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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 |
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269 } | 269 } |
270 | 270 |
271 void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) { | 271 void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) { |
272 ASSERT_TRUE(IsValidBasicEvent(event)); | 272 ASSERT_TRUE(IsValidBasicEvent(event)); |
273 ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type()); | 273 ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type()); |
274 const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event(); | 274 const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event(); |
275 ASSERT_TRUE(playout_event.has_local_ssrc()); | 275 ASSERT_TRUE(playout_event.has_local_ssrc()); |
276 EXPECT_EQ(ssrc, playout_event.local_ssrc()); | 276 EXPECT_EQ(ssrc, playout_event.local_ssrc()); |
277 } | 277 } |
278 | 278 |
| 279 void VerifyBweLossEvent(const rtclog::Event& event, |
| 280 int32_t bitrate, |
| 281 uint8_t fraction_loss, |
| 282 int32_t total_packets) { |
| 283 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 284 ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type()); |
| 285 const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event(); |
| 286 ASSERT_TRUE(bwe_event.has_bitrate()); |
| 287 EXPECT_EQ(bitrate, bwe_event.bitrate()); |
| 288 ASSERT_TRUE(bwe_event.has_fraction_loss()); |
| 289 EXPECT_EQ(fraction_loss, bwe_event.fraction_loss()); |
| 290 ASSERT_TRUE(bwe_event.has_total_packets()); |
| 291 EXPECT_EQ(total_packets, bwe_event.total_packets()); |
| 292 } |
| 293 |
279 void VerifyLogStartEvent(const rtclog::Event& event) { | 294 void VerifyLogStartEvent(const rtclog::Event& event) { |
280 ASSERT_TRUE(IsValidBasicEvent(event)); | 295 ASSERT_TRUE(IsValidBasicEvent(event)); |
281 EXPECT_EQ(rtclog::Event::LOG_START, event.type()); | 296 EXPECT_EQ(rtclog::Event::LOG_START, event.type()); |
282 } | 297 } |
283 | 298 |
284 /* | 299 /* |
285 * Bit number i of extension_bitvector is set to indicate the | 300 * Bit number i of extension_bitvector is set to indicate the |
286 * presence of extension number i from kExtensionTypes / kExtensionNames. | 301 * presence of extension number i from kExtensionTypes / kExtensionNames. |
287 * The least significant bit extension_bitvector has number 0. | 302 * The least significant bit extension_bitvector has number 0. |
288 */ | 303 */ |
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391 RtpExtension(kExtensionNames[i], rand())); | 406 RtpExtension(kExtensionNames[i], rand())); |
392 } | 407 } |
393 } | 408 } |
394 } | 409 } |
395 | 410 |
396 // Test for the RtcEventLog class. Dumps some RTP packets and other events | 411 // Test for the RtcEventLog class. Dumps some RTP packets and other events |
397 // to disk, then reads them back to see if they match. | 412 // to disk, then reads them back to see if they match. |
398 void LogSessionAndReadBack(size_t rtp_count, | 413 void LogSessionAndReadBack(size_t rtp_count, |
399 size_t rtcp_count, | 414 size_t rtcp_count, |
400 size_t playout_count, | 415 size_t playout_count, |
| 416 size_t bwe_loss_count, |
401 uint32_t extensions_bitvector, | 417 uint32_t extensions_bitvector, |
402 uint32_t csrcs_count, | 418 uint32_t csrcs_count, |
403 unsigned int random_seed) { | 419 unsigned int random_seed) { |
404 ASSERT_LE(rtcp_count, rtp_count); | 420 ASSERT_LE(rtcp_count, rtp_count); |
405 ASSERT_LE(playout_count, rtp_count); | 421 ASSERT_LE(playout_count, rtp_count); |
| 422 ASSERT_LE(bwe_loss_count, rtp_count); |
406 std::vector<rtc::Buffer> rtp_packets; | 423 std::vector<rtc::Buffer> rtp_packets; |
