| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifdef ENABLE_RTC_EVENT_LOG |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| #include "webrtc/base/buffer.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/random.h" |
| #include "webrtc/call.h" |
| #include "webrtc/call/rtc_event_log.h" |
| #include "webrtc/call/rtc_event_log_parser.h" |
| #include "webrtc/call/rtc_event_log_unittest_helper.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/test/test_suite.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| |
| // Files generated at build-time by the protobuf compiler. |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" |
| #else |
| #include "webrtc/call/rtc_event_log.pb.h" |
| #endif |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| const RTPExtensionType kExtensionTypes[] = { |
| RTPExtensionType::kRtpExtensionTransmissionTimeOffset, |
| RTPExtensionType::kRtpExtensionAudioLevel, |
| RTPExtensionType::kRtpExtensionAbsoluteSendTime, |
| RTPExtensionType::kRtpExtensionVideoRotation, |
| RTPExtensionType::kRtpExtensionTransportSequenceNumber}; |
| const char* kExtensionNames[] = { |
| RtpExtension::kTimestampOffsetUri, RtpExtension::kAudioLevelUri, |
| RtpExtension::kAbsSendTimeUri, RtpExtension::kVideoRotationUri, |
| RtpExtension::kTransportSequenceNumberUri}; |
| const size_t kNumExtensions = 5; |
| |
| void PrintActualEvents(const ParsedRtcEventLog& parsed_log) { |
| std::map<int, size_t> actual_event_counts; |
| for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) { |
| actual_event_counts[parsed_log.GetEventType(i)]++; |
| } |
| printf("Actual events: "); |
| for (auto kv : actual_event_counts) { |
| printf("%d_count = %zu, ", kv.first, kv.second); |
| } |
| printf("\n"); |
| for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) { |
| printf("%4d ", parsed_log.GetEventType(i)); |
| } |
| printf("\n"); |
| } |
| |
| void PrintExpectedEvents(size_t rtp_count, |
| size_t rtcp_count, |
| size_t playout_count, |
| size_t bwe_loss_count) { |
| printf( |
| "Expected events: rtp_count = %zu, rtcp_count = %zu," |
| "playout_count = %zu, bwe_loss_count = %zu\n", |
| rtp_count, rtcp_count, playout_count, bwe_loss_count); |
| size_t rtcp_index = 1, playout_index = 1, bwe_loss_index = 1; |
| printf("strt cfg cfg "); |
| for (size_t i = 1; i <= rtp_count; i++) { |
| printf(" rtp "); |
| if (i * rtcp_count >= rtcp_index * rtp_count) { |
| printf("rtcp "); |
| rtcp_index++; |
| } |
| if (i * playout_count >= playout_index * rtp_count) { |
| printf("play "); |
| playout_index++; |
| } |
| if (i * bwe_loss_count >= bwe_loss_index * rtp_count) { |
| printf("loss "); |
| bwe_loss_index++; |
| } |
| } |
| printf("end \n"); |
| } |
| } // namespace |
| |
| /* |
| * Bit number i of extension_bitvector is set to indicate the |
| * presence of extension number i from kExtensionTypes / kExtensionNames. |
| * The least significant bit extension_bitvector has number 0. |
| */ |
| RtpPacketToSend GenerateRtpPacket(const RtpHeaderExtensionMap* extensions, |
| uint32_t csrcs_count, |
| size_t packet_size, |
| Random* prng) { |
| RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions); |
| |
| std::vector<uint32_t> csrcs; |
| for (unsigned i = 0; i < csrcs_count; i++) { |
| csrcs.push_back(prng->Rand<uint32_t>()); |
| } |
| |
| RtpPacketToSend rtp_packet(extensions, packet_size); |
| rtp_packet.SetPayloadType(prng->Rand(127)); |
| rtp_packet.SetMarker(prng->Rand<bool>()); |
| rtp_packet.SetSequenceNumber(prng->Rand<uint16_t>()); |
| rtp_packet.SetSsrc(prng->Rand<uint32_t>()); |
| rtp_packet.SetTimestamp(prng->Rand<uint32_t>()); |
| rtp_packet.SetCsrcs(csrcs); |
| |
| rtp_packet.SetExtension<TransmissionOffset>(prng->Rand(0x00ffffff)); |
| rtp_packet.SetExtension<AudioLevel>(prng->Rand<bool>(), prng->Rand(127)); |
| rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand<int32_t>()); |
| rtp_packet.SetExtension<VideoOrientation>(prng->Rand(2)); |
| rtp_packet.SetExtension<TransportSequenceNumber>(prng->Rand<uint16_t>()); |
| |
| size_t payload_size = packet_size - rtp_packet.headers_size(); |
| uint8_t* payload = rtp_packet.AllocatePayload(payload_size); |
| for (size_t i = 0; i < payload_size; i++) { |
| payload[i] = prng->Rand<uint8_t>(); |
| } |
| return rtp_packet; |
| } |
| |
