Replace AudioSendStream::Config with rtclog::StreamConfig.

webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc

 /*
  *  Copyright (c) 2016 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 321 matching lines @@
   for (size_t i = 0; i < parsed_config.rtp_extensions.size(); i++) {
     EXPECT_EQ(config.rtp_extensions[i].uri,
               parsed_config.rtp_extensions[i].uri);
     EXPECT_EQ(config.rtp_extensions[i].id, parsed_config.rtp_extensions[i].id);
   }
 }
 
 void RtcEventLogTestHelper::VerifyAudioSendStreamConfig(
     const ParsedRtcEventLog& parsed_log,
     size_t index,
-    const AudioSendStream::Config& config) {
+    const rtclog::StreamConfig& config) {
   const rtclog::Event& event = parsed_log.events_[index];
   ASSERT_TRUE(IsValidBasicEvent(event));
   ASSERT_EQ(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT, event.type());
   const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
   // Check SSRCs.
-  EXPECT_EQ(config.rtp.ssrc, sender_config.ssrc());
+  EXPECT_EQ(config.local_ssrc, sender_config.ssrc());
   // Check header extensions.
-  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
+  ASSERT_EQ(static_cast<int>(config.rtp_extensions.size()),
             sender_config.header_extensions_size());
   for (int i = 0; i < sender_config.header_extensions_size(); i++) {
     ASSERT_TRUE(sender_config.header_extensions(i).has_name());
     ASSERT_TRUE(sender_config.header_extensions(i).has_id());
     const std::string& name = sender_config.header_extensions(i).name();
     int id = sender_config.header_extensions(i).id();
-    EXPECT_EQ(config.rtp.extensions[i].id, id);
-    EXPECT_EQ(config.rtp.extensions[i].uri, name);
+    EXPECT_EQ(config.rtp_extensions[i].id, id);
+    EXPECT_EQ(config.rtp_extensions[i].uri, name);
   }
 
   // Check consistency of the parser.
-  AudioSendStream::Config parsed_config(nullptr);
+  rtclog::StreamConfig parsed_config;
   parsed_log.GetAudioSendConfig(index, &parsed_config);
-  // Check SSRCs
-  EXPECT_EQ(config.rtp.ssrc, parsed_config.rtp.ssrc);
-  // Check header extensions.
-  EXPECT_EQ(config.rtp.extensions.size(), parsed_config.rtp.extensions.size());
-  for (size_t i = 0; i < parsed_config.rtp.extensions.size(); i++) {
-    EXPECT_EQ(config.rtp.extensions[i].uri,
-              parsed_config.rtp.extensions[i].uri);
-    EXPECT_EQ(config.rtp.extensions[i].id, parsed_config.rtp.extensions[i].id);
-  }
+  VerifyStreamConfigsAreEqual(config, parsed_config);
 }
 
 void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
                                            size_t index,
                                            PacketDirection direction,
                                            MediaType media_type,
                                            const uint8_t* header,
                                            size_t header_size,
                                            size_t total_size) {
   const rtclog::Event& event = parsed_log.events_[index];
@@ skipping 215 matching lines @@
   ASSERT_TRUE(bwe_event.has_id());
   EXPECT_EQ(id, bwe_event.id());
   ASSERT_TRUE(bwe_event.has_result());
   EXPECT_EQ(GetProbeResultType(failure_reason), bwe_event.result());
   ASSERT_FALSE(bwe_event.has_bitrate_bps());
 
   // TODO(philipel): Verify the parser when parsing has been implemented.
 }
 
 }  // namespace webrtc