commit 55d932b33115ae6e3a542b822b33605a03edc324
author sakal <sakal@webrtc.org> Fri Sep 30 13:19:08 2016
committer Commit bot <commit-bot@chromium.org> Fri Sep 30 13:19:12 2016
tree 7f83bbb88f33a9342f85f8767fa5281db355f4d8
parent 115bd153c7f5a64eabbac222bad66e223abbfe2b

Add logging statements to places where the frame might be dropped in WebRTC pipeline.

BUG=b/31645554

Review-Url: https://codereview.webrtc.org/2361803003
Cr-Commit-Position: refs/heads/master@{#14457}

webrtc/test/BUILD.gn

diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 8de7ae1..6e494c76 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -288,6 +288,7 @@
     "fake_encoder.h",
     "fake_network_pipe.cc",
     "fake_network_pipe.h",
+    "fake_videorenderer.h",
     "frame_generator_capturer.cc",
     "frame_generator_capturer.h",
     "layer_filtering_transport.cc",

webrtc/test/call_test.cc

diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index d6cbaa6..a2133fc 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -210,6 +210,7 @@
     video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     for (const RtpExtension& extension : video_send_config_.rtp.extensions)
       video_config.rtp.extensions.push_back(extension);
+    video_config.renderer = &fake_renderer_;
     for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
       VideoReceiveStream::Decoder decoder =
           test::CreateMatchingDecoder(video_send_config_.encoder_settings);

webrtc/test/call_test.h

diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index d8019e5..4667e08 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -17,6 +17,7 @@
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
+#include "webrtc/test/fake_videorenderer.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 
@@ -103,6 +104,7 @@
   size_t num_video_streams_;
   size_t num_audio_streams_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  test::FakeVideoRenderer fake_renderer_;
 
  private:
   // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.

webrtc/test/fake_videorenderer.h

diff --git a/webrtc/test/fake_videorenderer.h b/webrtc/test/fake_videorenderer.h
new file mode 100644
index 0000000..ff43fc0
--- /dev/null
+++ b/webrtc/test/fake_videorenderer.h
@@ -0,0 +1,28 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_TEST_FAKE_VIDEORENDERER_H_
+#define WEBRTC_TEST_FAKE_VIDEORENDERER_H_
+
+#include "webrtc/media/base/videosinkinterface.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeVideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  void OnFrame(const webrtc::VideoFrame& frame) override {}
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_TEST_FAKE_VIDEORENDERER_H_

webrtc/test/test.gyp

diff --git a/webrtc/test/test.gyp b/webrtc/test/test.gyp
index c2223a8..c80714a 100644
--- a/webrtc/test/test.gyp
+++ b/webrtc/test/test.gyp
@@ -177,6 +177,7 @@
        'fake_encoder.h',
        'fake_network_pipe.cc',
        'fake_network_pipe.h',
+       'fake_videorenderer.h',
        'frame_generator_capturer.cc',
        'frame_generator_capturer.h',
        'layer_filtering_transport.cc',