Convenience functions to convert RtcEvents to webrtc types.

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.
  */
 
 #ifdef ENABLE_RTC_EVENT_LOG
 
 #include <stdio.h>
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.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"
+#include "webrtc/video/rtc_event_log_parser.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 {
 
 namespace {
 
 const RTPExtensionType kExtensionTypes[] = {
     RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
     RTPExtensionType::kRtpExtensionAudioLevel,
     RTPExtensionType::kRtpExtensionAbsoluteSendTime,
     RTPExtensionType::kRtpExtensionVideoRotation,
     RTPExtensionType::kRtpExtensionTransportSequenceNumber};
 const char* kExtensionNames[] = {RtpExtension::kTOffset,
                                  RtpExtension::kAudioLevel,
                                  RtpExtension::kAbsSendTime,
                                  RtpExtension::kVideoRotation,
                                  RtpExtension::kTransportSequenceNumber};
 const size_t kNumExtensions = 5;
 
-}  // namepsace
-
-// 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;
-}
+}  // namespace
 
 // 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())
@@ skipping 42 matching lines @@
   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());
+  EXPECT_EQ(config.rtp.rtcp_mode,
+            RtcEventLogParser::GetRuntimeRtcpMode(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());
@@ skipping 81 matching lines @@
                     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()));
+  EXPECT_EQ(media_type,
+            RtcEventLogParser::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()));
+  EXPECT_EQ(media_type,
+            RtcEventLogParser::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());
@@ skipping 199 matching lines @@
       }
       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));
+  ASSERT_TRUE(
+      RtcEventLogParser::ParseRtcEventLog(temp_filename, &parsed_stream));
 
   // Verify the result.
   const int event_count =
       config_count + debug_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;
   for (size_t i = 1; i <= rtp_count; i++) {
     VerifyRtpEvent(parsed_stream.stream(event_index),
@@ skipping 54 matching lines @@
                             extensions,       // Bit vector choosing extensions
                             csrcs_count,      // Number of contributing sources
                             rand());
     }
   }
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG