video/rtc_event_log_unittest.cc

/*
 *  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 <stdio.h>
#include <string>
#include <vector>

#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/video/rtc_event_log.h"

// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/video/rtc_event_log.pb.h"
#else
#include "webrtc/video/rtc_event_log.pb.h"
#endif

namespace webrtc {

// TODO(terelius): Place this definition with other parsing functions?
MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
  switch (media_type) {
    case rtclog::MediaType::ANY:
      return MediaType::ANY;
    case rtclog::MediaType::AUDIO:
      return MediaType::AUDIO;
    case rtclog::MediaType::VIDEO:
      return MediaType::VIDEO;
    case rtclog::MediaType::DATA:
      return MediaType::DATA;
  }
  RTC_NOTREACHED();
  return MediaType::ANY;
}

// Checks that the event has a timestamp, a type and exactly the data field
// corresponding to the type.
::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
  if (!event.has_timestamp_us())
    return ::testing::AssertionFailure() << "Event has no timestamp";
  if (!event.has_type())
    return ::testing::AssertionFailure() << "Event has no event type";
  rtclog::Event_EventType type = event.type();
  if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet())
    return ::testing::AssertionFailure()
           << "Event of type " << type << " has "
           << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
  if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet())
    return ::testing::AssertionFailure()
           << "Event of type " << type << " has "
           << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
  if ((type == rtclog::Event::DEBUG_EVENT) != event.has_debug_event())
    return ::testing::AssertionFailure()
           << "Event of type " << type << " has "
           << (event.has_debug_event() ? "" : "no ") << "debug event";
  if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) !=
      event.has_video_receiver_config())
    return ::testing::AssertionFailure()
           << "Event of type " << type << " has "
           << (event.has_video_receiver_config() ? "" : "no ")
           << "receiver config";
  if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
      event.has_video_sender_config())
    return ::testing::AssertionFailure()
           << "Event of type " << type << " has "
           << (event.has_video_sender_config() ? "" : "no ") << "sender config";
  if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
      event.has_audio_receiver_config()) {
    return ::testing::AssertionFailure()
           << "Event of type " << type << " has "
           << (event.has_audio_receiver_config() ? "" : "no ")
           << "audio receiver config";
  }
  if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) !=
      event.has_audio_sender_config()) {
    return ::testing::AssertionFailure()
           << "Event of type " << type << " has "
           << (event.has_audio_sender_config() ? "" : "no ")
           << "audio sender config";
  }
  return ::testing::AssertionSuccess();
}

void VerifyReceiveStreamConfig(const rtclog::Event& event,
                               const VideoReceiveStream::Config& config) {
  ASSERT_TRUE(IsValidBasicEvent(event));
  ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
  const rtclog::VideoReceiveConfig& receiver_config =
      event.video_receiver_config();
  // Check SSRCs.
  ASSERT_TRUE(receiver_config.has_remote_ssrc());
  EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
  ASSERT_TRUE(receiver_config.has_local_ssrc());
  EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
  // Check RTCP settings.
  ASSERT_TRUE(receiver_config.has_rtcp_mode());
  if (config.rtp.rtcp_mode == newapi::kRtcpCompound)
    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
              receiver_config.rtcp_mode());
  else
    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
              receiver_config.rtcp_mode());
  ASSERT_TRUE(receiver_config.has_receiver_reference_time_report());
  EXPECT_EQ(config.rtp.rtcp_xr.receiver_reference_time_report,
            receiver_config.receiver_reference_time_report());
  ASSERT_TRUE(receiver_config.has_remb());
  EXPECT_EQ(config.rtp.remb, receiver_config.remb());
  // Check RTX map.
  ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
            receiver_config.rtx_map_size());
  for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
    ASSERT_TRUE(rtx_map.has_payload_type());
    ASSERT_TRUE(rtx_map.has_config());
    EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
    const rtclog::RtxConfig& rtx_config = rtx_map.config();
    const VideoReceiveStream::Config::Rtp::Rtx& rtx =
        config.rtp.rtx.at(rtx_map.payload_type());
    ASSERT_TRUE(rtx_config.has_rtx_ssrc());
    ASSERT_TRUE(rtx_config.has_rtx_payload_type());
    EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
    EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
  }
  // Check header extensions.
  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
            receiver_config.header_extensions_size());
  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
    ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
    ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
    const std::string& name = receiver_config.header_extensions(i).name();
    int id = receiver_config.header_extensions(i).id();
    EXPECT_EQ(config.rtp.extensions[i].id, id);
    EXPECT_EQ(config.rtp.extensions[i].name, name);
  }
  // Check decoders.
  ASSERT_EQ(static_cast<int>(config.decoders.size()),
            receiver_config.decoders_size());
  for (int i = 0; i < receiver_config.decoders_size(); i++) {
    ASSERT_TRUE(receiver_config.decoders(i).has_name());
    ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
    const std::string& decoder_name = receiver_config.decoders(i).name();
    int decoder_type = receiver_config.decoders(i).payload_type();
    EXPECT_EQ(config.decoders[i].payload_name, decoder_name);
    EXPECT_EQ(config.decoders[i].payload_type, decoder_type);
  }
}

