Roll chromium_revision 226126:228675 and fix clang warnings
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.
This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.
TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2400004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/build/common.gypi b/build/common.gypi
index 1207cc1..73dda8d 100644
--- a/build/common.gypi
+++ b/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/modules/audio_device/linux/audio_device_alsa_linux.cc b/modules/audio_device/linux/audio_device_alsa_linux.cc
index 8ffc306..b3d87c6 100644
--- a/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -55,7 +55,6 @@
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/modules/audio_processing/test/audio_processing_unittest.cc b/modules/audio_processing/test/audio_processing_unittest.cc
index d8870a5..fffc498 100644
--- a/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/modules/audio_processing/test/audio_processing_unittest.cc
@@ -55,12 +55,12 @@
// 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/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 39c5c6d..be8f2fc 100644
--- a/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -20,7 +20,6 @@
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/video_engine/test/libvietest/testbed/fake_network_pipe.cc b/video_engine/test/libvietest/testbed/fake_network_pipe.cc
index 6e6b912..b1c8eb2 100644
--- a/video_engine/test/libvietest/testbed/fake_network_pipe.cc
+++ b/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/voice_engine/test/auto_test/standard/network_test.cc b/voice_engine/test/auto_test/standard/network_test.cc
index 79daf59..f189d57 100644
--- a/voice_engine/test/auto_test/standard/network_test.cc
+++ b/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/voice_engine/test/auto_test/voe_extended_test.cc b/voice_engine/test/auto_test/voe_extended_test.cc
index 9b5b6d5..d656d51 100644
--- a/voice_engine/test/auto_test/voe_extended_test.cc
+++ b/voice_engine/test/auto_test/voe_extended_test.cc
@@ -29,16 +29,6 @@
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;