Replace AudioSendStream::Config with rtclog::StreamConfig.

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 188 matching lines @@
   // 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 GenerateAudioSendConfig(uint32_t extensions_bitvector,
-                             AudioSendStream::Config* config,
+                             rtclog::StreamConfig* config,
                              Random* prng) {
   // Add SSRC to the stream.
-  config->rtp.ssrc = prng->Rand<uint32_t>();
+  config->local_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(
+      config->rtp_extensions.push_back(
           RtpExtension(kExtensionNames[i], prng->Rand<int>()));
     }
   }
 }
 
 void GenerateAudioNetworkAdaptation(uint32_t extensions_bitvector,
                                     AudioEncoderRuntimeConfig* config,
                                     Random* prng) {
   config->bitrate_bps = rtc::Optional<int>(prng->Rand(0, 3000000));
   config->enable_fec = rtc::Optional<bool>(prng->Rand<bool>());
@@ skipping 554 matching lines @@
   void VerifyConfig(const ParsedRtcEventLog& parsed_log,
                     size_t index) override {
     RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig(parsed_log, index,
                                                           config);
   }
   rtclog::StreamConfig config;
 };
 
 class AudioSendConfigReadWriteTest : public ConfigReadWriteTest {
  public:
-  AudioSendConfigReadWriteTest() : config(nullptr) {}
+  AudioSendConfigReadWriteTest() {}
   void GenerateConfig(uint32_t extensions_bitvector) override {
     GenerateAudioSendConfig(extensions_bitvector, &config, &prng);
   }
   void LogConfig(RtcEventLog* event_log) override {
     event_log->LogAudioSendStreamConfig(config);
   }
   void VerifyConfig(const ParsedRtcEventLog& parsed_log,
                     size_t index) override {
     RtcEventLogTestHelper::VerifyAudioSendStreamConfig(parsed_log, index,
                                                        config);
   }
-  AudioSendStream::Config config;
+  rtclog::StreamConfig config;
 };
 
 class VideoReceiveConfigReadWriteTest : public ConfigReadWriteTest {
  public:
   VideoReceiveConfigReadWriteTest() {}
   void GenerateConfig(uint32_t extensions_bitvector) override {
     GenerateVideoReceiveConfig(extensions_bitvector, &config, &prng);
   }
   void LogConfig(RtcEventLog* event_log) override {
     event_log->LogVideoReceiveStreamConfig(config);
@@ skipping 58 matching lines @@
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
   AudioNetworkAdaptationReadWriteTest test;
   test.DoTest();
 }
 
 }  // namespace webrtc