407 std::vector<rtc::Buffer> rtcp_packets; | 424 std::vector<rtc::Buffer> rtcp_packets; |
408 std::vector<size_t> rtp_header_sizes; | 425 std::vector<size_t> rtp_header_sizes; |
409 std::vector<uint32_t> playout_ssrcs; | 426 std::vector<uint32_t> playout_ssrcs; |
| 427 std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates; |
410 | 428 |
411 VideoReceiveStream::Config receiver_config(nullptr); | 429 VideoReceiveStream::Config receiver_config(nullptr); |
412 VideoSendStream::Config sender_config(nullptr); | 430 VideoSendStream::Config sender_config(nullptr); |
413 | 431 |
414 srand(random_seed); | 432 srand(random_seed); |
415 | 433 |
416 // Create rtp_count RTP packets containing random data. | 434 // Create rtp_count RTP packets containing random data. |
417 for (size_t i = 0; i < rtp_count; i++) { | 435 for (size_t i = 0; i < rtp_count; i++) { |
418 size_t packet_size = 1000 + rand() % 64; | 436 size_t packet_size = 1000 + rand() % 64; |
419 rtp_packets.push_back(rtc::Buffer(packet_size)); | 437 rtp_packets.push_back(rtc::Buffer(packet_size)); |
420 size_t header_size = GenerateRtpPacket(extensions_bitvector, csrcs_count, | 438 size_t header_size = GenerateRtpPacket(extensions_bitvector, csrcs_count, |
421 rtp_packets[i].data(), packet_size); | 439 rtp_packets[i].data(), packet_size); |
422 rtp_header_sizes.push_back(header_size); | 440 rtp_header_sizes.push_back(header_size); |
423 } | 441 } |
424 // Create rtcp_count RTCP packets containing random data. | 442 // Create rtcp_count RTCP packets containing random data. |
425 for (size_t i = 0; i < rtcp_count; i++) { | 443 for (size_t i = 0; i < rtcp_count; i++) { |
426 size_t packet_size = 1000 + rand() % 64; | 444 size_t packet_size = 1000 + rand() % 64; |
427 rtcp_packets.push_back(rtc::Buffer(packet_size)); | 445 rtcp_packets.push_back(rtc::Buffer(packet_size)); |
428 GenerateRtcpPacket(rtcp_packets[i].data(), packet_size); | 446 GenerateRtcpPacket(rtcp_packets[i].data(), packet_size); |
429 } | 447 } |
430 // Create playout_count random SSRCs to use when logging AudioPlayout events. | 448 // Create playout_count random SSRCs to use when logging AudioPlayout events. |
431 for (size_t i = 0; i < playout_count; i++) { | 449 for (size_t i = 0; i < playout_count; i++) { |
432 playout_ssrcs.push_back(static_cast<uint32_t>(rand())); | 450 playout_ssrcs.push_back(static_cast<uint32_t>(rand())); |
433 } | 451 } |
| 452 // Create bwe_loss_count random bitrate updates for BwePacketLoss. |
| 453 for (size_t i = 0; i < bwe_loss_count; i++) { |
| 454 bwe_loss_updates.push_back(std::pair<int32_t, uint8_t>(rand(), rand())); |
| 455 } |
434 // Create configurations for the video streams. | 456 // Create configurations for the video streams. |
435 GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config); | 457 GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config); |
436 GenerateVideoSendConfig(extensions_bitvector, &sender_config); | 458 GenerateVideoSendConfig(extensions_bitvector, &sender_config); |
437 const int config_count = 2; | 459 const int config_count = 2; |
438 | 460 |
439 // Find the name of the current test, in order to use it as a temporary | 461 // Find the name of the current test, in order to use it as a temporary |
440 // filename. | 462 // filename. |
441 auto test_info = ::testing::UnitTest::GetInstance()->current_test_info(); | 463 auto test_info = ::testing::UnitTest::GetInstance()->current_test_info(); |
442 const std::string temp_filename = | 464 const std::string temp_filename = |
443 test::OutputPath() + test_info->test_case_name() + test_info->name(); | 465 test::OutputPath() + test_info->test_case_name() + test_info->name(); |
