Expose Ivf logging through the native API

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  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.
  */
 
 #include "webrtc/media/engine/fakewebrtccall.h"
 
 #include <algorithm>
 #include <utility>
 
 #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"
 
 namespace cricket {
 FakeAudioSendStream::FakeAudioSendStream(
     const webrtc::AudioSendStream::Config& config) : config_(config) {
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
 const webrtc::AudioSendStream::Config&
@@ skipping 142 matching lines @@
 
 void FakeVideoSendStream::SetStats(
     const webrtc::VideoSendStream::Stats& stats) {
   stats_ = stats;
 }
 
 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) {
     if (config_.encoder_settings.payload_name == "VP8") {
       vpx_settings_.vp8 = *reinterpret_cast<const webrtc::VideoCodecVP8*>(
                               config.encoder_specific_settings);
       if (!config.streams.empty()) {
         vpx_settings_.vp8.numberOfTemporalLayers = static_cast<unsigned char>(
             config.streams.back().temporal_layer_thresholds_bps.size() + 1);
       }
@@ skipping 52 matching lines @@
 
 void FakeVideoReceiveStream::Stop() {
   receiving_ = false;
 }
 
 void FakeVideoReceiveStream::SetStats(
     const webrtc::VideoReceiveStream::Stats& stats) {
   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),
       video_network_state_(webrtc::kNetworkUp),
       num_created_send_streams_(0),
       num_created_receive_streams_(0) {}
 
 FakeCall::~FakeCall() {
   EXPECT_EQ(0u, video_send_streams_.size());
   EXPECT_EQ(0u, audio_send_streams_.size());
@@ skipping 218 matching lines @@
 }
 
 bool FakeCall::StartEventLog(rtc::PlatformFile log_file,
                              int64_t max_size_bytes) {
   return false;
 }
 
 void FakeCall::StopEventLog() {}
 
 }  // namespace cricket