1 /*
2 * Copyright (c) 2015 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
11 #ifdef ENABLE_RTC_EVENT_LOG
12
13 #include <string>
14 #include <utility>
15 #include <vector>
16
17 #include "testing/gtest/include/gtest/gtest.h"
18 #include "webrtc/base/buffer.h"
19 #include "webrtc/base/checks.h"
20 #include "webrtc/base/random.h"
21 #include "webrtc/base/scoped_ptr.h"
22 #include "webrtc/base/thread.h"
23 #include "webrtc/call.h"
24 #include "webrtc/call/rtc_event_log.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
26 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
27 #include "webrtc/system_wrappers/include/clock.h"
28 #include "webrtc/test/test_suite.h"
29 #include "webrtc/test/testsupport/fileutils.h"
30
31 // Files generated at build-time by the protobuf compiler.
32 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
33 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
34 #else
35 #include "webrtc/call/rtc_event_log.pb.h"
36 #endif
37
38 namespace webrtc {
39
40 namespace {
41
42 const RTPExtensionType kExtensionTypes[] = {
43 RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
44 RTPExtensionType::kRtpExtensionAudioLevel,
45 RTPExtensionType::kRtpExtensionAbsoluteSendTime,
46 RTPExtensionType::kRtpExtensionVideoRotation,
47 RTPExtensionType::kRtpExtensionTransportSequenceNumber};
48 const char* kExtensionNames[] = {RtpExtension::kTOffset,
49 RtpExtension::kAudioLevel,
50 RtpExtension::kAbsSendTime,
51 RtpExtension::kVideoRotation,
52 RtpExtension::kTransportSequenceNumber};
53 const size_t kNumExtensions = 5;
54
55 } // namespace
56
57 // TODO(terelius): Place this definition with other parsing functions?
GetRuntimeMediaType(rtclog::MediaType media_type)58 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
59 switch (media_type) {
60 case rtclog::MediaType::ANY:
61 return MediaType::ANY;
62 case rtclog::MediaType::AUDIO:
63 return MediaType::AUDIO;
64 case rtclog::MediaType::VIDEO:
65 return MediaType::VIDEO;
66 case rtclog::MediaType::DATA:
67 return MediaType::DATA;
68 }
69 RTC_NOTREACHED();
70 return MediaType::ANY;
71 }
72
73 // Checks that the event has a timestamp, a type and exactly the data field
74 // corresponding to the type.
IsValidBasicEvent(const rtclog::Event & event)75 ::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
76 if (!event.has_timestamp_us())
77 return ::testing::AssertionFailure() << "Event has no timestamp";
78 if (!event.has_type())
79 return ::testing::AssertionFailure() << "Event has no event type";
80 rtclog::Event_EventType type = event.type();
81 if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet())
82 return ::testing::AssertionFailure()
83 << "Event of type " << type << " has "
84 << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
85 if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet())
86 return ::testing::AssertionFailure()
87 << "Event of type " << type << " has "
88 << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
89 if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
90 event.has_audio_playout_event())
91 return ::testing::AssertionFailure()
92 << "Event of type " << type << " has "
93 << (event.has_audio_playout_event() ? "" : "no ")
94 << "audio_playout event";
95 if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) !=
96 event.has_video_receiver_config())
97 return ::testing::AssertionFailure()
98 << "Event of type " << type << " has "
99 << (event.has_video_receiver_config() ? "" : "no ")
100 << "receiver config";
101 if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
102 event.has_video_sender_config())
103 return ::testing::AssertionFailure()
104 << "Event of type " << type << " has "
105 << (event.has_video_sender_config() ? "" : "no ") << "sender config";
106 if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
107 event.has_audio_receiver_config()) {
108 return ::testing::AssertionFailure()
109 << "Event of type " << type << " has "
110 << (event.has_audio_receiver_config() ? "" : "no ")
111 << "audio receiver config";
112 }
113 if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) !=
114 event.has_audio_sender_config()) {
115 return ::testing::AssertionFailure()
116 << "Event of type " << type << " has "
117 << (event.has_audio_sender_config() ? "" : "no ")
118 << "audio sender config";
119 }
120 return ::testing::AssertionSuccess();
121 }
122
VerifyReceiveStreamConfig(const rtclog::Event & event,const VideoReceiveStream::Config & config)123 void VerifyReceiveStreamConfig(const rtclog::Event& event,
124 const VideoReceiveStream::Config& config) {
125 ASSERT_TRUE(IsValidBasicEvent(event));
126 ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
127 const rtclog::VideoReceiveConfig& receiver_config =
128 event.video_receiver_config();
129 // Check SSRCs.
