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pbos@webrtc.org744fbc72013-09-10 09:26:25 +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 */
pbos@webrtc.org744fbc72013-09-10 09:26:25 +000010
11#include "testing/gtest/include/gtest/gtest.h"
pbos@webrtc.org744fbc72013-09-10 09:26:25 +000012#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
13#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +000014#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
pbos@webrtc.org744fbc72013-09-10 09:26:25 +000015#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
16#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
17#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +000018#include "webrtc/test/testsupport/perf_test.h"
stefan@webrtc.org3d7da882014-07-08 13:59:46 +000019#include "webrtc/video/rampup_tests.h"
pbos@webrtc.org744fbc72013-09-10 09:26:25 +000020
21namespace webrtc {
pbos@webrtc.org29023282013-09-11 10:14:56 +000022namespace {
stefan@webrtc.org3d7da882014-07-08 13:59:46 +000023
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +000024static const int kMaxPacketSize = 1500;
pbos@webrtc.org29023282013-09-11 10:14:56 +000025
stefan@webrtc.org3d7da882014-07-08 13:59:46 +000026std::vector<uint32_t> GenerateSsrcs(size_t num_streams,
27 uint32_t ssrc_offset) {
28 std::vector<uint32_t> ssrcs;
29 for (size_t i = 0; i != num_streams; ++i)
30 ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i));
31 return ssrcs;
32}
mflodman@webrtc.orgeb16b812014-06-16 08:57:39 +000033} // namespace
henrik.lundin@webrtc.org845862f2014-03-06 07:19:28 +000034
stefan@webrtc.org3d7da882014-07-08 13:59:46 +000035StreamObserver::StreamObserver(const SsrcMap& rtx_media_ssrcs,
36 newapi::Transport* feedback_transport,
37 Clock* clock,
38 RemoteBitrateEstimatorFactory* rbe_factory,
39 RateControlType control_type)
40 : clock_(clock),
41 test_done_(EventWrapper::Create()),
42 rtp_parser_(RtpHeaderParser::Create()),
43 feedback_transport_(feedback_transport),
44 receive_stats_(ReceiveStatistics::Create(clock)),
45 payload_registry_(
46 new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
47 crit_(CriticalSectionWrapper::CreateCriticalSection()),
48 expected_bitrate_bps_(0),
49 start_bitrate_bps_(0),
50 rtx_media_ssrcs_(rtx_media_ssrcs),
51 total_sent_(0),
52 padding_sent_(0),
53 rtx_media_sent_(0),
54 total_packets_sent_(0),
55 padding_packets_sent_(0),
56 rtx_media_packets_sent_(0),
57 test_start_ms_(clock_->TimeInMilliseconds()),
58 ramp_up_finished_ms_(0) {
59 // Ideally we would only have to instantiate an RtcpSender, an
60 // RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current
61 // state of the RTP module we need a full module and receive statistics to
62 // be able to produce an RTCP with REMB.
63 RtpRtcp::Configuration config;
64 config.receive_statistics = receive_stats_.get();
65 feedback_transport_.Enable();
66 config.outgoing_transport = &feedback_transport_;
67 rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
68 rtp_rtcp_->SetREMBStatus(true);
69 rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
70 rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
71 kAbsSendTimeExtensionId);
72 rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
73 kTransmissionTimeOffsetExtensionId);
74 const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
75 remote_bitrate_estimator_.reset(
76 rbe_factory->Create(this, clock, control_type,
77 kRemoteBitrateEstimatorMinBitrateBps));
78}
79
80void StreamObserver::set_expected_bitrate_bps(
81 unsigned int expected_bitrate_bps) {
82 CriticalSectionScoped lock(crit_.get());
83 expected_bitrate_bps_ = expected_bitrate_bps;
84}
85
86void StreamObserver::set_start_bitrate_bps(unsigned int start_bitrate_bps) {
87 CriticalSectionScoped lock(crit_.get());
88 start_bitrate_bps_ = start_bitrate_bps;
89}
90
91void StreamObserver::OnReceiveBitrateChanged(
92 const std::vector<unsigned int>& ssrcs, unsigned int bitrate) {
93 CriticalSectionScoped lock(crit_.get());
94 assert(expected_bitrate_bps_ > 0);
95 if (start_bitrate_bps_ != 0) {
96 // For tests with an explicitly set start bitrate, verify the first
97 // bitrate estimate is close to the start bitrate and lower than the
98 // test target bitrate. This is to verify a call respects the configured
99 // start bitrate, but due to the BWE implementation we can't guarantee the
100 // first estimate really is as high as the start bitrate.
