Moved RtcEventLog files from call/ to logging/ (new top level dir)

webrtc/call/rtc_event_log_unittest.cc

Index: webrtc/call/rtc_event_log_unittest.cc
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
deleted file mode 100644
index 6c4ec6382e31bc0248684f28b7eb419f488ae315..0000000000000000000000000000000000000000
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ /dev/null
@@ -1,460 +0,0 @@
-/*
- *  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.
- */
-
-#include <map>
-#include <memory>
-#include <string>
-#include <utility>
-#include <vector>
-
-#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/gtest.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.SetMediaSsrc(prng->Rand<uint32_t>());  // Remote SSRC.
-  report_block.SetFractionLost(prng->Rand(50));
-
-  rtcp::SenderReport sender_report;
-  sender_report.SetSenderSsrc(prng->Rand<uint32_t>());
-  sender_report.SetNtp(NtpTime(prng->Rand<uint32_t>(), prng->Rand<uint32_t>()));
-  sender_report.SetPacketCount(prng->Rand<uint32_t>());
-  sender_report.AddReportBlock(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