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), |