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 cb904df66fdcf1275bf722ddb6220968bd0bd8ab..47185e490aac5be5bdf6f110e8fffea6c8006d06 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"
 
@@ -177,6 +178,14 @@ webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {
   return stats_;
 }
 
+void FakeVideoSendStream::EnableEncodedFrameRecording(
+    const std::vector<rtc::PlatformFile>& files,
+    size_t byte_limit) {

[stefan-webrtc, 2016/09/13 10:53:17]

Why does this take a byte_limit?

[palmkvist, 2016/09/13 11:59:37]

It's not used in the fake streams, but in the regular streams it's there to make
sure we don't create massive logs in case logging is left on.

+  for (auto it = files.begin(); it < files.end(); ++it) {

[stefan-webrtc, 2016/09/13 10:53:17]

Something like this instead:
for (rtc::PlatformFile* file : files) 
  rtc::ClosePlatformFile(*file);

[palmkvist, 2016/09/13 11:59:37]

Done.

+    rtc::ClosePlatformFile(*it);
+  }
+}
+
 void FakeVideoSendStream::ReconfigureVideoEncoder(
     webrtc::VideoEncoderConfig config) {
   if (config.encoder_specific_settings != NULL) {
@@ -249,6 +258,11 @@ 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),