Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )

Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.

Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}

TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695

Review URL: https://codereview.webrtc.org/1507903005

Cr-Commit-Position: refs/heads/master@{#10937}

.gitignore

@@ -162,5 +162,9 @@
 /webrtc/modules/audio_device/android/test/bin/
 /webrtc/modules/audio_device/android/test/gen/
 /webrtc/modules/audio_device/android/test/libs/
+/webrtc/video_engine/test/android/bin
+/webrtc/video_engine/test/android/gen
+/webrtc/video_engine/test/android/libs
+/webrtc/video_engine/test/android/obj
 /x86-generic_out/
 /xcodebuild

PRESUBMIT.py

@@ -19,6 +19,7 @@ CPPLINT_DIRS = [
   'webrtc/audio',
   'webrtc/call',
   'webrtc/modules/video_processing',
+  'webrtc/video_engine',
 ]
 
 # List of directories of "supported" native APIs. That means changes to headers

WATCHLISTS

@@ -32,8 +32,9 @@
     'java_files': {
       'filepath': '\.java$|\.xml$',
     },
-    'video': {
-      'filepath': 'webrtc/video/.*',
+    'video_engine': {
+      'filepath': 'webrtc/video_engine/.*|'\
+                  'webrtc/video/.*',
     },
     'voice_engine': {
       'filepath': 'webrtc/voice_engine/.*',
@@ -101,15 +102,14 @@
                      'bjornv@webrtc.org',
                      'minyue@webrtc.org',
                      'peah@webrtc.org'],
-    'video': ['andresp@webrtc.org',
-              'mflodman@webrtc.org',
-              'pbos@webrtc.org',
-              'perkj@webrtc.org',
-              'solenberg@webrtc.org',
-              'stefan@webrtc.org',
-              'video-team@agora.io',
-              'yujie.mao@webrtc.org',
-              'zhengzhonghou@agora.io'],
+    'video_engine': ['andresp@webrtc.org',
+                     'mflodman@webrtc.org',
+                     'perkj@webrtc.org',
+                     'solenberg@webrtc.org',
+                     'stefan@webrtc.org',
+                     'video-team@agora.io',
+                     'yujie.mao@webrtc.org',
+                     'zhengzhonghou@agora.io'],
     'voice_engine': ['andrew@webrtc.org',
                      'audio-team@agora.io',
                      'henrika@webrtc.org',

talk/media/base/videocommon.cc

@@ -107,6 +107,9 @@ void ComputeScaleMaxPixels(int frame_width, int frame_height, int max_pixels,
   ASSERT(scaled_width != NULL);
   ASSERT(scaled_height != NULL);
   ASSERT(max_pixels > 0);
+  // For VP8 the values for max width and height can be found here
+  // webrtc/src/video_engine/vie_defines.h (kViEMaxCodecWidth and
+  // kViEMaxCodecHeight)
   const int kMaxWidth = 4096;
   const int kMaxHeight = 3072;
   int new_frame_width = frame_width;

webrtc/audio/audio_send_stream_unittest.cc

@@ -21,7 +21,7 @@
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
-#include "webrtc/video/call_stats.h"
+#include "webrtc/video_engine/call_stats.h"
 
 namespace webrtc {
 namespace test {

webrtc/call/call.cc

@@ -39,9 +39,9 @@
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/video/call_stats.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
+#include "webrtc/video_engine/call_stats.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {

webrtc/call/congestion_controller.cc

@@ -25,10 +25,10 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/video/call_stats.h"
-#include "webrtc/video/payload_router.h"
-#include "webrtc/video/vie_encoder.h"
-#include "webrtc/video/vie_remb.h"
+#include "webrtc/video_engine/call_stats.h"
+#include "webrtc/video_engine/payload_router.h"
+#include "webrtc/video_engine/vie_encoder.h"
+#include "webrtc/video_engine/vie_remb.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {

webrtc/test/direct_transport.h

@@ -7,8 +7,8 @@
  *  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_DIRECT_TRANSPORT_H_
-#define WEBRTC_TEST_DIRECT_TRANSPORT_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_
 
 #include <assert.h>
 
@@ -63,4 +63,4 @@ class DirectTransport : public Transport {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_DIRECT_TRANSPORT_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_

webrtc/test/fake_decoder.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TEST_FAKE_DECODER_H_
-#define WEBRTC_TEST_FAKE_DECODER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_
 
 #include <vector>
 
@@ -71,4 +71,4 @@ class FakeNullDecoder : public FakeDecoder {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_FAKE_DECODER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_

webrtc/test/fake_encoder.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TEST_FAKE_ENCODER_H_
-#define WEBRTC_TEST_FAKE_ENCODER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_
 
 #include <vector>
 
@@ -82,4 +82,4 @@ class DelayedEncoder : public test::FakeEncoder {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_FAKE_ENCODER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_

webrtc/test/frame_generator_capturer.h

@@ -7,8 +7,8 @@
  *  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_FRAME_GENERATOR_CAPTURER_H_
-#define WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_
 
 #include <string>
 
@@ -74,4 +74,4 @@ class FrameGeneratorCapturer : public VideoCapturer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_

webrtc/test/gl/gl_renderer.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TEST_GL_GL_RENDERER_H_
-#define WEBRTC_TEST_GL_GL_RENDERER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_
 
 #ifdef WEBRTC_MAC
 #include <OpenGL/gl.h>
@@ -48,4 +48,4 @@ class GlRenderer : public VideoRenderer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_TEST_GL_GL_RENDERER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_

webrtc/test/linux/glx_renderer.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TEST_LINUX_GLX_RENDERER_H_
-#define WEBRTC_TEST_LINUX_GLX_RENDERER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_
 
 #include <GL/glx.h>
 #include <X11/Xlib.h>
@@ -45,4 +45,4 @@ class GlxRenderer : public GlRenderer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_TEST_LINUX_GLX_RENDERER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_

webrtc/test/mac/video_renderer_mac.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_
-#define WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/test/gl/gl_renderer.h"
@@ -38,4 +38,4 @@ class MacRenderer : public GlRenderer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_

webrtc/test/null_transport.h

@@ -7,8 +7,8 @@
  *  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_NULL_TRANSPORT_H_
-#define WEBRTC_TEST_NULL_TRANSPORT_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_
 
 #include "webrtc/transport.h"
 
@@ -27,4 +27,4 @@ class NullTransport : public Transport {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_NULL_TRANSPORT_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_

webrtc/test/rtp_rtcp_observer.h

@@ -7,8 +7,8 @@
  *  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_RTP_RTCP_OBSERVER_H_
-#define WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
 
 #include <map>
 #include <vector>
@@ -138,4 +138,4 @@ class PacketTransport : public test::DirectTransport {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_

webrtc/test/run_loop.h

@@ -7,8 +7,8 @@
  *  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_RUN_LOOP_H_
-#define WEBRTC_TEST_RUN_LOOP_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_
 
 namespace webrtc {
 namespace test {
@@ -19,4 +19,4 @@ void PressEnterToContinue();
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_RUN_LOOP_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_

webrtc/test/statistics.h

@@ -7,8 +7,8 @@
  *  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_STATISTICS_H_
-#define WEBRTC_TEST_STATISTICS_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_
 
 #include "webrtc/typedefs.h"
 
@@ -33,4 +33,4 @@ class Statistics {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_STATISTICS_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_

webrtc/test/vcm_capturer.h

@@ -7,8 +7,8 @@
  *  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_VCM_CAPTURER_H_
-#define WEBRTC_TEST_VCM_CAPTURER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/common_types.h"
@@ -47,4 +47,4 @@ class VcmCapturer : public VideoCapturer, public VideoCaptureDataCallback {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_TEST_VCM_CAPTURER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_

webrtc/test/video_capturer.h

@@ -7,8 +7,8 @@
  *  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_VIDEO_CAPTURER_H_
-#define WEBRTC_TEST_VIDEO_CAPTURER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_
 