101 EXPECT_GT(bitrate, 0.9 * start_bitrate_bps_);
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000102 start_bitrate_bps_ = 0;
103 }
104 if (bitrate >= expected_bitrate_bps_) {
105 ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
106 // Just trigger if there was any rtx padding packet.
107 if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) {
108 TriggerTestDone();
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +0000109 }
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000110 }
pbos@webrtc.org49ff40e2014-11-13 14:42:37 +0000111 rtp_rtcp_->SetREMBData(bitrate, ssrcs);
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000112 rtp_rtcp_->Process();
113}
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +0000114
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000115bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
116 CriticalSectionScoped lock(crit_.get());
117 RTPHeader header;
118 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
119 receive_stats_->IncomingPacket(header, length, false);
120 payload_registry_->SetIncomingPayloadType(header);
121 remote_bitrate_estimator_->IncomingPacket(
122 clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
123 if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
124 remote_bitrate_estimator_->Process();
125 }
126 total_sent_ += length;
127 padding_sent_ += header.paddingLength;
128 ++total_packets_sent_;
129 if (header.paddingLength > 0)
130 ++padding_packets_sent_;
131 if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) {
132 rtx_media_sent_ += length - header.headerLength - header.paddingLength;
133 if (header.paddingLength == 0)
134 ++rtx_media_packets_sent_;
135 uint8_t restored_packet[kMaxPacketSize];
136 uint8_t* restored_packet_ptr = restored_packet;
137 int restored_length = static_cast<int>(length);
138 payload_registry_->RestoreOriginalPacket(&restored_packet_ptr,
139 packet,
140 &restored_length,
141 rtx_media_ssrcs_[header.ssrc],
142 header);
143 length = restored_length;
144 EXPECT_TRUE(rtp_parser_->Parse(
145 restored_packet, static_cast<int>(length), &header));
146 } else {
147 rtp_rtcp_->SetRemoteSSRC(header.ssrc);
148 }
149 return true;
150}
151
152bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
153 return true;
154}
155
pbos@webrtc.org32452b22014-10-22 12:15:24 +0000156EventTypeWrapper StreamObserver::Wait() {
157 return test_done_->Wait(test::CallTest::kLongTimeoutMs);
158}
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000159
160void StreamObserver::ReportResult(const std::string& measurement,
161 size_t value,
162 const std::string& units) {
163 webrtc::test::PrintResult(
164 measurement, "",
165 ::testing::UnitTest::GetInstance()->current_test_info()->name(),
166 value, units, false);
167}
168
169void StreamObserver::TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
170 ReportResult("ramp-up-total-sent", total_sent_, "bytes");
171 ReportResult("ramp-up-padding-sent", padding_sent_, "bytes");
172 ReportResult("ramp-up-rtx-media-sent", rtx_media_sent_, "bytes");
173 ReportResult("ramp-up-total-packets-sent", total_packets_sent_, "packets");
174 ReportResult("ramp-up-padding-packets-sent",
175 padding_packets_sent_,
176 "packets");
177 ReportResult("ramp-up-rtx-packets-sent",
178 rtx_media_packets_sent_,
179 "packets");
180 ReportResult("ramp-up-time",
181 ramp_up_finished_ms_ - test_start_ms_,
182 "milliseconds");
183 test_done_->Set();
184}
185
186LowRateStreamObserver::LowRateStreamObserver(
187 newapi::Transport* feedback_transport,
188 Clock* clock,
189 size_t number_of_streams,
190 bool rtx_used)
191 : clock_(clock),
192 number_of_streams_(number_of_streams),
193 rtx_used_(rtx_used),
194 test_done_(EventWrapper::Create()),
195 rtp_parser_(RtpHeaderParser::Create()),
196 feedback_transport_(feedback_transport),
197 receive_stats_(ReceiveStatistics::Create(clock)),
198 crit_(CriticalSectionWrapper::CreateCriticalSection()),
199 send_stream_(NULL),
200 test_state_(kFirstRampup),
201 state_start_ms_(clock_->TimeInMilliseconds()),
202 interval_start_ms_(state_start_ms_),
203 last_remb_bps_(0),
204 sent_bytes_(0),
205 total_overuse_bytes_(0),
206 suspended_in_stats_(false) {
207 RtpRtcp::Configuration config;
208 config.receive_statistics = receive_stats_.get();
209 feedback_transport_.Enable();
210 config.outgoing_transport = &feedback_transport_;
211 rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
212 rtp_rtcp_->SetREMBStatus(true);
213 rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
214 rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
215 kTransmissionTimeOffsetExtensionId);
216 AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
217 const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000;
218 remote_bitrate_estimator_.reset(
219 rbe_factory.Create(this, clock, kMimdControl,
220 kRemoteBitrateEstimatorMinBitrateBps));
221 forward_transport_config_.link_capacity_kbps =
222 kHighBandwidthLimitBps / 1000;
stefan@webrtc.orgb8e9e442014-07-09 11:29:06 +0000223 forward_transport_config_.queue_length_packets = 100; // Something large.