130 ASSERT_TRUE(receiver_config.has_remote_ssrc());
131 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
132 ASSERT_TRUE(receiver_config.has_local_ssrc());
133 EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
134 // Check RTCP settings.
135 ASSERT_TRUE(receiver_config.has_rtcp_mode());
136 if (config.rtp.rtcp_mode == RtcpMode::kCompound)
137 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
138 receiver_config.rtcp_mode());
139 else
140 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
141 receiver_config.rtcp_mode());
142 ASSERT_TRUE(receiver_config.has_remb());
143 EXPECT_EQ(config.rtp.remb, receiver_config.remb());
144 // Check RTX map.
145 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
146 receiver_config.rtx_map_size());
147 for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
148 ASSERT_TRUE(rtx_map.has_payload_type());
149 ASSERT_TRUE(rtx_map.has_config());
150 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
151 const rtclog::RtxConfig& rtx_config = rtx_map.config();
152 const VideoReceiveStream::Config::Rtp::Rtx& rtx =
153 config.rtp.rtx.at(rtx_map.payload_type());
154 ASSERT_TRUE(rtx_config.has_rtx_ssrc());
155 ASSERT_TRUE(rtx_config.has_rtx_payload_type());
156 EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
157 EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
158 }
159 // Check header extensions.
160 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
161 receiver_config.header_extensions_size());
162 for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
163 ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
164 ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
165 const std::string& name = receiver_config.header_extensions(i).name();
166 int id = receiver_config.header_extensions(i).id();
167 EXPECT_EQ(config.rtp.extensions[i].id, id);
168 EXPECT_EQ(config.rtp.extensions[i].name, name);
169 }
170 // Check decoders.
171 ASSERT_EQ(static_cast<int>(config.decoders.size()),
172 receiver_config.decoders_size());
173 for (int i = 0; i < receiver_config.decoders_size(); i++) {
174 ASSERT_TRUE(receiver_config.decoders(i).has_name());
175 ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
176 const std::string& decoder_name = receiver_config.decoders(i).name();
177 int decoder_type = receiver_config.decoders(i).payload_type();
178 EXPECT_EQ(config.decoders[i].payload_name, decoder_name);
179 EXPECT_EQ(config.decoders[i].payload_type, decoder_type);
180 }
181 }
182
VerifySendStreamConfig(const rtclog::Event & event,const VideoSendStream::Config & config)183 void VerifySendStreamConfig(const rtclog::Event& event,
184 const VideoSendStream::Config& config) {
185 ASSERT_TRUE(IsValidBasicEvent(event));
186 ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
187 const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
188 // Check SSRCs.
189 ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()),
190 sender_config.ssrcs_size());
191 for (int i = 0; i < sender_config.ssrcs_size(); i++) {
192 EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
193 }
194 // Check header extensions.
195 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
196 sender_config.header_extensions_size());
197 for (int i = 0; i < sender_config.header_extensions_size(); i++) {
198 ASSERT_TRUE(sender_config.header_extensions(i).has_name());
199 ASSERT_TRUE(sender_config.header_extensions(i).has_id());
200 const std::string& name = sender_config.header_extensions(i).name();
201 int id = sender_config.header_extensions(i).id();
202 EXPECT_EQ(config.rtp.extensions[i].id, id);
203 EXPECT_EQ(config.rtp.extensions[i].name, name);
204 }
205 // Check RTX settings.
206 ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
207 sender_config.rtx_ssrcs_size());
208 for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
209 EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
210 }
211 if (sender_config.rtx_ssrcs_size() > 0) {
212 ASSERT_TRUE(sender_config.has_rtx_payload_type());
213 EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
214 }
215 // Check encoder.