 #include <stddef.h>
 
@@ -39,4 +39,4 @@ class VideoCapturer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_TEST_VIDEO_CAPTURER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_

webrtc/test/video_renderer.h

@@ -7,8 +7,8 @@
  *  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_VIDEO_RENDERER_H_
-#define WEBRTC_TEST_VIDEO_RENDERER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_
 
 #include <stddef.h>
 
@@ -36,4 +36,4 @@ class VideoRenderer : public webrtc::VideoRenderer {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_VIDEO_RENDERER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_

webrtc/test/win/d3d_renderer.h

@@ -7,8 +7,8 @@
  *  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_WIN_D3D_RENDERER_H_
-#define WEBRTC_TEST_WIN_D3D_RENDERER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_
 
 #include <Windows.h>
 #include <d3d9.h>
@@ -51,4 +51,4 @@ class D3dRenderer : public VideoRenderer {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_WIN_D3D_RENDERER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_

webrtc/transport.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TRANSPORT_H_
-#define WEBRTC_TRANSPORT_H_
+#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_TRANSPORT_H_
+#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_TRANSPORT_H_
 
 #include <stddef.h>
 
@@ -38,4 +38,4 @@ class Transport {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_TRANSPORT_H_
+#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_TRANSPORT_H_

webrtc/video/BUILD.gn

@@ -10,24 +10,34 @@ import("../build/webrtc.gni")
 
 source_set("video") {
   sources = [
-    "call_stats.cc",
-    "call_stats.h",
+    "../video_engine/call_stats.cc",
+    "../video_engine/call_stats.h",
+    "../video_engine/encoder_state_feedback.cc",
+    "../video_engine/encoder_state_feedback.h",
+    "../video_engine/overuse_frame_detector.cc",
+    "../video_engine/overuse_frame_detector.h",
+    "../video_engine/payload_router.cc",
+    "../video_engine/payload_router.h",
+    "../video_engine/report_block_stats.cc",
+    "../video_engine/report_block_stats.h",
+    "../video_engine/stream_synchronization.cc",
+    "../video_engine/stream_synchronization.h",
+    "../video_engine/vie_channel.cc",
+    "../video_engine/vie_channel.h",
+    "../video_engine/vie_encoder.cc",
+    "../video_engine/vie_encoder.h",
+    "../video_engine/vie_receiver.cc",
+    "../video_engine/vie_receiver.h",
+    "../video_engine/vie_remb.cc",
+    "../video_engine/vie_remb.h",
+    "../video_engine/vie_sync_module.cc",
+    "../video_engine/vie_sync_module.h",
     "encoded_frame_callback_adapter.cc",
     "encoded_frame_callback_adapter.h",
-    "encoder_state_feedback.cc",
-    "encoder_state_feedback.h",
-    "overuse_frame_detector.cc",
-    "overuse_frame_detector.h",
-    "payload_router.cc",
-    "payload_router.h",
     "receive_statistics_proxy.cc",
     "receive_statistics_proxy.h",
-    "report_block_stats.cc",
-    "report_block_stats.h",
     "send_statistics_proxy.cc",
     "send_statistics_proxy.h",
-    "stream_synchronization.cc",
-    "stream_synchronization.h",
     "video_capture_input.cc",
     "video_capture_input.h",
     "video_decoder.cc",
@@ -36,16 +46,6 @@ source_set("video") {
     "video_receive_stream.h",
     "video_send_stream.cc",
     "video_send_stream.h",
-    "vie_channel.cc",
-    "vie_channel.h",
-    "vie_encoder.cc",
-    "vie_encoder.h",
-    "vie_receiver.cc",
-    "vie_receiver.h",
-    "vie_remb.cc",
-    "vie_remb.h",
-    "vie_sync_module.cc",
-    "vie_sync_module.h",
   ]
 
   configs += [ "..:common_config" ]
@@ -69,7 +69,7 @@ source_set("video") {
     "../modules/video_coding",
     "../modules/video_processing",
     "../modules/video_render:video_render_module",
-    "../system_wrappers",
     "../voice_engine",
+    "../system_wrappers",
   ]
 }

webrtc/video/receive_statistics_proxy.h

@@ -20,8 +20,8 @@
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
-#include "webrtc/video/report_block_stats.h"
-#include "webrtc/video/vie_channel.h"
+#include "webrtc/video_engine/report_block_stats.h"
+#include "webrtc/video_engine/vie_channel.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_renderer.h"
 

webrtc/video/send_statistics_proxy.h

@@ -22,8 +22,8 @@
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/video/overuse_frame_detector.h"
-#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video_engine/overuse_frame_detector.h"
+#include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {

webrtc/video/video_capture_input.cc

@@ -22,9 +22,9 @@
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/video/overuse_frame_detector.h"
 #include "webrtc/video/send_statistics_proxy.h"
-#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video_engine/overuse_frame_detector.h"
+#include "webrtc/video_engine/vie_encoder.h"
 
 namespace webrtc {
 

webrtc/video/video_receive_stream.cc

@@ -19,8 +19,8 @@
 #include "webrtc/call/congestion_controller.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/video/call_stats.h"
 #include "webrtc/video/receive_statistics_proxy.h"
+#include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {

webrtc/video/video_receive_stream.h

@@ -22,9 +22,9 @@
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 #include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video/vie_channel.h"
-#include "webrtc/video/vie_encoder.h"
 #include "webrtc/video_encoder.h"
+#include "webrtc/video_engine/vie_channel.h"
+#include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {

webrtc/video/video_send_stream.cc

@@ -22,12 +22,12 @@
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
-#include "webrtc/video/call_stats.h"
-#include "webrtc/video/encoder_state_feedback.h"
-#include "webrtc/video/payload_router.h"
 #include "webrtc/video/video_capture_input.h"
-#include "webrtc/video/vie_channel.h"
-#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video_engine/call_stats.h"
+#include "webrtc/video_engine/encoder_state_feedback.h"
+#include "webrtc/video_engine/payload_router.h"
+#include "webrtc/video_engine/vie_channel.h"
+#include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {

webrtc/video/webrtc_video.gypi

@@ -24,24 +24,12 @@
       '<(webrtc_root)/webrtc.gyp:rtc_event_log',
     ],
     'webrtc_video_sources': [
-      'video/call_stats.cc',
-      'video/call_stats.h',
       'video/encoded_frame_callback_adapter.cc',
       'video/encoded_frame_callback_adapter.h',
-      'video/encoder_state_feedback.cc',
-      'video/encoder_state_feedback.h',
-      'video/overuse_frame_detector.cc',
-      'video/overuse_frame_detector.h',
-      'video/payload_router.cc',
-      'video/payload_router.h',
       'video/receive_statistics_proxy.cc',
       'video/receive_statistics_proxy.h',
-      'video/report_block_stats.cc',
-      'video/report_block_stats.h',
       'video/send_statistics_proxy.cc',
       'video/send_statistics_proxy.h',
-      'video/stream_synchronization.cc',
-      'video/stream_synchronization.h',
       'video/video_capture_input.cc',
       'video/video_capture_input.h',
       'video/video_decoder.cc',
@@ -50,16 +38,28 @@
       'video/video_receive_stream.h',
       'video/video_send_stream.cc',
       'video/video_send_stream.h',
-      'video/vie_channel.cc',
-      'video/vie_channel.h',
-      'video/vie_encoder.cc',
-      'video/vie_encoder.h',
-      'video/vie_receiver.cc',
-      'video/vie_receiver.h',
-      'video/vie_remb.cc',
-      'video/vie_remb.h',
-      'video/vie_sync_module.cc',
-      'video/vie_sync_module.h',
+      'video_engine/call_stats.cc',
+      'video_engine/call_stats.h',
+      'video_engine/encoder_state_feedback.cc',
+      'video_engine/encoder_state_feedback.h',
+      'video_engine/overuse_frame_detector.cc',
+      'video_engine/overuse_frame_detector.h',
+      'video_engine/payload_router.cc',
+      'video_engine/payload_router.h',
+      'video_engine/report_block_stats.cc',
+      'video_engine/report_block_stats.h',
+      'video_engine/stream_synchronization.cc',
+      'video_engine/stream_synchronization.h',
+      'video_engine/vie_channel.cc',
+      'video_engine/vie_channel.h',
+      'video_engine/vie_encoder.cc',
+      'video_engine/vie_encoder.h',
+      'video_engine/vie_receiver.cc',
+      'video_engine/vie_receiver.h',
+      'video_engine/vie_remb.cc',
+      'video_engine/vie_remb.h',
+      'video_engine/vie_sync_module.cc',
+      'video_engine/vie_sync_module.h',
     ],
   },
 }