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000224 test::DirectTransport::SetConfig(forward_transport_config_);
225 test::DirectTransport::SetReceiver(this);
226}
227
228void LowRateStreamObserver::SetSendStream(const VideoSendStream* send_stream) {
229 CriticalSectionScoped lock(crit_.get());
230 send_stream_ = send_stream;
231}
232
233void LowRateStreamObserver::OnReceiveBitrateChanged(
234 const std::vector<unsigned int>& ssrcs,
235 unsigned int bitrate) {
236 CriticalSectionScoped lock(crit_.get());
pbos@webrtc.org49ff40e2014-11-13 14:42:37 +0000237 rtp_rtcp_->SetREMBData(bitrate, ssrcs);
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000238 rtp_rtcp_->Process();
239 last_remb_bps_ = bitrate;
240}
241
242bool LowRateStreamObserver::SendRtp(const uint8_t* data, size_t length) {
243 CriticalSectionScoped lock(crit_.get());
244 sent_bytes_ += length;
245 int64_t now_ms = clock_->TimeInMilliseconds();
246 if (now_ms > interval_start_ms_ + 1000) { // Let at least 1 second pass.
247 // Verify that the send rate was about right.
248 unsigned int average_rate_bps = static_cast<unsigned int>(sent_bytes_) *
249 8 * 1000 / (now_ms - interval_start_ms_);
250 // TODO(holmer): Why is this failing?
251 // EXPECT_LT(average_rate_bps, last_remb_bps_ * 1.1);
252 if (average_rate_bps > last_remb_bps_ * 1.1) {
253 total_overuse_bytes_ +=
254 sent_bytes_ -
255 last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000;
mflodman@webrtc.orgeb16b812014-06-16 08:57:39 +0000256 }
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000257 EvolveTestState(average_rate_bps);
258 interval_start_ms_ = now_ms;
259 sent_bytes_ = 0;
260 }
261 return test::DirectTransport::SendRtp(data, length);
262}
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +0000263
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000264PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket(
265 const uint8_t* packet, size_t length) {
266 CriticalSectionScoped lock(crit_.get());
267 RTPHeader header;
268 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
269 receive_stats_->IncomingPacket(header, length, false);
270 remote_bitrate_estimator_->IncomingPacket(
271 clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
272 if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
273 remote_bitrate_estimator_->Process();
274 }
275 suspended_in_stats_ = send_stream_->GetStats().suspended;
276 return DELIVERY_OK;
277}
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +0000278
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000279bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
280 return true;
281}
andresp@webrtc.orgc1480792014-03-20 03:23:55 +0000282
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000283std::string LowRateStreamObserver::GetModifierString() {
284 std::string str("_");
285 char temp_str[5];
286 sprintf(temp_str, "%i",
287 static_cast<int>(number_of_streams_));
288 str += std::string(temp_str);
289 str += "stream";
290 str += (number_of_streams_ > 1 ? "s" : "");
291 str += "_";
292 str += (rtx_used_ ? "" : "no");
293 str += "rtx";
294 return str;
295}
andresp@webrtc.orgc1480792014-03-20 03:23:55 +0000296
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000297void LowRateStreamObserver::EvolveTestState(unsigned int bitrate_bps) {
298 int64_t now = clock_->TimeInMilliseconds();
299 CriticalSectionScoped lock(crit_.get());
300 assert(send_stream_ != NULL);
301 switch (test_state_) {
302 case kFirstRampup: {
303 EXPECT_FALSE(suspended_in_stats_);
304 if (bitrate_bps > kExpectedHighBitrateBps) {
305 // The first ramp-up has reached the target bitrate. Change the
306 // channel limit, and move to the next test state.
307 forward_transport_config_.link_capacity_kbps =
308 kLowBandwidthLimitBps / 1000;
309 test::DirectTransport::SetConfig(forward_transport_config_);
310 test_state_ = kLowRate;
311 webrtc::test::PrintResult("ramp_up_down_up",
312 GetModifierString(),
313 "first_rampup",
314 now - state_start_ms_,
315 "ms",
316 false);
317 state_start_ms_ = now;
318 interval_start_ms_ = now;
319 sent_bytes_ = 0;
andresp@webrtc.orgc1480792014-03-20 03:23:55 +0000320 }
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000321 break;
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000322 }
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000323 case kLowRate: {
324 if (bitrate_bps < kExpectedLowBitrateBps && suspended_in_stats_) {
325 // The ramp-down was successful. Change the channel limit back to a
326 // high value, and move to the next test state.
