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Side by Side Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 1768773002: New parser for event log. Manually parse the outermost EventStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifdef ENABLE_RTC_EVENT_LOG 11 #ifdef ENABLE_RTC_EVENT_LOG
12 12
13 #include <map> 13 #include <map>
14 #include <memory> 14 #include <memory>
15 #include <string> 15 #include <string>
16 #include <utility> 16 #include <utility>
17 #include <vector> 17 #include <vector>
18 18
19 #include "testing/gtest/include/gtest/gtest.h" 19 #include "testing/gtest/include/gtest/gtest.h"
20 #include "webrtc/base/buffer.h" 20 #include "webrtc/base/buffer.h"
21 #include "webrtc/base/checks.h" 21 #include "webrtc/base/checks.h"
22 #include "webrtc/base/random.h" 22 #include "webrtc/base/random.h"
23 #include "webrtc/call.h" 23 #include "webrtc/call.h"
24 #include "webrtc/call/rtc_event_log.h" 24 #include "webrtc/call/rtc_event_log.h"
25 #include "webrtc/call/rtc_event_log_parser.h"
26 #include "webrtc/call/rtc_event_log_unittest_helper.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" 27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" 28 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
27 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" 29 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
28 #include "webrtc/system_wrappers/include/clock.h" 30 #include "webrtc/system_wrappers/include/clock.h"
29 #include "webrtc/test/test_suite.h" 31 #include "webrtc/test/test_suite.h"
30 #include "webrtc/test/testsupport/fileutils.h" 32 #include "webrtc/test/testsupport/fileutils.h"
31 33
32 // Files generated at build-time by the protobuf compiler. 34 // Files generated at build-time by the protobuf compiler.
33 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD 35 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
34 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" 36 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
(...skipping 11 matching lines...) Expand all
46 RTPExtensionType::kRtpExtensionAbsoluteSendTime, 48 RTPExtensionType::kRtpExtensionAbsoluteSendTime,
47 RTPExtensionType::kRtpExtensionVideoRotation, 49 RTPExtensionType::kRtpExtensionVideoRotation,
48 RTPExtensionType::kRtpExtensionTransportSequenceNumber}; 50 RTPExtensionType::kRtpExtensionTransportSequenceNumber};
49 const char* kExtensionNames[] = {RtpExtension::kTOffset, 51 const char* kExtensionNames[] = {RtpExtension::kTOffset,
50 RtpExtension::kAudioLevel, 52 RtpExtension::kAudioLevel,
51 RtpExtension::kAbsSendTime, 53 RtpExtension::kAbsSendTime,
52 RtpExtension::kVideoRotation, 54 RtpExtension::kVideoRotation,
53 RtpExtension::kTransportSequenceNumber}; 55 RtpExtension::kTransportSequenceNumber};
54 const size_t kNumExtensions = 5; 56 const size_t kNumExtensions = 5;
55 57
56 } // namespace 58 void PrintActualEvents(const ParsedRtcEventLog& parsed_log) {
57 59 std::map<int, size_t> actual_event_counts;
58 // TODO(terelius): Place this definition with other parsing functions? 60 for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
59 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { 61 actual_event_counts[parsed_log.GetEventType(i)]++;
60 switch (media_type) {
61 case rtclog::MediaType::ANY:
62 return MediaType::ANY;
63 case rtclog::MediaType::AUDIO:
64 return MediaType::AUDIO;
65 case rtclog::MediaType::VIDEO:
66 return MediaType::VIDEO;
67 case rtclog::MediaType::DATA:
68 return MediaType::DATA;
69 } 62 }
70 RTC_NOTREACHED(); 63 printf("Actual events: ");
71 return MediaType::ANY; 64 for (auto kv : actual_event_counts) {
65 printf("%d_count = %zu, ", kv.first, kv.second);
66 }
67 printf("\n");
68 for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
69 printf("%4d ", parsed_log.GetEventType(i));
70 }
71 printf("\n");
72 } 72 }
73 73
74 // Checks that the event has a timestamp, a type and exactly the data field 74 void PrintExpectedEvents(size_t rtp_count,
75 // corresponding to the type. 75 size_t rtcp_count,
76 ::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) { 76 size_t playout_count,
77 if (!event.has_timestamp_us()) 77 size_t bwe_loss_count) {
78 return ::testing::AssertionFailure() << "Event has no timestamp"; 78 printf(