webrtc/video_engine/OWNERS

@@ -0,0 +1,13 @@
+mflodman@webrtc.org
+pbos@webrtc.org
+stefan@webrtc.org
+
+per-file *.isolate=kjellander@webrtc.org
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
+per-file BUILD.gn=kjellander@webrtc.org
+

webrtc/video_engine/call_stats.cc

@@ -8,12 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/call_stats.h"
+#include "webrtc/video_engine/call_stats.h"
 
 #include <assert.h>
 
-#include <algorithm>
-
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"

webrtc/video_engine/call_stats.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_CALL_STATS_H_
-#define WEBRTC_VIDEO_CALL_STATS_H_
+#ifndef WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
+#define WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
 
 #include <list>
 
@@ -78,4 +78,4 @@ class CallStats : public Module {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_CALL_STATS_H_
+#endif  // WEBRTC_VIDEO_ENGINE_CALL_STATS_H_

webrtc/video_engine/call_stats_unittest.cc

@@ -14,7 +14,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/video/call_stats.h"
+#include "webrtc/video_engine/call_stats.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;

webrtc/video_engine/encoder_state_feedback.cc

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/encoder_state_feedback.h"
+#include "webrtc/video_engine/encoder_state_feedback.h"
 
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video_engine/vie_encoder.h"
 
 namespace webrtc {
 

webrtc/video_engine/encoder_state_feedback.h

@@ -11,8 +11,8 @@
 // TODO(mflodman) ViEEncoder has a time check to not send key frames too often,
 // move the logic to this class.
 
-#ifndef WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
-#define WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
+#ifndef WEBRTC_VIDEO_ENGINE_ENCODER_STATE_FEEDBACK_H_
+#define WEBRTC_VIDEO_ENGINE_ENCODER_STATE_FEEDBACK_H_
 
 #include <map>
 #include <vector>
@@ -68,4 +68,4 @@ class EncoderStateFeedback {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
+#endif  // WEBRTC_VIDEO_ENGINE_ENCODER_STATE_FEEDBACK_H_

webrtc/video_engine/encoder_state_feedback_unittest.cc

@@ -10,7 +10,7 @@
 
 
 // This file includes unit tests for EncoderStateFeedback.
-#include "webrtc/video/encoder_state_feedback.h"
+#include "webrtc/video_engine/encoder_state_feedback.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -22,8 +22,8 @@
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/utility/include/mock/mock_process_thread.h"
-#include "webrtc/video/payload_router.h"
-#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video_engine/payload_router.h"
+#include "webrtc/video_engine/vie_encoder.h"
 
 using ::testing::NiceMock;
 

webrtc/video_engine/overuse_frame_detector.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/overuse_frame_detector.h"
+#include "webrtc/video_engine/overuse_frame_detector.h"
 