216 ASSERT_TRUE(sender_config.has_encoder());
217 ASSERT_TRUE(sender_config.encoder().has_name());
218 ASSERT_TRUE(sender_config.encoder().has_payload_type());
219 EXPECT_EQ(config.encoder_settings.payload_name,
220 sender_config.encoder().name());
221 EXPECT_EQ(config.encoder_settings.payload_type,
222 sender_config.encoder().payload_type());
223 }
224
VerifyRtpEvent(const rtclog::Event & event,bool incoming,MediaType media_type,const uint8_t * header,size_t header_size,size_t total_size)225 void VerifyRtpEvent(const rtclog::Event& event,
226 bool incoming,
227 MediaType media_type,
228 const uint8_t* header,
229 size_t header_size,
230 size_t total_size) {
231 ASSERT_TRUE(IsValidBasicEvent(event));
232 ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
233 const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
234 ASSERT_TRUE(rtp_packet.has_incoming());
235 EXPECT_EQ(incoming, rtp_packet.incoming());
236 ASSERT_TRUE(rtp_packet.has_type());
237 EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
238 ASSERT_TRUE(rtp_packet.has_packet_length());
239 EXPECT_EQ(total_size, rtp_packet.packet_length());
240 ASSERT_TRUE(rtp_packet.has_header());
241 ASSERT_EQ(header_size, rtp_packet.header().size());
242 for (size_t i = 0; i < header_size; i++) {
243 EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
244 }
245 }
246
VerifyRtcpEvent(const rtclog::Event & event,bool incoming,MediaType media_type,const uint8_t * packet,size_t total_size)247 void VerifyRtcpEvent(const rtclog::Event& event,
248 bool incoming,
249 MediaType media_type,
250 const uint8_t* packet,
251 size_t total_size) {
252 ASSERT_TRUE(IsValidBasicEvent(event));
253 ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
254 const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
255 ASSERT_TRUE(rtcp_packet.has_incoming());
256 EXPECT_EQ(incoming, rtcp_packet.incoming());
257 ASSERT_TRUE(rtcp_packet.has_type());
258 EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
259 ASSERT_TRUE(rtcp_packet.has_packet_data());
260 ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
261 for (size_t i = 0; i < total_size; i++) {
262 EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
263 }
264 }
265
VerifyPlayoutEvent(const rtclog::Event & event,uint32_t ssrc)266 void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) {
267 ASSERT_TRUE(IsValidBasicEvent(event));
268 ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
269 const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
270 ASSERT_TRUE(playout_event.has_local_ssrc());
271 EXPECT_EQ(ssrc, playout_event.local_ssrc());
272 }
273
VerifyBweLossEvent(const rtclog::Event & event,int32_t bitrate,uint8_t fraction_loss,int32_t total_packets)274 void VerifyBweLossEvent(const rtclog::Event& event,
275 int32_t bitrate,
276 uint8_t fraction_loss,
277 int32_t total_packets) {
278 ASSERT_TRUE(IsValidBasicEvent(event));
279 ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
280 const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
281 ASSERT_TRUE(bwe_event.has_bitrate());
282 EXPECT_EQ(bitrate, bwe_event.bitrate());
283 ASSERT_TRUE(bwe_event.has_fraction_loss());
284 EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
285 ASSERT_TRUE(bwe_event.has_total_packets());
286 EXPECT_EQ(total_packets, bwe_event.total_packets());
287 }
288
VerifyLogStartEvent(const rtclog::Event & event)289 void VerifyLogStartEvent(const rtclog::Event& event) {
290 ASSERT_TRUE(IsValidBasicEvent(event));
291 EXPECT_EQ(rtclog::Event::LOG_START, event.type());
292 }
293
294 /*
295 * Bit number i of extension_bitvector is set to indicate the
296 * presence of extension number i from kExtensionTypes / kExtensionNames.
297 * The least significant bit extension_bitvector has number 0.
