Replace AudioReceiveStream::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 173 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 GenerateAudioReceiveConfig(uint32_t extensions_bitvector,
-                                AudioReceiveStream::Config* config,
+                                rtclog::StreamConfig* config,
                                 Random* prng) {
   // Add SSRCs for the stream.
-  config->rtp.remote_ssrc = prng->Rand<uint32_t>();
-  config->rtp.local_ssrc = prng->Rand<uint32_t>();
+  config->remote_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 GenerateAudioSendConfig(uint32_t extensions_bitvector,
                              AudioSendStream::Config* config,
                              Random* prng) {
   // Add SSRC to the stream.
   config->rtp.ssrc = prng->Rand<uint32_t>();
@@ skipping 563 matching lines @@
     GenerateAudioReceiveConfig(extensions_bitvector, &config, &prng);
   }
   void LogConfig(RtcEventLog* event_log) override {
     event_log->LogAudioReceiveStreamConfig(config);
   }
   void VerifyConfig(const ParsedRtcEventLog& parsed_log,
                     size_t index) override {
     RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig(parsed_log, index,
                                                           config);
   }
-  AudioReceiveStream::Config config;
+  rtclog::StreamConfig config;
 };
 
 class AudioSendConfigReadWriteTest : public ConfigReadWriteTest {
  public:
   AudioSendConfigReadWriteTest() : config(nullptr) {}
   void GenerateConfig(uint32_t extensions_bitvector) override {
     GenerateAudioSendConfig(extensions_bitvector, &config, &prng);
   }
   void LogConfig(RtcEventLog* event_log) override {
     event_log->LogAudioSendStreamConfig(config);
@@ skipping 75 matching lines @@
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
   AudioNetworkAdaptationReadWriteTest test;
   test.DoTest();
 }
 
 }  // namespace webrtc