OLD | NEW |
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" |
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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, |
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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) { |
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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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