444 | 466 |
445 // When log_dumper goes out of scope, it causes the log file to be flushed | 467 // When log_dumper goes out of scope, it causes the log file to be flushed |
446 // to disk. | 468 // to disk. |
447 { | 469 { |
448 rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create()); | 470 rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create()); |
449 log_dumper->LogVideoReceiveStreamConfig(receiver_config); | 471 log_dumper->LogVideoReceiveStreamConfig(receiver_config); |
450 log_dumper->LogVideoSendStreamConfig(sender_config); | 472 log_dumper->LogVideoSendStreamConfig(sender_config); |
451 size_t rtcp_index = 1, playout_index = 1; | 473 size_t rtcp_index = 1; |
| 474 size_t playout_index = 1; |
| 475 size_t bwe_loss_index = 1; |
452 for (size_t i = 1; i <= rtp_count; i++) { | 476 for (size_t i = 1; i <= rtp_count; i++) { |
453 log_dumper->LogRtpHeader( | 477 log_dumper->LogRtpHeader( |
454 (i % 2 == 0), // Every second packet is incoming. | 478 (i % 2 == 0), // Every second packet is incoming. |
455 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, | 479 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, |
456 rtp_packets[i - 1].data(), rtp_packets[i - 1].size()); | 480 rtp_packets[i - 1].data(), rtp_packets[i - 1].size()); |
457 if (i * rtcp_count >= rtcp_index * rtp_count) { | 481 if (i * rtcp_count >= rtcp_index * rtp_count) { |
458 log_dumper->LogRtcpPacket( | 482 log_dumper->LogRtcpPacket( |
459 rtcp_index % 2 == 0, // Every second packet is incoming | 483 rtcp_index % 2 == 0, // Every second packet is incoming |
460 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, | 484 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, |
461 rtcp_packets[rtcp_index - 1].data(), | 485 rtcp_packets[rtcp_index - 1].data(), |
462 rtcp_packets[rtcp_index - 1].size()); | 486 rtcp_packets[rtcp_index - 1].size()); |
463 rtcp_index++; | 487 rtcp_index++; |
464 } | 488 } |
465 if (i * playout_count >= playout_index * rtp_count) { | 489 if (i * playout_count >= playout_index * rtp_count) { |
466 log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]); | 490 log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]); |
467 playout_index++; | 491 playout_index++; |
468 } | 492 } |
| 493 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) { |
| 494 log_dumper->LogBwePacketLossEvent( |
| 495 bwe_loss_updates[bwe_loss_index - 1].first, |
| 496 bwe_loss_updates[bwe_loss_index - 1].second, i); |
| 497 bwe_loss_index++; |
| 498 } |
469 if (i == rtp_count / 2) { | 499 if (i == rtp_count / 2) { |
470 log_dumper->StartLogging(temp_filename, 10000000); | 500 log_dumper->StartLogging(temp_filename, 10000000); |
471 } | 501 } |
472 } | 502 } |
473 } | 503 } |
474 | 504 |
475 // Read the generated file from disk. | 505 // Read the generated file from disk. |
476 rtclog::EventStream parsed_stream; | 506 rtclog::EventStream parsed_stream; |
477 | 507 |
478 ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream)); | 508 ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream)); |
479 | 509 |
480 // Verify that what we read back from the event log is the same as | 510 // Verify that what we read back from the event log is the same as |
481 // what we wrote down. For RTCP we log the full packets, but for | 511 // what we wrote down. For RTCP we log the full packets, but for |
482 // RTP we should only log the header. | 512 // RTP we should only log the header. |
483 const int event_count = | 513 const int event_count = config_count + playout_count + bwe_loss_count + |
