Update to the RtcEventLog protobuf to remove the DebugEvent message.

webrtc/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.
  */
 
@@ skipping 66 matching lines @@
     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())
+  if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
+      event.has_audio_playout_event())
     return ::testing::AssertionFailure()
            << "Event of type " << type << " has "
-           << (event.has_debug_event() ? "" : "no ") << "debug event";
+           << (event.has_audio_playout_event() ? "" : "no ")
+           << "audio_playout 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 "
@@ skipping 159 matching lines @@
   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 VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) {
   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::AUDIO_PLAYOUT, debug_event.type());
-  ASSERT_TRUE(debug_event.has_local_ssrc());
-  EXPECT_EQ(ssrc, debug_event.local_ssrc());
+  ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
+  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
+  ASSERT_TRUE(playout_event.has_local_ssrc());
+  EXPECT_EQ(ssrc, playout_event.local_ssrc());
 }
 
 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());
+  EXPECT_EQ(rtclog::Event::LOG_START, event.type());
 }
 
 /*
  * 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.
  */
 size_t GenerateRtpPacket(uint32_t extensions_bitvector,
                          uint32_t csrcs_count,
                          uint8_t* packet,
@@ skipping 99 matching lines @@
       config->rtp.extensions.push_back(
           RtpExtension(kExtensionNames[i], 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,
                            size_t rtcp_count,
-                           size_t debug_count,
+                           size_t playout_count,
                            uint32_t extensions_bitvector,
                            uint32_t csrcs_count,
                            unsigned random_seed) {
   ASSERT_LE(rtcp_count, rtp_count);
-  ASSERT_LE(debug_count, rtp_count);
+  ASSERT_LE(playout_count, rtp_count);
   std::vector<rtc::Buffer> rtp_packets;
   std::vector<rtc::Buffer> rtcp_packets;
   std::vector<size_t> rtp_header_sizes;
   std::vector<uint32_t> playout_ssrcs;
 
   VideoReceiveStream::Config receiver_config(nullptr);
   VideoSendStream::Config sender_config(nullptr);
 
   srand(random_seed);
 
   // Create rtp_count RTP packets containing random data.
   for (size_t i = 0; i < rtp_count; i++) {
     size_t packet_size = 1000 + rand() % 64;
     rtp_packets.push_back(rtc::Buffer(packet_size));
     size_t header_size = GenerateRtpPacket(extensions_bitvector, csrcs_count,
                                            rtp_packets[i].data(), packet_size);
     rtp_header_sizes.push_back(header_size);
   }
   // Create rtcp_count RTCP packets containing random data.
   for (size_t i = 0; i < rtcp_count; i++) {
     size_t packet_size = 1000 + rand() % 64;
     rtcp_packets.push_back(rtc::Buffer(packet_size));
     GenerateRtcpPacket(rtcp_packets[i].data(), packet_size);
   }
-  // Create debug_count random SSRCs to use when logging AudioPlayout events.
-  for (size_t i = 0; i < debug_count; i++) {
+  // Create playout_count random SSRCs to use when logging AudioPlayout events.
+  for (size_t i = 0; i < playout_count; i++) {
     playout_ssrcs.push_back(static_cast<uint32_t>(rand()));
   }
   // Create configurations for the video streams.
   GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config);
   GenerateVideoSendConfig(extensions_bitvector, &sender_config);
   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.
   {
     rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
     log_dumper->LogVideoReceiveStreamConfig(receiver_config);
     log_dumper->LogVideoSendStreamConfig(sender_config);
-    size_t rtcp_index = 1, debug_index = 1;
+    size_t rtcp_index = 1, playout_index = 1;
     for (size_t i = 1; 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 - 1].data(), rtp_packets[i - 1].size());
       if (i * rtcp_count >= rtcp_index * rtp_count) {
         log_dumper->LogRtcpPacket(
             rtcp_index % 2 == 0,  // Every second packet is incoming
             rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
             rtcp_packets[rtcp_index - 1].data(),
             rtcp_packets[rtcp_index - 1].size());
         rtcp_index++;
       }
-      if (i * debug_count >= debug_index * rtp_count) {
-        log_dumper->LogAudioPlayout(playout_ssrcs[debug_index - 1]);
-        debug_index++;
+      if (i * playout_count >= playout_index * rtp_count) {
+        log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
+        playout_index++;
       }
       if (i == rtp_count / 2) {
         log_dumper->StartLogging(temp_filename, 10000000);
       }
     }
   }
 
   // Read the generated file from disk.
   rtclog::EventStream parsed_stream;
 
   ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
 
   // Verify the result.
   const int event_count =
-      config_count + debug_count + rtcp_count + rtp_count + 1;
+      config_count + playout_count + rtcp_count + rtp_count + 1;
   EXPECT_EQ(event_count, parsed_stream.stream_size());
   VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
   VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
-  size_t event_index = config_count, rtcp_index = 1, debug_index = 1;
+  size_t event_index = config_count, rtcp_index = 1, playout_index = 1;
   for (size_t i = 1; i <= rtp_count; i++) {
     VerifyRtpEvent(parsed_stream.stream(event_index),
                    (i % 2 == 0),  // Every second packet is incoming.
                    (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
                    rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
                    rtp_packets[i - 1].size());
     event_index++;
     if (i * rtcp_count >= rtcp_index * rtp_count) {
       VerifyRtcpEvent(parsed_stream.stream(event_index),
                       rtcp_index % 2 == 0,  // Every second packet is incoming.
                       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 * debug_count >= debug_index * rtp_count) {
+    if (i * playout_count >= playout_index * rtp_count) {
       VerifyPlayoutEvent(parsed_stream.stream(event_index),
-                         playout_ssrcs[debug_index - 1]);
+                         playout_ssrcs[playout_index - 1]);
       event_index++;
-      debug_index++;
+      playout_index++;
     }
     if (i == rtp_count / 2) {
       VerifyLogStartEvent(parsed_stream.stream(event_index));
       event_index++;
     }
   }
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
@@ skipping 25 matching lines @@
                             extensions,       // Bit vector choosing extensions
                             csrcs_count,      // Number of contributing sources
                             rand());
     }
   }
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG