Reland of "Log audio network adapter decisions in event log."

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 210 matching lines @@
   config->rtp.ssrc = prng->Rand<uint32_t>();
   // 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 GenerateAudioNetworkAdaptation(
+    uint32_t extensions_bitvector,
+    AudioNetworkAdaptor::EncoderRuntimeConfig* config,
+    Random* prng) {
+  config->bitrate_bps = rtc::Optional<int>(prng->Rand(0, 3000000));
+  config->enable_fec = rtc::Optional<bool>(prng->Rand<bool>());
+  config->enable_dtx = rtc::Optional<bool>(prng->Rand<bool>());
+  config->frame_length_ms = rtc::Optional<int>(prng->Rand(10, 120));
+  config->num_channels = rtc::Optional<size_t>(prng->Rand(1, 2));
+  config->uplink_packet_loss_fraction =
+      rtc::Optional<float>(prng->Rand<float>());
+}
+
 // 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);
@@ skipping 356 matching lines @@
     event_log->LogVideoSendStreamConfig(config);
   }
   void VerifyConfig(const ParsedRtcEventLog& parsed_log,
                     size_t index) override {
     RtcEventLogTestHelper::VerifyVideoSendStreamConfig(parsed_log, index,
                                                        config);
   }
   VideoSendStream::Config config;
 };
 
+class AudioNetworkAdaptationReadWriteTest : public ConfigReadWriteTest {
+ public:
+  void GenerateConfig(uint32_t extensions_bitvector) override {
+    GenerateAudioNetworkAdaptation(extensions_bitvector, &config, &prng);
+  }
+  void LogConfig(RtcEventLog* event_log) override {
+    event_log->LogAudioNetworkAdaptation(config);
+  }
+  void VerifyConfig(const ParsedRtcEventLog& parsed_log,
+                    size_t index) override {
+    RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(parsed_log, index,
+                                                        config);
+  }
+  AudioNetworkAdaptor::EncoderRuntimeConfig config;
+};
+
 TEST(RtcEventLogTest, LogAudioReceiveConfig) {
   AudioReceiveConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioSendConfig) {
   AudioSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogVideoReceiveConfig) {
   VideoReceiveConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogVideoSendConfig) {
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
+
+TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
+  AudioNetworkAdaptationReadWriteTest test;
+  test.DoTest();
+}
+
 }  // namespace webrtc