diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index 8ed07b14a5..75f1a31776 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -12,6 +12,7 @@ import("../build/webrtc.gni")
 group("media") {
   public_deps = [
     ":rtc_media",
+    ":rtc_media_base",
   ]
 }
 
@@ -41,7 +42,7 @@ if (is_linux && rtc_use_gtk) {
   }
 }
 
-rtc_static_library("rtc_media") {
+rtc_static_library("rtc_media_base") {
   defines = []
   libs = []
   deps = []
@@ -81,6 +82,60 @@ rtc_static_library("rtc_media") {
     "base/videoframe.h",
     "base/videosourcebase.cc",
     "base/videosourcebase.h",
+  ]
+
+  configs += [ ":rtc_media_warnings_config" ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+
+  include_dirs = []
+  if (rtc_build_libyuv) {
+    deps += [ "$rtc_libyuv_dir" ]
+    public_deps = [
+      "$rtc_libyuv_dir",
+    ]
+  } else {
+    # Need to add a directory normally exported by libyuv.
+    include_dirs += [ "$rtc_libyuv_dir/include" ]
+  }
+
+  deps += [
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../p2p",
+  ]
+}
+
+rtc_static_library("rtc_media") {
+  defines = []
+  libs = []
+  deps = []
+  sources = [
+    # TODO(magjed): Remove base header files once Chromium is updated.
+    "base/adaptedvideotracksource.h",
+    "base/audiosource.h",
+    "base/codec.h",
+    "base/cryptoparams.h",
+    "base/device.h",
+    "base/hybriddataengine.h",
+    "base/mediachannel.h",
+    "base/mediaconstants.h",
+    "base/mediaengine.h",
+    "base/rtpdataengine.h",
+    "base/rtpdump.h",
+    "base/rtputils.h",
+    "base/streamparams.h",
+    "base/turnutils.h",
+    "base/videoadapter.h",
+    "base/videobroadcaster.h",
+    "base/videocapturer.h",
+    "base/videocapturerfactory.h",
+    "base/videocommon.h",
+    "base/videoframe.h",
+    "base/videosourcebase.h",
     "engine/internalencoderfactory.cc",
     "engine/internalencoderfactory.h",
     "engine/nullwebrtcvideoengine.h",
@@ -168,15 +223,14 @@ rtc_static_library("rtc_media") {
     public_configs += [ ":gtk-lib" ]
   }
   deps += [
+    ":rtc_media_base",
     "..:webrtc_common",
     "../api:call_api",
     "../base:rtc_base_approved",
     "../call",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/video_coding",
-    "../p2p",
     "../system_wrappers",
-    "../video",
     "../voice_engine",
   ]
 }
diff --git a/webrtc/media/base/codec.cc b/webrtc/media/base/codec.cc
index 93cd4f3daf..835cf77b3e 100644
--- a/webrtc/media/base/codec.cc
+++ b/webrtc/media/base/codec.cc
@@ -13,7 +13,7 @@
 #include <algorithm>
 #include <sstream>
 
-#include "webrtc/base/common.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
@@ -54,7 +54,7 @@ void FeedbackParams::Add(const FeedbackParam& param) {
     return;
   }
   params_.push_back(param);
-  ASSERT(!HasDuplicateEntries());
+  RTC_CHECK(!HasDuplicateEntries());
 }
 
 void FeedbackParams::Intersect(const FeedbackParams& from) {
@@ -192,7 +192,7 @@ bool AudioCodec::Matches(const AudioCodec& codec) const {
 
 webrtc::RtpCodecParameters AudioCodec::ToCodecParameters() const {
   webrtc::RtpCodecParameters codec_params = Codec::ToCodecParameters();
-  codec_params.channels = channels;
+  codec_params.channels = static_cast<int>(channels);
   return codec_params;
 }
 
diff --git a/webrtc/media/base/videocapturer.cc b/webrtc/media/base/videocapturer.cc
index e7d3e3a3d6..d129557bce 100644
--- a/webrtc/media/base/videocapturer.cc
+++ b/webrtc/media/base/videocapturer.cc
@@ -14,7 +14,6 @@
 
 #include <algorithm>
 
-#include "libyuv/scale_argb.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/systeminfo.h"