void VerifySendStreamConfig(const rtclog::Event& event,
                            const VideoSendStream::Config& config) {
  ASSERT_TRUE(IsValidBasicEvent(event));
  ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
  // Check SSRCs.
  ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()),
            sender_config.ssrcs_size());
  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
    EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
  }
  // Check header extensions.
  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
            sender_config.header_extensions_size());
  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
    ASSERT_TRUE(sender_config.header_extensions(i).has_name());
    ASSERT_TRUE(sender_config.header_extensions(i).has_id());
    const std::string& name = sender_config.header_extensions(i).name();
    int id = sender_config.header_extensions(i).id();
    EXPECT_EQ(config.rtp.extensions[i].id, id);
    EXPECT_EQ(config.rtp.extensions[i].name, name);
  }
  // Check RTX settings.
  ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
            sender_config.rtx_ssrcs_size());
  for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
    EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
  }
  if (sender_config.rtx_ssrcs_size() > 0) {
    ASSERT_TRUE(sender_config.has_rtx_payload_type());
    EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
  }
  // Check CNAME.
  ASSERT_TRUE(sender_config.has_c_name());
  EXPECT_EQ(config.rtp.c_name, sender_config.c_name());
  // Check encoder.
  ASSERT_TRUE(sender_config.has_encoder());
  ASSERT_TRUE(sender_config.encoder().has_name());
  ASSERT_TRUE(sender_config.encoder().has_payload_type());
  EXPECT_EQ(config.encoder_settings.payload_name,
            sender_config.encoder().name());
  EXPECT_EQ(config.encoder_settings.payload_type,
            sender_config.encoder().payload_type());
}

void VerifyRtpEvent(const rtclog::Event& event,
                    bool incoming,
                    MediaType media_type,
                    uint8_t* header,
                    size_t header_size,
                    size_t total_size) {
  ASSERT_TRUE(IsValidBasicEvent(event));
  ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
  ASSERT_TRUE(rtp_packet.has_incoming());
  EXPECT_EQ(incoming, rtp_packet.incoming());
  ASSERT_TRUE(rtp_packet.has_type());
  EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
  ASSERT_TRUE(rtp_packet.has_packet_length());
  EXPECT_EQ(total_size, rtp_packet.packet_length());
  ASSERT_TRUE(rtp_packet.has_header());
  ASSERT_EQ(header_size, rtp_packet.header().size());
  for (size_t i = 0; i < header_size; i++) {
    EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
  }
}

void VerifyRtcpEvent(const rtclog::Event& event,
                     bool incoming,
                     MediaType media_type,
                     uint8_t* packet,
                     size_t total_size) {
  ASSERT_TRUE(IsValidBasicEvent(event));
  ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
  ASSERT_TRUE(rtcp_packet.has_incoming());
  EXPECT_EQ(incoming, rtcp_packet.incoming());
  ASSERT_TRUE(rtcp_packet.has_type());
  EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
  ASSERT_TRUE(rtcp_packet.has_packet_data());
  ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
  for (size_t i = 0; i < total_size; i++) {
    EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
  }
}

void VerifyLogStartEvent(const rtclog::Event& event) {
  ASSERT_TRUE(IsValidBasicEvent(event));
  ASSERT_EQ(rtclog::Event::DEBUG_EVENT, event.type());
  const rtclog::DebugEvent& debug_event = event.debug_event();
  ASSERT_TRUE(debug_event.has_type());
  EXPECT_EQ(rtclog::DebugEvent::LOG_START, debug_event.type());
}

void GenerateVideoReceiveConfig(VideoReceiveStream::Config* config) {
  // Create a map from a payload type to an encoder name.
  VideoReceiveStream::Decoder decoder;
  decoder.payload_type = rand();
  decoder.payload_name = (rand() % 2 ? "VP8" : "H264");
  config->decoders.push_back(decoder);
  // Add SSRCs for the stream.
  config->rtp.remote_ssrc = rand();
  config->rtp.local_ssrc = rand();
  // Add extensions and settings for RTCP.
  config->rtp.rtcp_mode = rand() % 2 ? newapi::kRtcpCompound
                                     : newapi::kRtcpReducedSize;
  config->rtp.rtcp_xr.receiver_reference_time_report =
      static_cast<bool>(rand() % 2);
  config->rtp.remb = static_cast<bool>(rand() % 2);
  // 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 = rand();
  rtx_pair.payload_type = rand();
  config->rtp.rtx.insert(std::make_pair(rand(), rtx_pair));
  // Add two random header extensions.
  const char* extension_name = rand() % 2 ? RtpExtension::kTOffset
                                          : RtpExtension::kVideoRotation;
  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
  extension_name = rand() % 2 ? RtpExtension::kAudioLevel
                              : RtpExtension::kAbsSendTime;
  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
}

void GenerateVideoSendConfig(VideoSendStream::Config* config) {
  // Create a map from a payload type to an encoder name.
  config->encoder_settings.payload_type = rand();
  config->encoder_settings.payload_name = (rand() % 2 ? "VP8" : "H264");
  // Add SSRCs for the stream.
  config->rtp.ssrcs.push_back(rand());
  // Add a map from a payload type to new ssrcs and a new payload type for RTX.
  config->rtp.rtx.ssrcs.push_back(rand());
  config->rtp.rtx.payload_type = rand();
  // Add a CNAME.
  config->rtp.c_name = "some.user@some.host";
  // Add two random header extensions.
  const char* extension_name = rand() % 2 ? RtpExtension::kTOffset
                                          : RtpExtension::kVideoRotation;
  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
  extension_name = rand() % 2 ? RtpExtension::kAudioLevel
                              : RtpExtension::kAbsSendTime;
  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
}

// Test for the RtcEventLog class. Dumps some RTP packets to disk, then reads
// them back to see if they match.
void LogSessionAndReadBack(size_t rtp_count, unsigned random_seed) {
  std::vector<std::vector<uint8_t>> rtp_packets;
  std::vector<uint8_t> incoming_rtcp_packet;
  std::vector<uint8_t> outgoing_rtcp_packet;

  VideoReceiveStream::Config receiver_config;
  VideoSendStream::Config sender_config;

  srand(random_seed);