298 */
GenerateRtpPacket(uint32_t extensions_bitvector,uint32_t csrcs_count,uint8_t * packet,size_t packet_size,Random * prng)299 size_t GenerateRtpPacket(uint32_t extensions_bitvector,
300 uint32_t csrcs_count,
301 uint8_t* packet,
302 size_t packet_size,
303 Random* prng) {
304 RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
305 Clock* clock = Clock::GetRealTimeClock();
306
307 RTPSender rtp_sender(false, // bool audio
308 clock, // Clock* clock
309 nullptr, // Transport*
310 nullptr, // RtpAudioFeedback*
311 nullptr, // PacedSender*
312 nullptr, // PacketRouter*
313 nullptr, // SendTimeObserver*
314 nullptr, // BitrateStatisticsObserver*
315 nullptr, // FrameCountObserver*
316 nullptr); // SendSideDelayObserver*
317
318 std::vector<uint32_t> csrcs;
319 for (unsigned i = 0; i < csrcs_count; i++) {
320 csrcs.push_back(prng->Rand<uint32_t>());
321 }
322 rtp_sender.SetCsrcs(csrcs);
323 rtp_sender.SetSSRC(prng->Rand<uint32_t>());
324 rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
325 rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
326
327 for (unsigned i = 0; i < kNumExtensions; i++) {
328 if (extensions_bitvector & (1u << i)) {
329 rtp_sender.RegisterRtpHeaderExtension(kExtensionTypes[i], i + 1);
330 }
331 }
332
333 int8_t payload_type = prng->Rand(0, 127);
334 bool marker_bit = prng->Rand<bool>();
335 uint32_t capture_timestamp = prng->Rand<uint32_t>();
336 int64_t capture_time_ms = prng->Rand<uint32_t>();
337 bool timestamp_provided = prng->Rand<bool>();
338 bool inc_sequence_number = prng->Rand<bool>();
339
340 size_t header_size = rtp_sender.BuildRTPheader(
341 packet, payload_type, marker_bit, capture_timestamp, capture_time_ms,
342 timestamp_provided, inc_sequence_number);
343
344 for (size_t i = header_size; i < packet_size; i++) {
345 packet[i] = prng->Rand<uint8_t>();
346 }
347
348 return header_size;
349 }
350
GenerateRtcpPacket(Random * prng)351 rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(Random* prng) {
352 rtcp::ReportBlock report_block;
353 report_block.To(prng->Rand<uint32_t>()); // Remote SSRC.
354 report_block.WithFractionLost(prng->Rand(50));
355
356 rtcp::SenderReport sender_report;
357 sender_report.From(prng->Rand<uint32_t>()); // Sender SSRC.
358 sender_report.WithNtpSec(prng->Rand<uint32_t>());
359 sender_report.WithNtpFrac(prng->Rand<uint32_t>());
360 sender_report.WithPacketCount(prng->Rand<uint32_t>());
361 sender_report.WithReportBlock(report_block);
362
363 return sender_report.Build();
364 }
365
GenerateVideoReceiveConfig(uint32_t extensions_bitvector,VideoReceiveStream::Config * config,Random * prng)366 void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
367 VideoReceiveStream::Config* config,
368 Random* prng) {
369 // Create a map from a payload type to an encoder name.
370 VideoReceiveStream::Decoder decoder;
371 decoder.payload_type = prng->Rand(0, 127);
372 decoder.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
373 config->decoders.push_back(decoder);
374 // Add SSRCs for the stream.
375 config->rtp.remote_ssrc = prng->Rand<uint32_t>();
376 config->rtp.local_ssrc = prng->Rand<uint32_t>();
377 // Add extensions and settings for RTCP.
378 config->rtp.rtcp_mode =
379 prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
380 config->rtp.remb = prng->Rand<bool>();
381 // Add a map from a payload type to a new ssrc and a new payload type for RTX.
382 VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
383 rtx_pair.ssrc = prng->Rand<uint32_t>();
384 rtx_pair.payload_type = prng->Rand(0, 127);
385 config->rtp.rtx.insert(std::make_pair(prng->Rand(0, 127), rtx_pair));
386 // Add header extensions.
387 for (unsigned i = 0; i < kNumExtensions; i++) {
388 if (extensions_bitvector & (1u << i)) {
389 config->rtp.extensions.push_back(
390 RtpExtension(kExtensionNames[i], prng->Rand<int>()));
391 }
392 }
393 }
394
GenerateVideoSendConfig(uint32_t extensions_bitvector,VideoSendStream::Config * config,Random * prng)395 void GenerateVideoSendConfig(uint32_t extensions_bitvector,
396 VideoSendStream::Config* config,
397 Random* prng) {
398 // Create a map from a payload type to an encoder name.
399 config->encoder_settings.payload_type = prng->Rand(0, 127);
400 config->encoder_settings.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
401 // Add SSRCs for the stream.
402 config->rtp.ssrcs.push_back(prng->Rand<uint32_t>());
403 // Add a map from a payload type to new ssrcs and a new payload type for RTX.
