diff --git a/DEPS b/DEPS
index 1233bcb143..f65bcbb529 100644
--- a/DEPS
+++ b/DEPS
@@ -11,7 +11,7 @@ vars = {
   "googlecode_url": "http://%s.googlecode.com/svn",
   "sourceforge_url": "http://svn.code.sf.net/p/%(repo)s/code",
   "chromium_trunk" : "http://src.chromium.org/svn/trunk",
-  "chromium_revision": "226126",
+  "chromium_revision": "228675",
 
   # A small subset of WebKit is needed for the Android Python test framework.
   "webkit_trunk": "http://src.chromium.org/blink/trunk",
diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi
index 1207cc1956..73dda8dcdf 100644
--- a/webrtc/build/common.gypi
+++ b/webrtc/build/common.gypi
@@ -311,6 +311,12 @@
            }],
          ],
       }],
+      ['clang==1', {
+        'cflags!': [
+          # TODO(kjellander): Remove when Chromium's common.gypi enables it.
+          '-Wno-unused-const-variable',
+        ],
+      }],
     ], # conditions
     'direct_dependent_settings': {
       'include_dirs': [
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
index 8ffc306b86..b3d87c67ee 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -55,7 +55,6 @@ static const unsigned int ALSA_PLAYOUT_LATENCY = 40*1000; // in us
 static const unsigned int ALSA_CAPTURE_FREQ = 48000;
 static const unsigned int ALSA_CAPTURE_CH = 2;
 static const unsigned int ALSA_CAPTURE_LATENCY = 40*1000; // in us
-static const unsigned int ALSA_PLAYOUT_WAIT_TIMEOUT = 5; // in ms
 static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
 
 #define FUNC_GET_NUM_OF_DEVICE 0
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index d8870a5275..fffc498697 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -55,12 +55,12 @@ namespace {
 // be set to true with the command-line switch --write_ref_data.
 #ifdef WEBRTC_AUDIOPROC_BIT_EXACT
 bool write_ref_data = false;
+const int kChannels[] = {1, 2};
+const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
 #endif
 
 const int kSampleRates[] = {8000, 16000, 32000};
 const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
-const int kChannels[] = {1, 2};
-const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
 
 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
 // AECM doesn't support super-wb.
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 39c5c6d748..be8f2fcdcc 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -20,7 +20,6 @@ const int kPacketSize1 = 100;
 const int kPacketSize2 = 300;
 const uint32_t kSsrc1 = 1;
 const uint32_t kSsrc2 = 2;
-const uint32_t kSsrc3 = 3;
 
 class ReceiveStatisticsTest : public ::testing::Test {
  public:
diff --git a/webrtc/video_engine/test/libvietest/testbed/fake_network_pipe.cc b/webrtc/video_engine/test/libvietest/testbed/fake_network_pipe.cc
index 6e6b9121db..b1c8eb2920 100644
--- a/webrtc/video_engine/test/libvietest/testbed/fake_network_pipe.cc
+++ b/webrtc/video_engine/test/libvietest/testbed/fake_network_pipe.cc
@@ -19,7 +19,6 @@
 
 namespace webrtc {
 
-const int kNetworkProcessMaxWaitTime = 10;
 const double kPi = 3.14159265;
 
 static int GaussianRandom(int mean_delay_ms, int standard_deviation_ms) {
diff --git a/webrtc/voice_engine/test/auto_test/standard/network_test.cc b/webrtc/voice_engine/test/auto_test/standard/network_test.cc
index 79daf59b25..f189d57144 100644
--- a/webrtc/voice_engine/test/auto_test/standard/network_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/network_test.cc
@@ -15,9 +15,6 @@
 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
 #include "webrtc/voice_engine/test/auto_test/voe_test_interface.h"
 
-static const int kDefaultRtpPort = 8000;
-static const int kDefaultRtcpPort = 8001;
-
 class NetworkTest : public AfterStreamingFixture {
 };
 
diff --git a/webrtc/voice_engine/test/auto_test/voe_extended_test.cc b/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
index 9b5b6d5d64..d656d5110f 100644
--- a/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
+++ b/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
@@ -29,16 +29,6 @@ using namespace test;
 
 namespace voetest {
 
-// Set this flag to ensure that test packets are transmitted to
-// RemoteIP::RemotePort during tests of SetSendToS and SetSendGQos. Requires
-// receiver at the remote side and Wireshark with a proper ip.src filter.
-#define _SEND_TO_REMOTE_IP_
-
-#ifdef _SEND_TO_REMOTE_IP_
-const int RemotePort = 12345; // transmit to this UDP port
-const char* RemoteIP = "192.168.200.1"; // transmit to this IP address
-#endif
-
 #ifdef WEBRTC_ANDROID
 // Global pointers
 extern void* globalJavaVM;