  // Create rtp_count RTP packets containing random data.
  const size_t rtp_header_size = 20;
  for (size_t i = 0; i < rtp_count; i++) {
    size_t packet_size = 1000 + rand() % 30;
    rtp_packets.push_back(std::vector<uint8_t>());
    rtp_packets[i].reserve(packet_size);
    for (size_t j = 0; j < packet_size; j++) {
      rtp_packets[i].push_back(rand());
    }
  }
  // Create two RTCP packets containing random data.
  size_t packet_size = 1000 + rand() % 30;
  outgoing_rtcp_packet.reserve(packet_size);
  for (size_t j = 0; j < packet_size; j++) {
    outgoing_rtcp_packet.push_back(rand());
  }
  packet_size = 1000 + rand() % 30;
  incoming_rtcp_packet.reserve(packet_size);
  for (size_t j = 0; j < packet_size; j++) {
    incoming_rtcp_packet.push_back(rand());
  }
  // Create configurations for the video streams.
  GenerateVideoReceiveConfig(&receiver_config);
  GenerateVideoSendConfig(&sender_config);

  // 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.
  {
    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
    log_dumper->LogVideoReceiveStreamConfig(receiver_config);
    log_dumper->LogVideoSendStreamConfig(sender_config);
    size_t i = 0;
    for (; i < rtp_count / 2; i++) {
      log_dumper->LogRtpHeader(
          (i % 2 == 0),  // Every second packet is incoming.
          (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
          rtp_packets[i].data(), rtp_header_size, rtp_packets[i].size());
    }
    log_dumper->LogRtcpPacket(false, MediaType::AUDIO,
                              outgoing_rtcp_packet.data(),
                              outgoing_rtcp_packet.size());
    log_dumper->StartLogging(temp_filename, 10000000);
    for (; i < rtp_count; i++) {
      log_dumper->LogRtpHeader(
          (i % 2 == 0),  // Every second packet is incoming,
          (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
          rtp_packets[i].data(), rtp_header_size, rtp_packets[i].size());
    }
    log_dumper->LogRtcpPacket(true, MediaType::VIDEO,
                              incoming_rtcp_packet.data(),
                              incoming_rtcp_packet.size());
  }

  const int config_count = 2;
  const int rtcp_count = 2;
  const int debug_count = 1;  // Only LogStart event,
  const int event_count = config_count + debug_count + rtcp_count + rtp_count;

  // Read the generated file from disk.
  rtclog::EventStream parsed_stream;

  ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));

  // Verify the result.
  EXPECT_EQ(event_count, parsed_stream.stream_size());
  VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
  VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
  size_t i = 0;
  for (; i < rtp_count / 2; i++) {
    VerifyRtpEvent(parsed_stream.stream(config_count + i),
                   (i % 2 == 0),  // Every second packet is incoming.
                   (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
                   rtp_packets[i].data(), rtp_header_size,
                   rtp_packets[i].size());
  }
  VerifyRtcpEvent(parsed_stream.stream(config_count + rtp_count / 2),
                  false,  // Outgoing RTCP packet.
                  MediaType::AUDIO, outgoing_rtcp_packet.data(),
                  outgoing_rtcp_packet.size());

  VerifyLogStartEvent(parsed_stream.stream(1 + config_count + rtp_count / 2));
  for (; i < rtp_count; i++) {
    VerifyRtpEvent(parsed_stream.stream(2 + config_count + i),
                   (i % 2 == 0),  // Every second packet is incoming.
                   (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
                   rtp_packets[i].data(), rtp_header_size,
                   rtp_packets[i].size());
  }
  VerifyRtcpEvent(parsed_stream.stream(2 + config_count + rtp_count),
                  true,  // Incoming RTCP packet.
                  MediaType::VIDEO, incoming_rtcp_packet.data(),
                  incoming_rtcp_packet.size());

  // Clean up temporary file - can be pretty slow.
  remove(temp_filename.c_str());
}

TEST(RtcEventLogTest, LogSessionAndReadBack) {
  LogSessionAndReadBack(5, 321);
  LogSessionAndReadBack(8, 3141592653u);
  LogSessionAndReadBack(9, 2718281828u);
}

}  // namespace webrtc

#endif  // ENABLE_RTC_EVENT_LOG