79 if (!event.has_type()) 79 "Expected events: rtp_count = %zu, rtcp_count = %zu,"
80 return ::testing::AssertionFailure() << "Event has no event type"; 80 "playout_count = %zu, bwe_loss_count = %zu\n",
81 rtclog::Event_EventType type = event.type(); 81 rtp_count, rtcp_count, playout_count, bwe_loss_count);
82 if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet()) 82 size_t rtcp_index = 1, playout_index = 1, bwe_loss_index = 1;
83 return ::testing::AssertionFailure() 83 printf("strt cfg cfg ");
84 << "Event of type " << type << " has " 84 for (size_t i = 1; i <= rtp_count; i++) {
85 << (event.has_rtp_packet() ? "" : "no ") << "RTP packet"; 85 printf(" rtp ");
86 if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet()) 86 if (i * rtcp_count >= rtcp_index * rtp_count) {
87 return ::testing::AssertionFailure() 87 printf("rtcp ");
88 << "Event of type " << type << " has " 88 rtcp_index++;
89 << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet"; 89 }
90 if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) != 90 if (i * playout_count >= playout_index * rtp_count) {
91 event.has_audio_playout_event()) 91 printf("play ");
92 return ::testing::AssertionFailure() 92 playout_index++;
93 << "Event of type " << type << " has " 93 }
94 << (event.has_audio_playout_event() ? "" : "no ") 94 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
95 << "audio_playout event"; 95 printf("loss ");
96 if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) != 96 bwe_loss_index++;
97 event.has_video_receiver_config()) 97 }
98 return ::testing::AssertionFailure()
99 << "Event of type " << type << " has "
100 << (event.has_video_receiver_config() ? "" : "no ")
101 << "receiver config";
102 if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
103 event.has_video_sender_config())
104 return ::testing::AssertionFailure()
105 << "Event of type " << type << " has "
106 << (event.has_video_sender_config() ? "" : "no ") << "sender config";
107 if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
108 event.has_audio_receiver_config()) {
109 return ::testing::AssertionFailure()
110 << "Event of type " << type << " has "
111 << (event.has_audio_receiver_config() ? "" : "no ")
112 << "audio receiver config";
113 } 98 }
114 if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) != 99 printf("end \n");
115 event.has_audio_sender_config()) {
116 return ::testing::AssertionFailure()
117 << "Event of type " << type << " has "
118 << (event.has_audio_sender_config() ? "" : "no ")
119 << "audio sender config";
120 }
121 return ::testing::AssertionSuccess();
122 } 100 }
123 101 } // namespace
124 void VerifyReceiveStreamConfig(const rtclog::Event& event,
125 const VideoReceiveStream::Config& config) {
126 ASSERT_TRUE(IsValidBasicEvent(event));
127 ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
128 const rtclog::VideoReceiveConfig& receiver_config =
129 event.video_receiver_config();
130 // Check SSRCs.
131 ASSERT_TRUE(receiver_config.has_remote_ssrc());
132 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
133 ASSERT_TRUE(receiver_config.has_local_ssrc());
134 EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
135 // Check RTCP settings.
136 ASSERT_TRUE(receiver_config.has_rtcp_mode());
137 if (config.rtp.rtcp_mode == RtcpMode::kCompound)
138 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
139 receiver_config.rtcp_mode());
140 else
141 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
142 receiver_config.rtcp_mode());
143 ASSERT_TRUE(receiver_config.has_remb());
144 EXPECT_EQ(config.rtp.remb, receiver_config.remb());
145 // Check RTX map.
146 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
147 receiver_config.rtx_map_size());
148 for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
149 ASSERT_TRUE(rtx_map.has_payload_type());
150 ASSERT_TRUE(rtx_map.has_config());
151 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
152 const rtclog::RtxConfig& rtx_config = rtx_map.config();
153 const VideoReceiveStream::Config::Rtp::Rtx& rtx =
154 config.rtp.rtx.at(rtx_map.payload_type());
155 ASSERT_TRUE(rtx_config.has_rtx_ssrc());
156 ASSERT_TRUE(rtx_config.has_rtx_payload_type());
157 EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
158 EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
159 }
160 // Check header extensions.
161 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
162 receiver_config.header_extensions_size());
163 for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
164 ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
165 ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
166 const std::string& name = receiver_config.header_extensions(i).name();
167 int id = receiver_config.header_extensions(i).id();
168 EXPECT_EQ(config.rtp.extensions[i].id, id);
169 EXPECT_EQ(config.rtp.extensions[i].name, name);
170 }
171 // Check decoders.
172 ASSERT_EQ(static_cast<int>(config.decoders.size()),
173 receiver_config.decoders_size());
174 for (int i = 0; i < receiver_config.decoders_size(); i++) {
175 ASSERT_TRUE(receiver_config.decoders(i).has_name());
176 ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
177 const std::string& decoder_name = receiver_config.decoders(i).name();
178 int decoder_type = receiver_config.decoders(i).payload_type();
179 EXPECT_EQ(config.decoders[i].payload_name, decoder_name);
180 EXPECT_EQ(config.decoders[i].payload_type, decoder_type);
181 }
182 }
183
184 void VerifySendStreamConfig(const rtclog::Event& event,
185 const VideoSendStream::Config& config) {
186 ASSERT_TRUE(IsValidBasicEvent(event));
187 ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
188 const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
189 // Check SSRCs.
190 ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()),
191 sender_config.ssrcs_size());
192 for (int i = 0; i < sender_config.ssrcs_size(); i++) {
193 EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
194 }
195 // Check header extensions.
196 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
197 sender_config.header_extensions_size());
198 for (int i = 0; i < sender_config.header_extensions_size(); i++) {
199 ASSERT_TRUE(sender_config.header_extensions(i).has_name());
200 ASSERT_TRUE(sender_config.header_extensions(i).has_id());
201 const std::string& name = sender_config.header_extensions(i).name();
202 int id = sender_config.header_extensions(i).id();
203 EXPECT_EQ(config.rtp.extensions[i].id, id);
204 EXPECT_EQ(config.rtp.extensions[i].name, name);
205 }
206 // Check RTX settings.
207 ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
208 sender_config.rtx_ssrcs_size());
209 for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
210 EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
211 }
212 if (sender_config.rtx_ssrcs_size() > 0) {
213 ASSERT_TRUE(sender_config.has_rtx_payload_type());
214 EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
215 }
216 // Check encoder.
217 ASSERT_TRUE(sender_config.has_encoder());
218 ASSERT_TRUE(sender_config.encoder().has_name());
219 ASSERT_TRUE(sender_config.encoder().has_payload_type());
220 EXPECT_EQ(config.encoder_settings.payload_name,
221 sender_config.encoder().name());
222 EXPECT_EQ(config.encoder_settings.payload_type,
223 sender_config.encoder().payload_type());
224 }
225
226 void VerifyRtpEvent(const rtclog::Event& event,
227 PacketDirection direction,
228 MediaType media_type,
229 const uint8_t* header,
230 size_t header_size,
231 size_t total_size) {
232 ASSERT_TRUE(IsValidBasicEvent(event));
233 ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
234 const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
235 ASSERT_TRUE(rtp_packet.has_incoming());
236 EXPECT_EQ(direction == kIncomingPacket, rtp_packet.incoming());
237 ASSERT_TRUE(rtp_packet.has_type());
238 EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
239 ASSERT_TRUE(rtp_packet.has_packet_length());
240 EXPECT_EQ(total_size, rtp_packet.packet_length());
241 ASSERT_TRUE(rtp_packet.has_header());
242 ASSERT_EQ(header_size, rtp_packet.header().size());
243 for (size_t i = 0; i < header_size; i++) {
244 EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
245 }
246 }
247
248 void VerifyRtcpEvent(const rtclog::Event& event,
249 PacketDirection direction,
250 MediaType media_type,
251 const uint8_t* packet,
252 size_t total_size) {
253 ASSERT_TRUE(IsValidBasicEvent(event));