| rtc::Buffer GenerateRtcpPacket(Random* prng) { |
| rtcp::ReportBlock report_block; |
| report_block.To(prng->Rand<uint32_t>()); // Remote SSRC. |
| report_block.WithFractionLost(prng->Rand(50)); |
| |
| rtcp::SenderReport sender_report; |
| sender_report.From(prng->Rand<uint32_t>()); // Sender SSRC. |
| sender_report.WithNtp( |
| NtpTime(prng->Rand<uint32_t>(), prng->Rand<uint32_t>())); |
| sender_report.WithPacketCount(prng->Rand<uint32_t>()); |
| sender_report.WithReportBlock(report_block); |
| |
| return sender_report.Build(); |
| } |
| |
| void GenerateVideoReceiveConfig(uint32_t extensions_bitvector, |
| VideoReceiveStream::Config* config, |
| Random* prng) { |
| // Create a map from a payload type to an encoder name. |
| VideoReceiveStream::Decoder decoder; |
| decoder.payload_type = prng->Rand(0, 127); |
| decoder.payload_name = (prng->Rand<bool>() ? "VP8" : "H264"); |
| config->decoders.push_back(decoder); |
| // Add SSRCs for the stream. |
| config->rtp.remote_ssrc = prng->Rand<uint32_t>(); |
| config->rtp.local_ssrc = prng->Rand<uint32_t>(); |
| // Add extensions and settings for RTCP. |
| config->rtp.rtcp_mode = |
| prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize; |
| config->rtp.remb = prng->Rand<bool>(); |
| // Add a map from a payload type to a new ssrc and a new payload type for RTX. |
| VideoReceiveStream::Config::Rtp::Rtx rtx_pair; |
| rtx_pair.ssrc = prng->Rand<uint32_t>(); |
| rtx_pair.payload_type = prng->Rand(0, 127); |
| config->rtp.rtx.insert(std::make_pair(prng->Rand(0, 127), rtx_pair)); |
| // Add header extensions. |
| for (unsigned i = 0; i < kNumExtensions; i++) { |
| if (extensions_bitvector & (1u << i)) { |
| config->rtp.extensions.push_back( |
| RtpExtension(kExtensionNames[i], prng->Rand<int>())); |
| } |
| } |
| } |
| |
| void GenerateVideoSendConfig(uint32_t extensions_bitvector, |
| VideoSendStream::Config* config, |
| Random* prng) { |
| // Create a map from a payload type to an encoder name. |
| config->encoder_settings.payload_type = prng->Rand(0, 127); |
| config->encoder_settings.payload_name = (prng->Rand<bool>() ? "VP8" : "H264"); |
| // Add SSRCs for the stream. |
| config->rtp.ssrcs.push_back(prng->Rand<uint32_t>()); |
| // Add a map from a payload type to new ssrcs and a new payload type for RTX. |
| config->rtp.rtx.ssrcs.push_back(prng->Rand<uint32_t>()); |
| config->rtp.rtx.payload_type = prng->Rand(0, 127); |
| // Add header extensions. |
| for (unsigned i = 0; i < kNumExtensions; i++) { |
| if (extensions_bitvector & (1u << i)) { |
| config->rtp.extensions.push_back( |
| RtpExtension(kExtensionNames[i], prng->Rand<int>())); |
| } |
| } |
| } |
| |
| // Test for the RtcEventLog class. Dumps some RTP packets and other events |
| // to disk, then reads them back to see if they match. |
| void 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) { |
| ASSERT_LE(rtcp_count, rtp_count); |
| ASSERT_LE(playout_count, rtp_count); |
| ASSERT_LE(bwe_loss_count, rtp_count); |
| std::vector<RtpPacketToSend> rtp_packets; |
| std::vector<rtc::Buffer> rtcp_packets; |
| std::vector<uint32_t> playout_ssrcs; |
| std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates; |
| |
| VideoReceiveStream::Config receiver_config(nullptr); |
| VideoSendStream::Config sender_config(nullptr); |
| |
| Random prng(random_seed); |
| |
| // Initialize rtp header extensions to be used in generated rtp packets. |
| RtpHeaderExtensionMap extensions; |
| for (unsigned i = 0; i < kNumExtensions; i++) { |
| if (extensions_bitvector & (1u << i)) { |
| extensions.Register(kExtensionTypes[i], i + 1); |
| } |
| } |
| // Create rtp_count RTP packets containing random data. |
| for (size_t i = 0; i < rtp_count; i++) { |
| size_t packet_size = prng.Rand(1000, 1100); |
| rtp_packets.push_back( |
| GenerateRtpPacket(&extensions, csrcs_count, packet_size, &prng)); |
| } |
| // Create rtcp_count RTCP packets containing random data. |
| for (size_t i = 0; i < rtcp_count; i++) { |
| rtcp_packets.push_back(GenerateRtcpPacket(&prng)); |
| } |
| // Create playout_count random SSRCs to use when logging AudioPlayout events. |
| for (size_t i = 0; i < playout_count; i++) { |
| playout_ssrcs.push_back(prng.Rand<uint32_t>()); |
| } |
| // Create bwe_loss_count random bitrate updates for BwePacketLoss. |
| for (size_t i = 0; i < bwe_loss_count; i++) { |
