diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index 695cb8d5d0..590450bd29 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -12,7 +12,6 @@ import("../build/webrtc.gni")
 group("media") {
   public_deps = [
     ":rtc_media",
-    ":rtc_media_base",
   ]
 }
 
@@ -42,7 +41,7 @@ if (is_linux && rtc_use_gtk) {
   }
 }
 
-rtc_static_library("rtc_media_base") {
+rtc_static_library("rtc_media") {
   defines = []
   libs = []
   deps = []
@@ -82,38 +81,6 @@ rtc_static_library("rtc_media_base") {
     "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 = [
     "engine/internalencoderfactory.cc",
     "engine/internalencoderfactory.h",
     "engine/nullwebrtcvideoengine.h",
@@ -199,14 +166,15 @@ 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 aac21002b6..0320e58c72 100644
--- a/webrtc/media/base/codec.cc
+++ b/webrtc/media/base/codec.cc
@@ -13,7 +13,7 @@
 #include <algorithm>
 #include <sstream>
 
-#include "webrtc/base/checks.h"
+#include "webrtc/base/common.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);
-  RTC_CHECK(!HasDuplicateEntries());
+  ASSERT(!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 = static_cast<int>(channels);
+  codec_params.channels = channels;
   return codec_params;
 }