254 ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
255 const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
256 ASSERT_TRUE(rtcp_packet.has_incoming());
257 EXPECT_EQ(direction == kIncomingPacket, rtcp_packet.incoming());
258 ASSERT_TRUE(rtcp_packet.has_type());
259 EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
260 ASSERT_TRUE(rtcp_packet.has_packet_data());
261 ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
262 for (size_t i = 0; i < total_size; i++) {
263 EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
264 }
265 }
266
267 void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) {
268 ASSERT_TRUE(IsValidBasicEvent(event));
269 ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
270 const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
271 ASSERT_TRUE(playout_event.has_local_ssrc());
272 EXPECT_EQ(ssrc, playout_event.local_ssrc());
273 }
274
275 void VerifyBweLossEvent(const rtclog::Event& event,
276 int32_t bitrate,
277 uint8_t fraction_loss,
278 int32_t total_packets) {
279 ASSERT_TRUE(IsValidBasicEvent(event));
280 ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
281 const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
282 ASSERT_TRUE(bwe_event.has_bitrate());
283 EXPECT_EQ(bitrate, bwe_event.bitrate());
284 ASSERT_TRUE(bwe_event.has_fraction_loss());
285 EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
286 ASSERT_TRUE(bwe_event.has_total_packets());
287 EXPECT_EQ(total_packets, bwe_event.total_packets());
288 }
289
290 void VerifyLogStartEvent(const rtclog::Event& event) {
291 ASSERT_TRUE(IsValidBasicEvent(event));
292 EXPECT_EQ(rtclog::Event::LOG_START, event.type());
293 }
294
295 void VerifyLogEndEvent(const rtclog::Event& event) {
296 ASSERT_TRUE(IsValidBasicEvent(event));
297 EXPECT_EQ(rtclog::Event::LOG_END, event.type());
298 }
299 102
300 /* 103 /*
301 * Bit number i of extension_bitvector is set to indicate the 104 * Bit number i of extension_bitvector is set to indicate the
302 * presence of extension number i from kExtensionTypes / kExtensionNames. 105 * presence of extension number i from kExtensionTypes / kExtensionNames.
303 * The least significant bit extension_bitvector has number 0. 106 * The least significant bit extension_bitvector has number 0.
304 */ 107 */
305 size_t GenerateRtpPacket(uint32_t extensions_bitvector, 108 size_t GenerateRtpPacket(uint32_t extensions_bitvector,
306 uint32_t csrcs_count, 109 uint32_t csrcs_count,
307 uint8_t* packet, 110 uint8_t* packet,
308 size_t packet_size, 111 size_t packet_size,
(...skipping 207 matching lines...) Expand 10 before | Expand all | Expand 10 after
516 } 319 }
517 if (i == rtp_count / 2) { 320 if (i == rtp_count / 2) {
518 log_dumper->StartLogging(temp_filename, 10000000); 321 log_dumper->StartLogging(temp_filename, 10000000);
519 fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); 322 fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
520 } 323 }
521 } 324 }
522 log_dumper->StopLogging(); 325 log_dumper->StopLogging();
523 } 326 }
524 327
525 // Read the generated file from disk. 328 // Read the generated file from disk.
526 rtclog::EventStream parsed_stream; 329 ParsedRtcEventLog parsed_log;
527 330
528 ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream)); 331 ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
529 332
530 // Verify that what we read back from the event log is the same as 333 // Verify that what we read back from the event log is the same as
531 // what we wrote down. For RTCP we log the full packets, but for 334 // what we wrote down. For RTCP we log the full packets, but for
532 // RTP we should only log the header. 335 // RTP we should only log the header.
533 const int event_count = config_count + playout_count + bwe_loss_count + 336 const size_t event_count = config_count + playout_count + bwe_loss_count +
534 rtcp_count + rtp_count + 2; 337 rtcp_count + rtp_count + 2;
535 EXPECT_GE(1000, event_count); // The events must fit in the message queue. 338 EXPECT_GE(1000u, event_count); // The events must fit in the message queue.