404 config->rtp.rtx.ssrcs.push_back(prng->Rand<uint32_t>());
405 config->rtp.rtx.payload_type = prng->Rand(0, 127);
406 // Add header extensions.
407 for (unsigned i = 0; i < kNumExtensions; i++) {
408 if (extensions_bitvector & (1u << i)) {
409 config->rtp.extensions.push_back(
410 RtpExtension(kExtensionNames[i], prng->Rand<int>()));
411 }
412 }
413 }
414
415 // Test for the RtcEventLog class. Dumps some RTP packets and other events
416 // to disk, then reads them back to see if they match.
LogSessionAndReadBack(size_t rtp_count,size_t rtcp_count,size_t playout_count,size_t bwe_loss_count,uint32_t extensions_bitvector,uint32_t csrcs_count,unsigned int random_seed)417 void LogSessionAndReadBack(size_t rtp_count,
418 size_t rtcp_count,
419 size_t playout_count,
420 size_t bwe_loss_count,
421 uint32_t extensions_bitvector,
422 uint32_t csrcs_count,
423 unsigned int random_seed) {
424 ASSERT_LE(rtcp_count, rtp_count);
425 ASSERT_LE(playout_count, rtp_count);
426 ASSERT_LE(bwe_loss_count, rtp_count);
427 std::vector<rtc::Buffer> rtp_packets;
428 std::vector<rtc::scoped_ptr<rtcp::RawPacket> > rtcp_packets;
429 std::vector<size_t> rtp_header_sizes;
430 std::vector<uint32_t> playout_ssrcs;
431 std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates;
432
433 VideoReceiveStream::Config receiver_config(nullptr);
434 VideoSendStream::Config sender_config(nullptr);
435
436 Random prng(random_seed);
437
438 // Create rtp_count RTP packets containing random data.
439 for (size_t i = 0; i < rtp_count; i++) {
440 size_t packet_size = prng.Rand(1000, 1100);
441 rtp_packets.push_back(rtc::Buffer(packet_size));
442 size_t header_size =
443 GenerateRtpPacket(extensions_bitvector, csrcs_count,
444 rtp_packets[i].data(), packet_size, &prng);
445 rtp_header_sizes.push_back(header_size);
446 }
447 // Create rtcp_count RTCP packets containing random data.
448 for (size_t i = 0; i < rtcp_count; i++) {
449 rtcp_packets.push_back(GenerateRtcpPacket(&prng));
450 }
451 // Create playout_count random SSRCs to use when logging AudioPlayout events.
452 for (size_t i = 0; i < playout_count; i++) {
453 playout_ssrcs.push_back(prng.Rand<uint32_t>());
454 }
455 // Create bwe_loss_count random bitrate updates for BwePacketLoss.
456 for (size_t i = 0; i < bwe_loss_count; i++) {
457 bwe_loss_updates.push_back(
458 std::make_pair(prng.Rand<int32_t>(), prng.Rand<uint8_t>()));
459 }
460 // Create configurations for the video streams.
461 GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
462 GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
463 const int config_count = 2;
464
465 // Find the name of the current test, in order to use it as a temporary
466 // filename.
467 auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
468 const std::string temp_filename =
469 test::OutputPath() + test_info->test_case_name() + test_info->name();
470
471 // When log_dumper goes out of scope, it causes the log file to be flushed
472 // to disk.
473 {
474 rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
475 log_dumper->LogVideoReceiveStreamConfig(receiver_config);
476 log_dumper->LogVideoSendStreamConfig(sender_config);
477 size_t rtcp_index = 1;
478 size_t playout_index = 1;
479 size_t bwe_loss_index = 1;
480 for (size_t i = 1; i <= rtp_count; i++) {
481 log_dumper->LogRtpHeader(
482 (i % 2 == 0), // Every second packet is incoming.
483 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
484 rtp_packets[i - 1].data(), rtp_packets[i - 1].size());
485 if (i * rtcp_count >= rtcp_index * rtp_count) {
486 log_dumper->LogRtcpPacket(
487 rtcp_index % 2 == 0, // Every second packet is incoming
488 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
489 rtcp_packets[rtcp_index - 1]->Buffer(),
490 rtcp_packets[rtcp_index - 1]->Length());
491 rtcp_index++;
492 }
493 if (i * playout_count >= playout_index * rtp_count) {
494 log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
495 playout_index++;
496 }
497 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
498 log_dumper->LogBwePacketLossEvent(
499 bwe_loss_updates[bwe_loss_index - 1].first,
500 bwe_loss_updates[bwe_loss_index - 1].second, i);
501 bwe_loss_index++;
502 }
503 if (i == rtp_count / 2) {
504 log_dumper->StartLogging(temp_filename, 10000000);
505 }
506 }
507 }
508
509 // Read the generated file from disk.
