-side downtime.";
1921 if (down_receiver_rtcp_ > kNumAcceptedDowntimeRtcp)
1922 receiver_packets_->Set();
1923 } else {
1924 receiver_packets_->Set();
1925 }
1926 return SEND_PACKET;
1927 }
1928
1929 virtual void OnCallsCreated(Call* sender_call,
1930 Call* receiver_call) OVERRIDE {
1931 sender_call_ = sender_call;
1932 receiver_call_ = receiver_call;
1933 }
1934
1935 virtual void ModifyConfigs(
1936 VideoSendStream::Config* send_config,
1937 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +00001938 VideoEncoderConfig* encoder_config) OVERRIDE {
pbos@webrtc.org26c0c412014-09-03 16:17:12 +00001939 send_config->encoder_settings.encoder = this;
1940 }
1941
1942 virtual void PerformTest() OVERRIDE {
1943 EXPECT_EQ(kEventSignaled, encoded_frames_->Wait(kDefaultTimeoutMs))
1944 << "No frames received by the encoder.";
1945 EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs))
1946 << "Timed out waiting for send-side packets.";
1947 EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
1948 << "Timed out waiting for receiver-side packets.";
1949
1950 // Sender-side network down.
1951 sender_call_->SignalNetworkState(Call::kNetworkDown);
1952 {
1953 CriticalSectionScoped lock(test_crit_.get());
1954 sender_packets_->Reset(); // Earlier packets should not count.
1955 sender_state_ = Call::kNetworkDown;
1956 }
1957 EXPECT_EQ(kEventTimeout, sender_packets_->Wait(kSilenceTimeoutMs))
1958 << "Packets sent during sender-network downtime.";
1959 EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
1960 << "Timed out waiting for receiver-side packets.";
1961 // Receiver-side network down.
1962 receiver_call_->SignalNetworkState(Call::kNetworkDown);
1963 {
1964 CriticalSectionScoped lock(test_crit_.get());
1965 receiver_packets_->Reset(); // Earlier packets should not count.
1966 receiver_state_ = Call::kNetworkDown;
1967 }
1968 EXPECT_EQ(kEventTimeout, receiver_packets_->Wait(kSilenceTimeoutMs))
1969 << "Packets sent during receiver-network downtime.";
1970
1971 // Network back up again for both.
1972 {
1973 CriticalSectionScoped lock(test_crit_.get());
1974 sender_packets_->Reset(); // Earlier packets should not count.
1975 receiver_packets_->Reset(); // Earlier packets should not count.
1976 sender_state_ = receiver_state_ = Call::kNetworkUp;
1977 }
1978 sender_call_->SignalNetworkState(Call::kNetworkUp);
1979 receiver_call_->SignalNetworkState(Call::kNetworkUp);
1980 EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs))
1981 << "Timed out waiting for send-side packets.";
1982 EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
1983 << "Timed out waiting for receiver-side packets.";
1984 }
1985
1986 virtual int32_t Encode(const I420VideoFrame& input_image,
1987 const CodecSpecificInfo* codec_specific_info,
1988 const std::vector<VideoFrameType>* frame_types)
1989 OVERRIDE {
1990 {
1991 CriticalSectionScoped lock(test_crit_.get());
1992 if (sender_state_ == Call::kNetworkDown) {
1993 ++down_frames_;
1994 EXPECT_LE(down_frames_, 1)
1995 << "Encoding more than one frame while network is down.";
1996 if (down_frames_ > 1)
1997 encoded_frames_->Set();
1998 } else {
1999 encoded_frames_->Set();
2000 }
2001 }
2002 return test::FakeEncoder::Encode(
2003 input_image, codec_specific_info, frame_types);
2004 }
2005
2006 private:
2007 const scoped_ptr<CriticalSectionWrapper> test_crit_;
2008 scoped_ptr<EventWrapper> encoded_frames_;
2009 scoped_ptr<EventWrapper> sender_packets_;
2010 scoped_ptr<EventWrapper> receiver_packets_;
2011 Call* sender_call_;
2012 Call* receiver_call_;
2013 Call::NetworkState sender_state_ GUARDED_BY(test_crit_);
2014 int down_sender_rtp_ GUARDED_BY(test_crit_);
2015 int down_sender_rtcp_ GUARDED_BY(test_crit_);
2016 Call::NetworkState receiver_state_ GUARDED_BY(test_crit_);
2017 int down_receiver_rtcp_ GUARDED_BY(test_crit_);
2018 int down_frames_ GUARDED_BY(test_crit_);
2019 } test;
2020
2021 RunBaseTest(&test);
2022}
2023
2024TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
2025 class UnusedEncoder : public test::FakeEncoder {
2026 public:
2027 UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
2028 virtual int32_t Encode(const I420VideoFrame& input_image,
2029 const CodecSpecificInfo* codec_specific_info,
2030 const std::vector<VideoFrameType>* frame_types)
2031 OVERRIDE {
2032 ADD_FAILURE() << "Unexpected frame encode.";
2033 return test::FakeEncoder::Encode(
2034 input_image, codec_specific_info, frame_types);
2035 }
2036 };
2037
2038 UnusedTransport transport;
2039 CreateSenderCall(Call::Config(&transport));
2040 sender_call_->SignalNetworkState(Call::kNetworkDown);
2041
2042 CreateSendConfig(1);
2043 UnusedEncoder unused_encoder;
2044 send_config_.encoder_settings.encoder = &unused_encoder;
2045 CreateStreams();
2046 CreateFrameGeneratorCapturer();
2047
2048 Start();
2049 SleepMs(kSilenceTimeoutMs);
2050 Stop();
2051
2052 DestroyStreams();
2053}
2054
2055TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {
2056 test::DirectTransport sender_transport;
2057 CreateSenderCall(Call::Config(&sender_transport));
2058 UnusedTransport transport;
2059 CreateReceiverCall(Call::Config(&transport));
2060 sender_transport.SetReceiver(receiver_call_->Receiver());
2061
2062 receiver_call_->SignalNetworkState(Call::kNetworkDown);
2063
2064 CreateSendConfig(1);
2065 CreateMatchingReceiveConfigs();
2066 CreateStreams();
2067 CreateFrameGeneratorCapturer();
2068
2069 Start();
2070 SleepMs(kSilenceTimeoutMs);
2071 Stop();
2072
2073 sender_transport.StopSending();
2074
2075 DestroyStreams();
2076}
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00002077} // namespace webrtc