Expose Ivf logging through the native API

webrtc/media/engine/fakewebrtccall.cc

Index: webrtc/media/engine/fakewebrtccall.cc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 9670f1371066318e68ea413de7c0b7d9b05387d6..9515505af2c2458f04e25b0ef4aae5f0abe74b5d 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -15,6 +15,7 @@
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/platform_file.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/media/base/rtputils.h"
 
@@ -182,6 +183,13 @@ webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {
   return stats_;
 }
 
+void FakeVideoSendStream::EnableEncodedFrameRecording(
+    const std::vector<rtc::PlatformFile>& files,
+    size_t byte_limit) {
+  for (rtc::PlatformFile file : files)
+    rtc::ClosePlatformFile(file);
+}
+
 void FakeVideoSendStream::ReconfigureVideoEncoder(
     webrtc::VideoEncoderConfig config) {
   if (config.encoder_specific_settings != NULL) {
@@ -258,6 +266,11 @@ void FakeVideoReceiveStream::SetStats(
   stats_ = stats;
 }
 
+void FakeVideoReceiveStream::EnableEncodedFrameRecording(rtc::PlatformFile file,
+                                                         size_t byte_limit) {
+  rtc::ClosePlatformFile(file);
+}
+
 FakeCall::FakeCall(const webrtc::Call::Config& config)
     : config_(config),
       audio_network_state_(webrtc::kNetworkUp),