327 forward_transport_config_.link_capacity_kbps =
328 kHighBandwidthLimitBps / 1000;
329 test::DirectTransport::SetConfig(forward_transport_config_);
330 test_state_ = kSecondRampup;
331 webrtc::test::PrintResult("ramp_up_down_up",
332 GetModifierString(),
333 "rampdown",
334 now - state_start_ms_,
335 "ms",
336 false);
337 state_start_ms_ = now;
338 interval_start_ms_ = now;
339 sent_bytes_ = 0;
340 }
341 break;
342 }
343 case kSecondRampup: {
344 if (bitrate_bps > kExpectedHighBitrateBps && !suspended_in_stats_) {
345 webrtc::test::PrintResult("ramp_up_down_up",
346 GetModifierString(),
347 "second_rampup",
348 now - state_start_ms_,
349 "ms",
350 false);
351 webrtc::test::PrintResult("ramp_up_down_up",
352 GetModifierString(),
353 "total_overuse",
354 total_overuse_bytes_,
355 "bytes",
356 false);
357 test_done_->Set();
358 }
359 break;
360 }
361 }
362}
pbos@webrtc.orgf577ae92014-03-19 08:43:57 +0000363
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000364EventTypeWrapper LowRateStreamObserver::Wait() {
365 return test_done_->Wait(test::CallTest::kLongTimeoutMs);
366}
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +0000367
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000368void RampUpTest::RunRampUpTest(bool rtx,
369 size_t num_streams,
370 unsigned int start_bitrate_bps,
371 const std::string& extension_type) {
372 std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100));
373 std::vector<uint32_t> rtx_ssrcs(GenerateSsrcs(num_streams, 200));
374 StreamObserver::SsrcMap rtx_ssrc_map;
375 if (rtx) {
376 for (size_t i = 0; i < ssrcs.size(); ++i)
377 rtx_ssrc_map[rtx_ssrcs[i]] = ssrcs[i];
stefan@webrtc.org7e9315b2013-12-04 10:24:26 +0000378 }
henrik.lundin@webrtc.org845862f2014-03-06 07:19:28 +0000379
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000380 CreateSendConfig(num_streams);
henrik.lundin@webrtc.org845862f2014-03-06 07:19:28 +0000381
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000382 scoped_ptr<RemoteBitrateEstimatorFactory> rbe_factory;
383 RateControlType control_type;
384 if (extension_type == RtpExtension::kAbsSendTime) {
385 control_type = kAimdControl;
386 rbe_factory.reset(new AbsoluteSendTimeRemoteBitrateEstimatorFactory);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000387 send_config_.rtp.extensions.push_back(RtpExtension(
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000388 extension_type.c_str(), kAbsSendTimeExtensionId));
389 } else {
390 control_type = kMimdControl;
391 rbe_factory.reset(new RemoteBitrateEstimatorFactory);
392 send_config_.rtp.extensions.push_back(RtpExtension(
393 extension_type.c_str(), kTransmissionTimeOffsetExtensionId));
henrik.lundin@webrtc.org845862f2014-03-06 07:19:28 +0000394 }
395
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000396 test::DirectTransport receiver_transport;
397 StreamObserver stream_observer(rtx_ssrc_map,
398 &receiver_transport,
399 Clock::GetRealTimeClock(),
400 rbe_factory.get(),
401 control_type);
402
403 Call::Config call_config(&stream_observer);
404 if (start_bitrate_bps != 0) {
pbos@webrtc.orga73a6782014-10-14 11:52:10 +0000405 call_config.stream_start_bitrate_bps = start_bitrate_bps;
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000406 stream_observer.set_start_bitrate_bps(start_bitrate_bps);
andresp@webrtc.orga714eaf2014-03-17 15:34:57 +0000407 }
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000408
409 CreateSenderCall(call_config);
410
411 receiver_transport.SetReceiver(sender_call_->Receiver());
412
413 if (num_streams == 1) {
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000414 encoder_config_.streams[0].target_bitrate_bps = 2000000;
415 encoder_config_.streams[0].max_bitrate_bps = 2000000;
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000416 }
417
418 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
419 send_config_.rtp.ssrcs = ssrcs;
420 if (rtx) {
421 send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
422 send_config_.rtp.rtx.ssrcs = rtx_ssrcs;
423 send_config_.rtp.rtx.pad_with_redundant_payloads = true;
424 }
425
426 if (num_streams == 1) {
427 // For single stream rampup until 1mbps
428 stream_observer.set_expected_bitrate_bps(kSingleStreamTargetBps);
429 } else {
430 // For multi stream rampup until all streams are being sent. That means
431 // enough birate to send all the target streams plus the min bitrate of
432 // the last one.
