diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 6d84c504a3..1517d85c6b 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -167,10 +167,12 @@ config("common_config") {
 
 source_set("webrtc") {
   sources = [
+    "audio_send_stream.h",
+    "audio_state.h",
     "call.h",
-    "config.h",
-    "frame_callback.h",
-    "transport.h",
+    "video_decoder.h",
+    "video_encoder.h",
+    "video_frame.h",
   ]
 
   defines = []
@@ -228,12 +230,20 @@ if (!build_with_chromium) {
 
 source_set("webrtc_common") {
   sources = [
+    "audio_receive_stream.h",
+    "audio_sink.h",
     "common_types.cc",
     "common_types.h",
     "config.cc",
     "config.h",
     "engine_configurations.h",
+    "frame_callback.h",
+    "stream.h",
+    "transport.h",
     "typedefs.h",
+    "video_receive_stream.h",
+    "video_renderer.h",
+    "video_send_stream.h",
   ]
 
   configs += [ ":common_config" ]
diff --git a/webrtc/api/remoteaudiosource.h b/webrtc/api/remoteaudiosource.h
index 20e5d90cdd..72ed17c58f 100644
--- a/webrtc/api/remoteaudiosource.h
+++ b/webrtc/api/remoteaudiosource.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/notifier.h"
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/media/base/audiorenderer.h"
 
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
index b2494140b4..e5cea14439 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -23,7 +23,7 @@
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/sctputils.h"
 #include "webrtc/api/webrtcsessiondescriptionfactory.h"
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/helpers.h"
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 2c58def560..9c25389471 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -13,7 +13,7 @@
 #include <string>
 #include <utility>
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi
index 53b7d16b1a..9b4879a70b 100644
--- a/webrtc/audio/webrtc_audio.gypi
+++ b/webrtc/audio/webrtc_audio.gypi
@@ -18,7 +18,6 @@
       'audio/audio_receive_stream.h',
       'audio/audio_send_stream.cc',
       'audio/audio_send_stream.h',
-      'audio/audio_sink.h',
       'audio/audio_state.cc',
       'audio/audio_state.h',
       'audio/conversion.h',
diff --git a/webrtc/audio/audio_sink.h b/webrtc/audio_sink.h
similarity index 93%
rename from webrtc/audio/audio_sink.h
rename to webrtc/audio_sink.h
index 999644f4ce..2c932c5ab8 100644
--- a/webrtc/audio/audio_sink.h
+++ b/webrtc/audio_sink.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_AUDIO_AUDIO_SINK_H_
-#define WEBRTC_AUDIO_AUDIO_SINK_H_
+#ifndef WEBRTC_AUDIO_SINK_H_
+#define WEBRTC_AUDIO_SINK_H_
 
 #if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
 // Avoid conflict with format_macros.h.
@@ -50,4 +50,4 @@ class AudioSinkInterface {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_AUDIO_SINK_H_
+#endif  // WEBRTC_AUDIO_SINK_H_
diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn
index 119001da89..7f951e8d0f 100644
--- a/webrtc/base/BUILD.gn
+++ b/webrtc/base/BUILD.gn
@@ -126,6 +126,7 @@ static_library("rtc_base_approved") {
     "event_tracer.h",
     "exp_filter.cc",
     "exp_filter.h",
+    "format_macros.h",
     "md5.cc",
     "md5.h",
     "md5digest.cc",
diff --git a/webrtc/base/base.gyp b/webrtc/base/base.gyp
index 18e4adfeaf..27527b58c3 100644
--- a/webrtc/base/base.gyp
+++ b/webrtc/base/base.gyp
@@ -94,6 +94,7 @@
         'event_tracer.h',
         'exp_filter.cc',
         'exp_filter.h',
+        'format_macros.h',
         'logging.cc',
         'logging.h',
         'md5.cc',
diff --git a/webrtc/common.gyp b/webrtc/common.gyp
index 3b5fe902dd..9ca4bf4e22 100644
--- a/webrtc/common.gyp
+++ b/webrtc/common.gyp
@@ -12,12 +12,20 @@
       'target_name': 'webrtc_common',
       'type': 'static_library',
       'sources': [
+        'audio_receive_stream.h',
+        'audio_sink.h',
         'common_types.cc',
         'common_types.h',
-        'config.h',
         'config.cc',
+        'config.h',
         'engine_configurations.h',
+        'frame_callback.h',
+        'stream.h',
+        'transport.h',
         'typedefs.h',
+        'video_receive_stream.h',
+        'video_renderer.h',
+        'video_send_stream.h',
       ],
     },
   ],
diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
index f2de5ac653..d68db70830 100644
--- a/webrtc/media/base/fakemediaengine.h
+++ b/webrtc/media/base/fakemediaengine.h
@@ -17,7 +17,7 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/media/base/audiorenderer.h"
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index af098af822..93b565dbf3 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -13,7 +13,7 @@
 #include <algorithm>
 #include <utility>
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/media/base/rtputils.h"
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 3709e807d3..1d235934da 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -21,7 +21,7 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc
index 6a696ea4b4..447687a425 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -12,7 +12,7 @@
 
 #include "webrtc/pc/channel.h"
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/byteorder.h"
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index abecd669e5..f72818924d 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -17,7 +17,7 @@
 #include <utility>
 #include <vector>
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/network.h"
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index e35772e22c..4f1b7ae197 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -60,9 +60,12 @@ source_set("video") {
   deps = [
     "..:rtc_event_log",
     "..:webrtc_common",
+    "../base:rtc_base_approved",
     "../common_video",
     "../modules/bitrate_controller",
+    "../modules/congestion_controller",
     "../modules/pacing",
+    "../modules/remote_bitrate_estimator",
     "../modules/rtp_rtcp",
     "../modules/utility",
     "../modules/video_capture:video_capture_module",
diff --git a/webrtc/video/webrtc_video.gypi b/webrtc/video/webrtc_video.gypi
index db8d5c7e89..f11ce95727 100644
--- a/webrtc/video/webrtc_video.gypi
+++ b/webrtc/video/webrtc_video.gypi
@@ -12,6 +12,7 @@
       '<(webrtc_root)/common.gyp:webrtc_common',
       '<(webrtc_root)/common_video/common_video.gyp:common_video',
       '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
+      '<(webrtc_root)/modules/modules.gyp:congestion_controller',
       '<(webrtc_root)/modules/modules.gyp:paced_sender',
       '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       '<(webrtc_root)/modules/modules.gyp:video_capture_module',
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 82cd92355c..13104c6c86 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -99,6 +99,7 @@ source_set("voice_engine") {
   deps = [
     "..:rtc_event_log",
     "..:webrtc_common",
+    "../base:rtc_base_approved",
     "../common_audio",
     "../modules/audio_coding",
     "../modules/audio_conference_mixer",
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 0e87252877..a3cd5d6535 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -13,7 +13,7 @@
 
 #include <memory>
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 3beaf9b294..da7864f15f 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -12,7 +12,7 @@
 
 #include <utility>
 
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/voice_engine/channel.h"
 
diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp
index ff588d8ead..cff2d8f2d9 100644
--- a/webrtc/voice_engine/voice_engine.gyp
+++ b/webrtc/voice_engine/voice_engine.gyp
@@ -15,6 +15,7 @@
       'target_name': 'voice_engine',
       'type': 'static_library',
       'dependencies': [
+        '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp
index 1adae73f9e..0c87e42031 100644
--- a/webrtc/webrtc.gyp
+++ b/webrtc/webrtc.gyp
@@ -110,18 +110,12 @@
       'target_name': 'webrtc',
       'type': 'static_library',
       'sources': [
-        'audio_receive_stream.h',
         'audio_send_stream.h',
         'audio_state.h',
         'call.h',
-        'config.h',
-        'frame_callback.h',
-        'stream.h',
-        'transport.h',
-        'video_receive_stream.h',
-        'video_renderer.h',
-        'video_send_stream.h',
-
+        'video_frame.h',
+        'video_decoder.h',
+        'video_encoder.h',
         '<@(webrtc_audio_sources)',
         '<@(webrtc_call_sources)',
         '<@(webrtc_video_sources)',