Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/call/rtc_event_log_proxy.cc

Index: webrtc/call/rtc_event_log_proxy.cc
diff --git a/webrtc/call/rtc_event_log_proxy.cc b/webrtc/call/rtc_event_log_proxy.cc
new file mode 100644
index 0000000000000000000000000000000000000000..b49847cfa83789638d36f7c535cc5b2536efd88d
--- /dev/null
+++ b/webrtc/call/rtc_event_log_proxy.cc
@@ -0,0 +1,47 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/call/rtc_event_log_proxy.h"
+
+namespace webrtc {
+
+void RtcEventLogProxy::SetEventLog(RtcEventLog* event_log) {
+  rtc::CritScope lock(&crit_);
+  event_log_ = event_log;
+}
+
+void RtcEventLogProxy::LogRtpHeader(PacketDirection direction,
+                                    MediaType media_type,
+                                    const uint8_t* header,
+                                    size_t packet_length) {
+  rtc::CritScope lock(&crit_);
+  if (event_log_) {

[stefan-webrtc, 2016/03/11 10:11:53]

Can we DCHECK on this instead so that we know that we won't miss some packets in
the beginning?

[ivoc, 2016/03/16 17:00:32]

Good idea.

+    event_log_->LogRtpHeader(direction, media_type, header, packet_length);
+  }
+}
+
+void RtcEventLogProxy::LogRtcpPacket(PacketDirection direction,
+                                     MediaType media_type,
+                                     const uint8_t* packet,
+                                     size_t length) {
+  rtc::CritScope lock(&crit_);
+  if (event_log_) {
+    event_log_->LogRtcpPacket(direction, media_type, packet, length);
+  }
+}
+
+void RtcEventLogProxy::LogAudioPlayout(uint32_t ssrc) {
+  rtc::CritScope lock(&crit_);
+  if (event_log_) {
+    event_log_->LogAudioPlayout(ssrc);
+  }
+}
+
+}  // namespace webrtc