pbos@webrtc.orgbbe0a852014-09-19 12:30:25 +0000433 int expected_bitrate_bps = encoder_config_.streams.back().min_bitrate_bps;
434 for (size_t i = 0; i < encoder_config_.streams.size() - 1; ++i) {
435 expected_bitrate_bps += encoder_config_.streams[i].target_bitrate_bps;
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000436 }
437 stream_observer.set_expected_bitrate_bps(expected_bitrate_bps);
438 }
439
440 CreateStreams();
441 CreateFrameGeneratorCapturer();
442
443 Start();
444
445 EXPECT_EQ(kEventSignaled, stream_observer.Wait());
446
447 Stop();
448 DestroyStreams();
449}
450
451void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) {
452 test::DirectTransport receiver_transport;
453 LowRateStreamObserver stream_observer(
454 &receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx);
455
456 Call::Config call_config(&stream_observer);
457 CreateSenderCall(call_config);
458 receiver_transport.SetReceiver(sender_call_->Receiver());
459
460 CreateSendConfig(number_of_streams);
461
462 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
463 send_config_.rtp.extensions.push_back(RtpExtension(
464 RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId));
465 send_config_.suspend_below_min_bitrate = true;
466 if (rtx) {
467 send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
468 send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200);
469 send_config_.rtp.rtx.pad_with_redundant_payloads = true;
470 }
471
472 CreateStreams();
473 stream_observer.SetSendStream(send_stream_);
474
475 CreateFrameGeneratorCapturer();
476
477 Start();
478
479 EXPECT_EQ(kEventSignaled, stream_observer.Wait());
480
481 Stop();
482 DestroyStreams();
483}
pbos@webrtc.org744fbc72013-09-10 09:26:25 +0000484
stefan@webrtc.orgcb254aa2014-06-12 15:12:25 +0000485TEST_F(RampUpTest, SingleStream) {
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000486 RunRampUpTest(false, 1, 0, RtpExtension::kTOffset);
andresp@webrtc.orgc1480792014-03-20 03:23:55 +0000487}
pbos@webrtc.org744fbc72013-09-10 09:26:25 +0000488
stefan@webrtc.orgcb254aa2014-06-12 15:12:25 +0000489TEST_F(RampUpTest, Simulcast) {
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000490 RunRampUpTest(false, 3, 0, RtpExtension::kTOffset);
andresp@webrtc.orgc1480792014-03-20 03:23:55 +0000491}
492
stefan@webrtc.orgcb254aa2014-06-12 15:12:25 +0000493TEST_F(RampUpTest, SimulcastWithRtx) {
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000494 RunRampUpTest(true, 3, 0, RtpExtension::kTOffset);
mflodman@webrtc.orgeb16b812014-06-16 08:57:39 +0000495}
496
497TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
stefan@webrtc.org3d7da882014-07-08 13:59:46 +0000498 RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset);
andresp@webrtc.orgc1480792014-03-20 03:23:55 +0000499}
pbos@webrtc.org744fbc72013-09-10 09:26:25 +0000500
henrik.lundin@webrtc.org845862f2014-03-06 07:19:28 +0000501TEST_F(RampUpTest, UpDownUpOneStream) { RunRampUpDownUpTest(1, false); }
502
henrik.lundin@webrtc.org6ea4f632014-03-13 09:21:26 +0000503TEST_F(RampUpTest, UpDownUpThreeStreams) { RunRampUpDownUpTest(3, false); }
henrik.lundin@webrtc.org998cb8f2014-03-06 09:12:00 +0000504
henrik.lundin@webrtc.org6ea4f632014-03-13 09:21:26 +0000505TEST_F(RampUpTest, UpDownUpOneStreamRtx) { RunRampUpDownUpTest(1, true); }
henrik.lundin@webrtc.org998cb8f2014-03-06 09:12:00 +0000506
henrik.lundin@webrtc.org6ea4f632014-03-13 09:21:26 +0000507TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) { RunRampUpDownUpTest(3, true); }
henrik.lundin@webrtc.org845862f2014-03-06 07:19:28 +0000508
pbos@webrtc.org744fbc72013-09-10 09:26:25 +0000509} // namespace webrtc