484 config_count + playout_count + rtcp_count + rtp_count + 1; | 514 rtcp_count + rtp_count + 1; |
485 EXPECT_EQ(event_count, parsed_stream.stream_size()); | 515 EXPECT_EQ(event_count, parsed_stream.stream_size()); |
486 VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config); | 516 VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config); |
487 VerifySendStreamConfig(parsed_stream.stream(1), sender_config); | 517 VerifySendStreamConfig(parsed_stream.stream(1), sender_config); |
488 size_t event_index = config_count, rtcp_index = 1, playout_index = 1; | 518 size_t event_index = config_count; |
| 519 size_t rtcp_index = 1; |
| 520 size_t playout_index = 1; |
| 521 size_t bwe_loss_index = 1; |
489 for (size_t i = 1; i <= rtp_count; i++) { | 522 for (size_t i = 1; i <= rtp_count; i++) { |
490 VerifyRtpEvent(parsed_stream.stream(event_index), | 523 VerifyRtpEvent(parsed_stream.stream(event_index), |
491 (i % 2 == 0), // Every second packet is incoming. | 524 (i % 2 == 0), // Every second packet is incoming. |
492 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, | 525 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, |
493 rtp_packets[i - 1].data(), rtp_header_sizes[i - 1], | 526 rtp_packets[i - 1].data(), rtp_header_sizes[i - 1], |
494 rtp_packets[i - 1].size()); | 527 rtp_packets[i - 1].size()); |
495 event_index++; | 528 event_index++; |
496 if (i * rtcp_count >= rtcp_index * rtp_count) { | 529 if (i * rtcp_count >= rtcp_index * rtp_count) { |
497 VerifyRtcpEvent(parsed_stream.stream(event_index), | 530 VerifyRtcpEvent(parsed_stream.stream(event_index), |
498 rtcp_index % 2 == 0, // Every second packet is incoming. | 531 rtcp_index % 2 == 0, // Every second packet is incoming. |
499 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, | 532 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, |
500 rtcp_packets[rtcp_index - 1].data(), | 533 rtcp_packets[rtcp_index - 1].data(), |
501 rtcp_packets[rtcp_index - 1].size()); | 534 rtcp_packets[rtcp_index - 1].size()); |
502 event_index++; | 535 event_index++; |
503 rtcp_index++; | 536 rtcp_index++; |
504 } | 537 } |
505 if (i * playout_count >= playout_index * rtp_count) { | 538 if (i * playout_count >= playout_index * rtp_count) { |
506 VerifyPlayoutEvent(parsed_stream.stream(event_index), | 539 VerifyPlayoutEvent(parsed_stream.stream(event_index), |
507 playout_ssrcs[playout_index - 1]); | 540 playout_ssrcs[playout_index - 1]); |
508 event_index++; | 541 event_index++; |
509 playout_index++; | 542 playout_index++; |
510 } | 543 } |
| 544 if (i * bwe_loss_count >= bwe_loss_index * rtp_count) { |
| 545 VerifyBweLossEvent(parsed_stream.stream(event_index), |
| 546 bwe_loss_updates[bwe_loss_index - 1].first, |
| 547 bwe_loss_updates[bwe_loss_index - 1].second, i); |
| 548 event_index++; |
| 549 bwe_loss_index++; |
| 550 } |
511 if (i == rtp_count / 2) { | 551 if (i == rtp_count / 2) { |
512 VerifyLogStartEvent(parsed_stream.stream(event_index)); | 552 VerifyLogStartEvent(parsed_stream.stream(event_index)); |
513 event_index++; | 553 event_index++; |
514 } | 554 } |
515 } | 555 } |
516 | 556 |
517 // Clean up temporary file - can be pretty slow. | 557 // Clean up temporary file - can be pretty slow. |
518 remove(temp_filename.c_str()); | 558 remove(temp_filename.c_str()); |
519 } | 559 } |
520 | 560 |
521 TEST(RtcEventLogTest, LogSessionAndReadBack) { | 561 TEST(RtcEventLogTest, LogSessionAndReadBack) { |
522 // Log 5 RTP, 2 RTCP, and 0 playout events with no header extensions or CSRCS. | 562 // Log 5 RTP, 2 RTCP, 0 playout events and 0 BWE events |
523 LogSessionAndReadBack(5, 2, 0, 0, 0, 321); | 563 // with no header extensions or CSRCS. |
| 564 LogSessionAndReadBack(5, 2, 0, 0, 0, 0, 321); |
524 | 565 |
525 // Enable AbsSendTime and TransportSequenceNumbers | 566 // Enable AbsSendTime and TransportSequenceNumbers. |
526 uint32_t extensions = 0; | 567 uint32_t extensions = 0; |
527 for (uint32_t i = 0; i < kNumExtensions; i++) { | 568 for (uint32_t i = 0; i < kNumExtensions; i++) { |
528 if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime || | 569 if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime || |
529 kExtensionTypes[i] == | 570 kExtensionTypes[i] == |
530 RTPExtensionType::kRtpExtensionTransportSequenceNumber) { | 571 RTPExtensionType::kRtpExtensionTransportSequenceNumber) { |
531 extensions |= 1u << i; | 572 extensions |= 1u << i; |
532 } | 573 } |
533 } | 574 } |
534 LogSessionAndReadBack(8, 2, 0, extensions, 0, 3141592653u); | 575 LogSessionAndReadBack(8, 2, 0, 0, extensions, 0, 3141592653u); |
535 | 576 |
536 extensions = (1u << kNumExtensions) - 1; // Enable all header extensions | 577 extensions = (1u << kNumExtensions) - 1; // Enable all header extensions. |
537 LogSessionAndReadBack(9, 2, 3, extensions, 2, 2718281828u); | 578 LogSessionAndReadBack(9, 2, 3, 2, extensions, 2, 2718281828u); |
538 | 579 |
539 // Try all combinations of header extensions and up to 2 CSRCS. | 580 // Try all combinations of header extensions and up to 2 CSRCS. |
540 for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) { | 581 for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) { |
541 for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) { | 582 for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) { |
542 LogSessionAndReadBack(5 + extensions, // Number of RTP packets. | 583 LogSessionAndReadBack(5 + extensions, // Number of RTP packets. |
543 2 + csrcs_count, // Number of RTCP packets. | 584 2 + csrcs_count, // Number of RTCP packets. |
544 3 + csrcs_count, // Number of playout events | 585 3 + csrcs_count, // Number of playout events. |
545 extensions, // Bit vector choosing extensions | 586 1 + csrcs_count, // Number of BWE loss events. |
546 csrcs_count, // Number of contributing sources | 587 extensions, // Bit vector choosing extensions. |
| 588 csrcs_count, // Number of contributing sources. |
547 rand()); | 589 rand()); |
548 } | 590 } |
549 } | 591 } |
550 } | 592 } |
551 | 593 |
552 // Tests that the event queue works correctly, i.e. drops old RTP, RTCP and | 594 // Tests that the event queue works correctly, i.e. drops old RTP, RTCP and |
553 // debug events, but keeps config events even if they are older than the limit. | 595 // debug events, but keeps config events even if they are older than the limit. |
554 void DropOldEvents(uint32_t extensions_bitvector, | 596 void DropOldEvents(uint32_t extensions_bitvector, |
555 uint32_t csrcs_count, | 597 uint32_t csrcs_count, |
556 unsigned int random_seed) { | 598 unsigned int random_seed) { |
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638 // Enable all header extensions | 680 // Enable all header extensions |
639 uint32_t extensions = (1u << kNumExtensions) - 1; | 681 uint32_t extensions = (1u << kNumExtensions) - 1; |
640 uint32_t csrcs_count = 2; | 682 uint32_t csrcs_count = 2; |
641 DropOldEvents(extensions, csrcs_count, 141421356); | 683 DropOldEvents(extensions, csrcs_count, 141421356); |
642 DropOldEvents(extensions, csrcs_count, 173205080); | 684 DropOldEvents(extensions, csrcs_count, 173205080); |
643 } | 685 } |
644 | 686 |
645 } // namespace webrtc | 687 } // namespace webrtc |
646 | 688 |
647 #endif // ENABLE_RTC_EVENT_LOG | 689 #endif // ENABLE_RTC_EVENT_LOG |
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