Build WebRTC with data channel only.

webrtc/logging/rtc_event_log/rtc_event_log.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/config.h"
 
 namespace webrtc {
 
 // Forward declaration of storage class that is automatically generated from
 // the protobuf file.
 namespace rtclog {
 class EventStream;
 
@@ skipping 154 matching lines @@
 };
 
 // No-op implementation is used if flag is not set, or in tests.
 class RtcEventLogNullImpl : public RtcEventLog {
  public:
   bool StartLogging(const std::string& file_name,
                     int64_t max_size_bytes) override {
     return false;
   }
   bool StartLogging(rtc::PlatformFile platform_file,
-                    int64_t max_size_bytes) override;
+                    int64_t max_size_bytes) override {
+    // The platform_file is open and needs to be closed.
+    if (!rtc::ClosePlatformFile(platform_file)) {
+      LOG(LS_ERROR) << "Can't close file.";

[Zhi Huang, 2017/06/12 05:59:28]

It would cause some macro definition conflict when building with Chromium. I
guess that's why the implementation of this method is in the .cc file. Any
suggestions about how to solve it? Can we just remove the log?

+    }
+    return false;
+  }
   void StopLogging() override {}
   void LogVideoReceiveStreamConfig(
       const rtclog::StreamConfig& config) override {}
   void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {}
   void LogAudioReceiveStreamConfig(
       const rtclog::StreamConfig& config) override {}
   void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {}
   void LogRtpHeader(PacketDirection direction,
                     const uint8_t* header,
                     size_t packet_length) override {}
@@ skipping 17 matching lines @@
                               int min_probes,
                               int min_bytes) override{};
   void LogProbeResultSuccess(int id, int bitrate_bps) override{};
   void LogProbeResultFailure(int id,
                              ProbeFailureReason failure_reason) override{};
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_