510 rtclog::EventStream parsed_stream;
511
512 ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
513
514 // Verify that what we read back from the event log is the same as
515 // what we wrote down. For RTCP we log the full packets, but for
516 // RTP we should only log the header.
517 const int event_count = config_count + playout_count + bwe_loss_count +
518 rtcp_count + rtp_count + 1;
519 EXPECT_EQ(event_count, parsed_stream.stream_size());
520 VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
521 VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
522 size_t event_index = config_count;
523 size_t rtcp_index = 1;
524 size_t playout_index = 1;
525 size_t bwe_loss_index = 1;
526 for (size_t i = 1; i <= rtp_count; i++) {
527 VerifyRtpEvent(parsed_stream.stream(event_index),
528 (i % 2 == 0), // Every second packet is incoming.
529 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
530 rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
531 rtp_packets[i - 1].size());
532 event_index++;
533 if (i * rtcp_count >= rtcp_index * rtp_count) {
534 VerifyRtcpEvent(parsed_stream.stream(event_index),
535 rtcp_index % 2 == 0, // Every second packet is incoming.
536 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
537 rtcp_packets[rtcp_index - 1]->Buffer(),
538 rtcp_packets[rtcp_index - 1]->Length());
539 event_index++;
540 rtcp_index++;
541 }
542 if (i * playout_count >= playout_index * rtp_count) {
543 VerifyPlayoutEvent(parsed_stream.stream(event_index),
544 playout_ssrcs[playout_index - 1]);
545 event_index++;
546 playout_index++;
547 }
548 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
549 VerifyBweLossEvent(parsed_stream.stream(event_index),
550 bwe_loss_updates[bwe_loss_index - 1].first,
551 bwe_loss_updates[bwe_loss_index - 1].second, i);
552 event_index++;
553 bwe_loss_index++;
554 }
555 if (i == rtp_count / 2) {
556 VerifyLogStartEvent(parsed_stream.stream(event_index));
557 event_index++;
558 }
559 }
560
561 // Clean up temporary file - can be pretty slow.
562 remove(temp_filename.c_str());
563 }
564
TEST(RtcEventLogTest,LogSessionAndReadBack)565 TEST(RtcEventLogTest, LogSessionAndReadBack) {
566 // Log 5 RTP, 2 RTCP, 0 playout events and 0 BWE events
567 // with no header extensions or CSRCS.
568 LogSessionAndReadBack(5, 2, 0, 0, 0, 0, 321);
569
570 // Enable AbsSendTime and TransportSequenceNumbers.
571 uint32_t extensions = 0;
572 for (uint32_t i = 0; i < kNumExtensions; i++) {
573 if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime ||
574 kExtensionTypes[i] ==
575 RTPExtensionType::kRtpExtensionTransportSequenceNumber) {
576 extensions |= 1u << i;
577 }
578 }
579 LogSessionAndReadBack(8, 2, 0, 0, extensions, 0, 3141592653u);
580
581 extensions = (1u << kNumExtensions) - 1; // Enable all header extensions.
582 LogSessionAndReadBack(9, 2, 3, 2, extensions, 2, 2718281828u);
583
584 // Try all combinations of header extensions and up to 2 CSRCS.
585 for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) {
586 for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) {
587 LogSessionAndReadBack(5 + extensions, // Number of RTP packets.
588 2 + csrcs_count, // Number of RTCP packets.
589 3 + csrcs_count, // Number of playout events.
590 1 + csrcs_count, // Number of BWE loss events.
591 extensions, // Bit vector choosing extensions.
592 csrcs_count, // Number of contributing sources.
593 extensions * 3 + csrcs_count + 1); // Random seed.
594 }
595 }
596 }
597
598 // Tests that the event queue works correctly, i.e. drops old RTP, RTCP and
599 // debug events, but keeps config events even if they are older than the limit.
