Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc

Index: webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
index 15404667856ed9bace4c3eb173061e2fc0921635..f0ee973a136fad18308332f033ab6ff02b1d6dd8 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -228,6 +228,19 @@ void GenerateAudioSendConfig(uint32_t extensions_bitvector,
   }
 }
 
+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,
@@ -604,6 +617,22 @@ class VideoSendConfigReadWriteTest : public ConfigReadWriteTest {
   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();
@@ -623,4 +652,10 @@ TEST(RtcEventLogTest, LogVideoSendConfig) {
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
+
+TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
+  AudioNetworkAdaptationReadWriteTest test;
+  test.DoTest();
+}
+
 }  // namespace webrtc