| bwe_loss_updates.push_back( |
| std::make_pair(prng.Rand<int32_t>(), prng.Rand<uint8_t>())); |
| } |
| // Create configurations for the video streams. |
| GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng); |
| GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng); |
| const int config_count = 2; |
| |
| // Find the name of the current test, in order to use it as a temporary |
| // filename. |
| auto test_info = ::testing::UnitTest::GetInstance()->current_test_info(); |
| const std::string temp_filename = |
| test::OutputPath() + test_info->test_case_name() + test_info->name(); |
| |
| // When log_dumper goes out of scope, it causes the log file to be flushed |
| // to disk. |
| { |
| SimulatedClock fake_clock(prng.Rand<uint32_t>()); |
| std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create(&fake_clock)); |
| log_dumper->LogVideoReceiveStreamConfig(receiver_config); |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| log_dumper->LogVideoSendStreamConfig(sender_config); |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| size_t rtcp_index = 1; |
| size_t playout_index = 1; |
| size_t bwe_loss_index = 1; |
| for (size_t i = 1; i <= rtp_count; i++) { |
| log_dumper->LogRtpHeader( |
| (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket, |
| (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, |
| rtp_packets[i - 1].data(), rtp_packets[i - 1].size()); |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| if (i * rtcp_count >= rtcp_index * rtp_count) { |
| log_dumper->LogRtcpPacket( |
| (rtcp_index % 2 == 0) ? kIncomingPacket : kOutgoingPacket, |
| rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, |
| rtcp_packets[rtcp_index - 1].data(), |
| rtcp_packets[rtcp_index - 1].size()); |
| rtcp_index++; |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| } |
| if (i * playout_count >= playout_index * rtp_count) { |
| log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]); |
| playout_index++; |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| } |
| if (i * bwe_loss_count >= bwe_loss_index * rtp_count) { |
| log_dumper->LogBwePacketLossEvent( |
| bwe_loss_updates[bwe_loss_index - 1].first, |
| bwe_loss_updates[bwe_loss_index - 1].second, i); |
| bwe_loss_index++; |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| } |
| if (i == rtp_count / 2) { |
| log_dumper->StartLogging(temp_filename, 10000000); |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| } |
| } |
| log_dumper->StopLogging(); |
| } |
| |
| // Read the generated file from disk. |
| ParsedRtcEventLog parsed_log; |
| |
| ASSERT_TRUE(parsed_log.ParseFile(temp_filename)); |
| |
| // Verify that what we read back from the event log is the same as |
| // what we wrote down. For RTCP we log the full packets, but for |
| // RTP we should only log the header. |
| const size_t event_count = config_count + playout_count + bwe_loss_count + |
| rtcp_count + rtp_count + 2; |
| EXPECT_GE(1000u, event_count); // The events must fit in the message queue. |
| EXPECT_EQ(event_count, parsed_log.GetNumberOfEvents()); |
| if (event_count != parsed_log.GetNumberOfEvents()) { |
| // Print the expected and actual event types for easier debugging. |
| PrintActualEvents(parsed_log); |
| PrintExpectedEvents(rtp_count, rtcp_count, playout_count, bwe_loss_count); |
| } |
| RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0); |
| RtcEventLogTestHelper::VerifyReceiveStreamConfig(parsed_log, 1, |
| receiver_config); |
| RtcEventLogTestHelper::VerifySendStreamConfig(parsed_log, 2, sender_config); |
| size_t event_index = config_count + 1; |
| size_t rtcp_index = 1; |
| size_t playout_index = 1; |
| size_t bwe_loss_index = 1; |
| for (size_t i = 1; i <= rtp_count; i++) { |
| RtcEventLogTestHelper::VerifyRtpEvent( |
| parsed_log, event_index, |
| (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket, |
| (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, |
| rtp_packets[i - 1].data(), rtp_packets[i - 1].headers_size(), |
| rtp_packets[i - 1].size()); |
| event_index++; |
| if (i * rtcp_count >= rtcp_index * rtp_count) { |
| RtcEventLogTestHelper::VerifyRtcpEvent( |
| parsed_log, event_index, |
| rtcp_index % 2 == 0 ? kIncomingPacket : kOutgoingPacket, |
| rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, |
| rtcp_packets[rtcp_index - 1].data(), |
| rtcp_packets[rtcp_index - 1].size()); |
| event_index++; |
| rtcp_index++; |
| } |
| if (i * playout_count >= playout_index * rtp_count) { |
| RtcEventLogTestHelper::VerifyPlayoutEvent( |
| parsed_log, event_index, playout_ssrcs[playout_index - 1]); |
| event_index++; |
| playout_index++; |
| } |
| if (i * bwe_loss_count >= bwe_loss_index * rtp_count) { |
| RtcEventLogTestHelper::VerifyBweLossEvent( |
| parsed_log, event_index, bwe_loss_updates[bwe_loss_index - 1].first, |
| bwe_loss_updates[bwe_loss_index - 1].second, i); |
| event_index++; |
| bwe_loss_index++; |
| } |
| } |
| |
| // Clean up temporary file - can be pretty slow. |
| remove(temp_filename.c_str()); |
| } |
| |
| TEST(RtcEventLogTest, LogSessionAndReadBack) { |
| // Log 5 RTP, 2 RTCP, 0 playout events and 0 BWE events |
| // with no header extensions or CSRCS. |
| LogSessionAndReadBack(5, 2, 0, 0, 0, 0, 321); |
| |
| // Enable AbsSendTime and TransportSequenceNumbers. |
| uint32_t extensions = 0; |
| for (uint32_t i = 0; i < kNumExtensions; i++) { |
| if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime || |
| kExtensionTypes[i] == |
| RTPExtensionType::kRtpExtensionTransportSequenceNumber) { |
| extensions |= 1u << i; |
| } |
| } |
| LogSessionAndReadBack(8, 2, 0, 0, extensions, 0, 3141592653u); |
| |
| extensions = (1u << kNumExtensions) - 1; // Enable all header extensions. |
| LogSessionAndReadBack(9, 2, 3, 2, extensions, 2, 2718281828u); |
| |
| // Try all combinations of header extensions and up to 2 CSRCS. |
| for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) { |
| for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) { |
| LogSessionAndReadBack(5 + extensions, // Number of RTP packets. |
| 2 + csrcs_count, // Number of RTCP packets. |
| 3 + csrcs_count, // Number of playout events. |
| 1 + csrcs_count, // Number of BWE loss events. |
| extensions, // Bit vector choosing extensions. |
| csrcs_count, // Number of contributing sources. |
| extensions * 3 + csrcs_count + 1); // Random seed. |
| } |
| } |
| } |
| |
| TEST(RtcEventLogTest, LogEventAndReadBack) { |
| Random prng(987654321); |
| |
| // Create one RTP and one RTCP packet containing random data. |
| size_t packet_size = prng.Rand(1000, 1100); |
| RtpPacketToSend rtp_packet = |
| GenerateRtpPacket(nullptr, 0, packet_size, &prng); |
| rtc::Buffer rtcp_packet = GenerateRtcpPacket(&prng); |
| |
| // Find the name of the current test, in order to use it as a temporary |
| // filename. |
| auto test_info = ::testing::UnitTest::GetInstance()->current_test_info(); |
| const std::string temp_filename = |
| test::OutputPath() + test_info->test_case_name() + test_info->name(); |
| |
| // Add RTP, start logging, add RTCP and then stop logging |
| SimulatedClock fake_clock(prng.Rand<uint32_t>()); |
| std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create(&fake_clock)); |
| |
| log_dumper->LogRtpHeader(kIncomingPacket, MediaType::VIDEO, rtp_packet.data(), |
| rtp_packet.size()); |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| |
| log_dumper->StartLogging(temp_filename, 10000000); |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| |
| log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO, |
| rtcp_packet.data(), rtcp_packet.size()); |
| fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000)); |
| |
| log_dumper->StopLogging(); |
| |
| // Read the generated file from disk. |
| ParsedRtcEventLog parsed_log; |
| ASSERT_TRUE(parsed_log.ParseFile(temp_filename)); |
| |
| // Verify that what we read back from the event log is the same as |
| // what we wrote down. |
| EXPECT_EQ(4u, parsed_log.GetNumberOfEvents()); |
| |
| RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0); |
| |
| RtcEventLogTestHelper::VerifyRtpEvent( |
| parsed_log, 1, kIncomingPacket, MediaType::VIDEO, rtp_packet.data(), |
| rtp_packet.headers_size(), rtp_packet.size()); |
| |
| RtcEventLogTestHelper::VerifyRtcpEvent(parsed_log, 2, kOutgoingPacket, |
| MediaType::VIDEO, rtcp_packet.data(), |
| rtcp_packet.size()); |
| |
| RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3); |
| |
| // Clean up temporary file - can be pretty slow. |
| remove(temp_filename.c_str()); |
| } |
| } // namespace webrtc |
| |
| #endif // ENABLE_RTC_EVENT_LOG |