Removed RtcEventLog deps to call:call_interfaces.

webrtc/logging/rtc_event_log/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 288 matching lines @@
     log_dumper->LogVideoReceiveStreamConfig(receiver_config);
     fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
     log_dumper->LogVideoSendStreamConfig(sender_config);
     fake_clock.AdvanceTimeMicros(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.AdvanceTimeMicros(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.AdvanceTimeMicros(prng.Rand(1, 1000));
       }
       if (i * playout_count >= playout_index * rtp_count) {
         log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
         playout_index++;
         fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
       }
@@ skipping 35 matching lines @@
   RtcEventLogTestHelper::VerifyVideoSendStreamConfig(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++;
@@ skipping 57 matching lines @@
   // 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
   rtc::ScopedFakeClock fake_clock;
   fake_clock.SetTimeMicros(prng.Rand<uint32_t>());
   std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
 
-  log_dumper->LogRtpHeader(kIncomingPacket, MediaType::VIDEO, rtp_packet.data(),
+  log_dumper->LogRtpHeader(kIncomingPacket, rtp_packet.data(),
                            rtp_packet.size());
   fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
 
   log_dumper->StartLogging(temp_filename, 10000000);
   fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
 
-  log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO,
-                            rtcp_packet.data(), rtcp_packet.size());
+  log_dumper->LogRtcpPacket(kOutgoingPacket, rtcp_packet.data(),
+                            rtcp_packet.size());
   fake_clock.AdvanceTimeMicros(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(),
+      parsed_log, 1, kIncomingPacket, 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::VerifyRtcpEvent(
+      parsed_log, 2, kOutgoingPacket, rtcp_packet.data(), rtcp_packet.size());
 
   RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3);
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
 
 TEST(RtcEventLogTest, LogLossBasedBweUpdateAndReadBack) {
   Random prng(1234);
 
@@ skipping 375 matching lines @@
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
   AudioNetworkAdaptationReadWriteTest test;
   test.DoTest();
 }
 
 }  // namespace webrtc