536 EXPECT_EQ(event_count, parsed_stream.stream_size()); 339 EXPECT_EQ(event_count, parsed_log.GetNumberOfEvents());
537 if (event_count != parsed_stream.stream_size()) { 340 if (event_count != parsed_log.GetNumberOfEvents()) {
538 // Print the expected and actual event types for easier debugging. 341 // Print the expected and actual event types for easier debugging.
539 std::map<int, size_t> actual_event_counts; 342 PrintActualEvents(parsed_log);
540 for (size_t i = 0; i < static_cast<size_t>(parsed_stream.stream_size()); 343 PrintExpectedEvents(rtp_count, rtcp_count, playout_count, bwe_loss_count);
541 i++) {
542 actual_event_counts[parsed_stream.stream(i).type()]++;
543 }
544 printf("Actual events: ");
545 for (auto kv : actual_event_counts) {
546 printf("%d_count = %zu, ", kv.first, kv.second);
547 }
548 printf("\n");
549 for (size_t i = 0; i < static_cast<size_t>(parsed_stream.stream_size());
550 i++) {
551 printf("%4d ", parsed_stream.stream(i).type());
552 }
553 printf("\n");
554 printf(
555 "Expected events: rtp_count = %zu, rtcp_count = %zu,"
556 "playout_count = %zu, bwe_loss_count = %zu\n",
557 rtp_count, rtcp_count, playout_count, bwe_loss_count);
558 size_t rtcp_index = 1, playout_index = 1, bwe_loss_index = 1;
559 printf("strt cfg cfg ");
560 for (size_t i = 1; i <= rtp_count; i++) {
561 printf(" rtp ");
562 if (i * rtcp_count >= rtcp_index * rtp_count) {
563 printf("rtcp ");
564 rtcp_index++;
565 }
566 if (i * playout_count >= playout_index * rtp_count) {
567 printf("play ");
568 playout_index++;
569 }
570 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
571 printf("loss ");
572 bwe_loss_index++;
573 }
574 }
575 printf("\n");
576 } 344 }
577 VerifyLogStartEvent(parsed_stream.stream(0)); 345 RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
578 VerifyReceiveStreamConfig(parsed_stream.stream(1), receiver_config); 346 RtcEventLogTestHelper::VerifyReceiveStreamConfig(parsed_log, 1,
579 VerifySendStreamConfig(parsed_stream.stream(2), sender_config); 347 receiver_config);
348 RtcEventLogTestHelper::VerifySendStreamConfig(parsed_log, 2, sender_config);
580 size_t event_index = config_count + 1; 349 size_t event_index = config_count + 1;
581 size_t rtcp_index = 1; 350 size_t rtcp_index = 1;
582 size_t playout_index = 1; 351 size_t playout_index = 1;
583 size_t bwe_loss_index = 1; 352 size_t bwe_loss_index = 1;
584 for (size_t i = 1; i <= rtp_count; i++) { 353 for (size_t i = 1; i <= rtp_count; i++) {
585 VerifyRtpEvent(parsed_stream.stream(event_index), 354 RtcEventLogTestHelper::VerifyRtpEvent(
586 (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket, 355 parsed_log, event_index,
587 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, 356 (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
588 rtp_packets[i - 1].data(), rtp_header_sizes[i - 1], 357 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
589 rtp_packets[i - 1].size()); 358 rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
359 rtp_packets[i - 1].size());
590 event_index++; 360 event_index++;
591 if (i * rtcp_count >= rtcp_index * rtp_count) { 361 if (i * rtcp_count >= rtcp_index * rtp_count) {
592 VerifyRtcpEvent(parsed_stream.stream(event_index), 362 RtcEventLogTestHelper::VerifyRtcpEvent(
593 (rtcp_index % 2 == 0) ? kIncomingPacket : kOutgoingPacket, 363 parsed_log, event_index,
594 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, 364 rtcp_index % 2 == 0 ? kIncomingPacket : kOutgoingPacket,
595 rtcp_packets[rtcp_index - 1].data(), 365 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
596 rtcp_packets[rtcp_index - 1].size()); 366 rtcp_packets[rtcp_index - 1].data(),
367 rtcp_packets[rtcp_index - 1].size());
597 event_index++; 368 event_index++;
598 rtcp_index++; 369 rtcp_index++;
599 } 370 }
600 if (i * playout_count >= playout_index * rtp_count) { 371 if (i * playout_count >= playout_index * rtp_count) {
601 VerifyPlayoutEvent(parsed_stream.stream(event_index), 372 RtcEventLogTestHelper::VerifyPlayoutEvent(
602 playout_ssrcs[playout_index - 1]); 373 parsed_log, event_index, playout_ssrcs[playout_index - 1]);
603 event_index++; 374 event_index++;
604 playout_index++; 375 playout_index++;
605 } 376 }
606 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) { 377 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
607 VerifyBweLossEvent(parsed_stream.stream(event_index), 378 RtcEventLogTestHelper::VerifyBweLossEvent(
608 bwe_loss_updates[bwe_loss_index - 1].first, 379 parsed_log, event_index, bwe_loss_updates[bwe_loss_index - 1].first,
609 bwe_loss_updates[bwe_loss_index - 1].second, i); 380 bwe_loss_updates[bwe_loss_index - 1].second, i);
610 event_index++; 381 event_index++;
611 bwe_loss_index++; 382 bwe_loss_index++;
612 } 383 }
613 } 384 }
614 385
615 // Clean up temporary file - can be pretty slow. 386 // Clean up temporary file - can be pretty slow.
616 remove(temp_filename.c_str()); 387 remove(temp_filename.c_str());
617 } 388 }
618 389
619 TEST(RtcEventLogTest, LogSessionAndReadBack) { 390 TEST(RtcEventLogTest, LogSessionAndReadBack) {
(...skipping 56 matching lines...) Expand 10 before | Expand all | Expand 10 after
676 log_dumper->StartLogging(temp_filename, 10000000); 447 log_dumper->StartLogging(temp_filename, 10000000);
677 fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); 448 fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
678 449
679 log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO, 450 log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO,
680 rtcp_packet.data(), rtcp_packet.size()); 451 rtcp_packet.data(), rtcp_packet.size());
681 fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); 452 fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
682 453
683 log_dumper->StopLogging(); 454 log_dumper->StopLogging();
684 455
685 // Read the generated file from disk. 456 // Read the generated file from disk.
686 rtclog::EventStream parsed_stream; 457 ParsedRtcEventLog parsed_log;
687 ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream)); 458 ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
688 459
689 // Verify that what we read back from the event log is the same as 460 // Verify that what we read back from the event log is the same as
690 // what we wrote down. 461 // what we wrote down.
691 EXPECT_EQ(4, parsed_stream.stream_size()); 462 EXPECT_EQ(4u, parsed_log.GetNumberOfEvents());
692 463
693 VerifyLogStartEvent(parsed_stream.stream(0)); 464 RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
694 465
695 VerifyRtpEvent(parsed_stream.stream(1), kIncomingPacket, MediaType::VIDEO, 466 RtcEventLogTestHelper::VerifyRtpEvent(parsed_log, 1, kIncomingPacket,
696 rtp_packet.data(), header_size, rtp_packet.size()); 467 MediaType::VIDEO, rtp_packet.data(),
468 header_size, rtp_packet.size());
697 469
698 VerifyRtcpEvent(parsed_stream.stream(2), kOutgoingPacket, MediaType::VIDEO, 470 RtcEventLogTestHelper::VerifyRtcpEvent(parsed_log, 2, kOutgoingPacket,
699 rtcp_packet.data(), rtcp_packet.size()); 471 MediaType::VIDEO, rtcp_packet.data(),
472 rtcp_packet.size());
700 473
701 VerifyLogEndEvent(parsed_stream.stream(3)); 474 RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3);
702 475
703 // Clean up temporary file - can be pretty slow. 476 // Clean up temporary file - can be pretty slow.
704 remove(temp_filename.c_str()); 477 remove(temp_filename.c_str());
705 } 478 }
706 } // namespace webrtc 479 } // namespace webrtc
707 480
708 #endif // ENABLE_RTC_EVENT_LOG 481 #endif // ENABLE_RTC_EVENT_LOG
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