 #include <assert.h>
 #include <math.h>

webrtc/video_engine/overuse_frame_detector.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_OVERUSE_FRAME_DETECTOR_H_
-#define WEBRTC_VIDEO_OVERUSE_FRAME_DETECTOR_H_
+#ifndef WEBRTC_VIDEO_ENGINE_OVERUSE_FRAME_DETECTOR_H_
+#define WEBRTC_VIDEO_ENGINE_OVERUSE_FRAME_DETECTOR_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
@@ -172,4 +172,4 @@ class OveruseFrameDetector : public Module {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_OVERUSE_FRAME_DETECTOR_H_
+#endif  // WEBRTC_VIDEO_ENGINE_OVERUSE_FRAME_DETECTOR_H_

webrtc/video_engine/overuse_frame_detector_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/overuse_frame_detector.h"
+#include "webrtc/video_engine/overuse_frame_detector.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"

webrtc/video_engine/payload_router.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/payload_router.h"
+#include "webrtc/video_engine/payload_router.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"

webrtc/video_engine/payload_router.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
-#define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
+#define WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
 
 #include <list>
 #include <vector>
@@ -82,4 +82,4 @@ class PayloadRouter {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_

webrtc/video_engine/payload_router_unittest.cc

@@ -16,7 +16,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "webrtc/video/payload_router.h"
+#include "webrtc/video_engine/payload_router.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;

webrtc/video_engine/report_block_stats.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/report_block_stats.h"
+#include "webrtc/video_engine/report_block_stats.h"
 
 namespace webrtc {
 

webrtc/video_engine/report_block_stats.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_REPORT_BLOCK_STATS_H_
-#define WEBRTC_VIDEO_REPORT_BLOCK_STATS_H_
+#ifndef WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
+#define WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
 
 #include <map>
 #include <vector>
@@ -58,5 +58,5 @@ class ReportBlockStats {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_REPORT_BLOCK_STATS_H_
+#endif  // WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
 

webrtc/video_engine/report_block_stats_unittest.cc

@@ -10,7 +10,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/video/report_block_stats.h"
+#include "webrtc/video_engine/report_block_stats.h"
 
 namespace webrtc {
 

webrtc/video_engine/stream_synchronization.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/stream_synchronization.h"
+#include "webrtc/video_engine/stream_synchronization.h"
 
 #include <assert.h>
 #include <math.h>

webrtc/video_engine/stream_synchronization.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
-#define WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
+#ifndef WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
+#define WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
 
 #include <list>
 
@@ -56,4 +56,4 @@ class StreamSynchronization {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
+#endif  // WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_

webrtc/video_engine/stream_synchronization_unittest.cc

@@ -13,7 +13,7 @@
 #include <algorithm>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/video/stream_synchronization.h"
+#include "webrtc/video_engine/stream_synchronization.h"
 
 namespace webrtc {
 

webrtc/video_engine/vie_channel.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/vie_channel.h"
+#include "webrtc/video_engine/vie_channel.h"
 
 #include <algorithm>
 #include <map>
@@ -31,10 +31,10 @@
 #include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/video/call_stats.h"
-#include "webrtc/video/payload_router.h"
 #include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video/report_block_stats.h"
+#include "webrtc/video_engine/call_stats.h"
+#include "webrtc/video_engine/payload_router.h"
+#include "webrtc/video_engine/report_block_stats.h"
 
 namespace webrtc {
 

webrtc/video_engine/vie_channel.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_VIE_CHANNEL_H_
-#define WEBRTC_VIDEO_VIE_CHANNEL_H_
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
 
 #include <list>
 #include <map>
@@ -25,8 +25,8 @@
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/video/vie_receiver.h"
-#include "webrtc/video/vie_sync_module.h"
+#include "webrtc/video_engine/vie_receiver.h"
+#include "webrtc/video_engine/vie_sync_module.h"
 
 namespace webrtc {
 
@@ -454,4 +454,4 @@ class ViEChannel : public VCMFrameTypeCallback,
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_VIE_CHANNEL_H_
+#endif  // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_

webrtc/video_engine/vie_encoder.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video_engine/vie_encoder.h"
 
 #include <assert.h>
 
@@ -32,8 +32,8 @@
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/video/payload_router.h"
 #include "webrtc/video/send_statistics_proxy.h"
+#include "webrtc/video_engine/payload_router.h"
 
 namespace webrtc {
 

webrtc/video_engine/vie_encoder.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_VIE_ENCODER_H_
-#define WEBRTC_VIDEO_VIE_ENCODER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
 