DropOldEvents(uint32_t extensions_bitvector,uint32_t csrcs_count,unsigned int random_seed)600 void DropOldEvents(uint32_t extensions_bitvector,
601 uint32_t csrcs_count,
602 unsigned int random_seed) {
603 rtc::Buffer old_rtp_packet;
604 rtc::Buffer recent_rtp_packet;
605 rtc::scoped_ptr<rtcp::RawPacket> old_rtcp_packet;
606 rtc::scoped_ptr<rtcp::RawPacket> recent_rtcp_packet;
607
608 VideoReceiveStream::Config receiver_config(nullptr);
609 VideoSendStream::Config sender_config(nullptr);
610
611 Random prng(random_seed);
612
613 // Create two RTP packets containing random data.
614 size_t packet_size = prng.Rand(1000, 1100);
615 old_rtp_packet.SetSize(packet_size);
616 GenerateRtpPacket(extensions_bitvector, csrcs_count, old_rtp_packet.data(),
617 packet_size, &prng);
618 packet_size = prng.Rand(1000, 1100);
619 recent_rtp_packet.SetSize(packet_size);
620 size_t recent_header_size =
621 GenerateRtpPacket(extensions_bitvector, csrcs_count,
622 recent_rtp_packet.data(), packet_size, &prng);
623
624 // Create two RTCP packets containing random data.
625 old_rtcp_packet = GenerateRtcpPacket(&prng);
626 recent_rtcp_packet = GenerateRtcpPacket(&prng);
627
628 // Create configurations for the video streams.
629 GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
630 GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
631
632 // Find the name of the current test, in order to use it as a temporary
633 // filename.
634 auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
635 const std::string temp_filename =
636 test::OutputPath() + test_info->test_case_name() + test_info->name();
637
638 // The log file will be flushed to disk when the log_dumper goes out of scope.
639 {
640 rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
641 // Reduce the time old events are stored to 50 ms.
642 log_dumper->SetBufferDuration(50000);
643 log_dumper->LogVideoReceiveStreamConfig(receiver_config);
644 log_dumper->LogVideoSendStreamConfig(sender_config);
645 log_dumper->LogRtpHeader(false, MediaType::AUDIO, old_rtp_packet.data(),
646 old_rtp_packet.size());
647 log_dumper->LogRtcpPacket(true, MediaType::AUDIO, old_rtcp_packet->Buffer(),
648 old_rtcp_packet->Length());
649 // Sleep 55 ms to let old events be removed from the queue.
650 rtc::Thread::SleepMs(55);
651 log_dumper->StartLogging(temp_filename, 10000000);
652 log_dumper->LogRtpHeader(true, MediaType::VIDEO, recent_rtp_packet.data(),
653 recent_rtp_packet.size());
654 log_dumper->LogRtcpPacket(false, MediaType::VIDEO,
655 recent_rtcp_packet->Buffer(),
656 recent_rtcp_packet->Length());
657 }
658
659 // Read the generated file from disk.
660 rtclog::EventStream parsed_stream;
661 ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
662
663 // Verify that what we read back from the event log is the same as
664 // what we wrote. Old RTP and RTCP events should have been discarded,
665 // but old configuration events should still be available.
666 EXPECT_EQ(5, parsed_stream.stream_size());
667 VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
668 VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
669 VerifyLogStartEvent(parsed_stream.stream(2));
670 VerifyRtpEvent(parsed_stream.stream(3), true, MediaType::VIDEO,
671 recent_rtp_packet.data(), recent_header_size,
672 recent_rtp_packet.size());
673 VerifyRtcpEvent(parsed_stream.stream(4), false, MediaType::VIDEO,
674 recent_rtcp_packet->Buffer(), recent_rtcp_packet->Length());
675
676 // Clean up temporary file - can be pretty slow.
677 remove(temp_filename.c_str());
678 }
679
TEST(RtcEventLogTest,DropOldEvents)680 TEST(RtcEventLogTest, DropOldEvents) {
681 // Enable all header extensions
682 uint32_t extensions = (1u << kNumExtensions) - 1;
683 uint32_t csrcs_count = 2;
684 DropOldEvents(extensions, csrcs_count, 141421356);
685 DropOldEvents(extensions, csrcs_count, 173205080);
686 }
687
688 } // namespace webrtc
689
690 #endif // ENABLE_RTC_EVENT_LOG
691