 #include <map>
 #include <vector>
@@ -192,4 +192,4 @@ class ViEEncoder : public RtcpIntraFrameObserver,
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_VIE_ENCODER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_

webrtc/video_engine/vie_receiver.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/vie_receiver.h"
+#include "webrtc/video_engine/vie_receiver.h"
 
 #include <vector>
 

webrtc/video_engine/vie_receiver.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_VIE_RECEIVER_H_
-#define WEBRTC_VIDEO_VIE_RECEIVER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
 
 #include <list>
 #include <vector>
@@ -129,4 +129,4 @@ class ViEReceiver : public RtpData {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_VIE_RECEIVER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_

webrtc/video_engine/vie_remb.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/vie_remb.h"
+#include "webrtc/video_engine/vie_remb.h"
 
 #include <assert.h>
 

webrtc/video_engine/vie_remb.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_VIE_REMB_H_
-#define WEBRTC_VIDEO_VIE_REMB_H_
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
 
 #include <list>
 #include <utility>
@@ -75,4 +75,4 @@ class VieRemb : public RemoteBitrateObserver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_VIE_REMB_H_
+#endif  // WEBRTC_VIDEO_ENGINE_VIE_REMB_H_

webrtc/video_engine/vie_remb_unittest.cc

@@ -20,7 +20,7 @@
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/video/vie_remb.h"
+#include "webrtc/video_engine/vie_remb.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;

webrtc/video_engine/vie_sync_module.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/vie_sync_module.h"
+#include "webrtc/video_engine/vie_sync_module.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
@@ -16,7 +16,7 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/video/stream_synchronization.h"
+#include "webrtc/video_engine/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {

webrtc/video_engine/vie_sync_module.h

@@ -11,13 +11,13 @@
 // ViESyncModule is responsible for synchronization audio and video for a given
 // VoE and ViE channel couple.
 
-#ifndef WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
-#define WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/video/stream_synchronization.h"
+#include "webrtc/video_engine/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
@@ -59,4 +59,4 @@ class ViESyncModule : public Module {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
+#endif  // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_

webrtc/video_renderer.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_RENDERER_H_
-#define WEBRTC_VIDEO_RENDERER_H_
+#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_RENDERER_H_
+#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_RENDERER_H_
 
 namespace webrtc {
 
@@ -35,4 +35,4 @@ class VideoRenderer {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_RENDERER_H_
+#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_RENDERER_H_

webrtc/webrtc_tests.gypi

@@ -160,20 +160,20 @@
         'call/packet_injection_tests.cc',
         'test/common_unittest.cc',
         'test/testsupport/metrics/video_metrics_unittest.cc',
-        'video/call_stats_unittest.cc',
-        'video/encoder_state_feedback_unittest.cc',
         'video/end_to_end_tests.cc',
-        'video/overuse_frame_detector_unittest.cc',
-        'video/payload_router_unittest.cc',
-        'video/report_block_stats_unittest.cc',
         'video/send_statistics_proxy_unittest.cc',
-        'video/stream_synchronization_unittest.cc',
         'video/video_capture_input_unittest.cc',
         'video/video_decoder_unittest.cc',
         'video/video_encoder_unittest.cc',
         'video/video_send_stream_tests.cc',
-        'video/vie_codec_unittest.cc',
-        'video/vie_remb_unittest.cc',
+        'video_engine/call_stats_unittest.cc',
+        'video_engine/encoder_state_feedback_unittest.cc',
+        'video_engine/overuse_frame_detector_unittest.cc',
+        'video_engine/payload_router_unittest.cc',
+        'video_engine/report_block_stats_unittest.cc',
+        'video_engine/stream_synchronization_unittest.cc',
+        'video_engine/vie_codec_unittest.cc',
+        'video_engine/vie_remb_unittest.cc',
       ],
       'dependencies': [
         '<(DEPTH)/testing/gmock.gyp:gmock',