generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Roll chromium_revision 226126:228675 and fix clang warnings

Browse Source 
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@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kjellander@webrtc.org 2013-10-15 20:10:17 +00:00
parent e5021fe590
commit 3555303cb0
8 changed files with 9 additions and 19 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
DEPS
View File

@ -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",

6
webrtc/build/common.gypi
View File

@ -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': [

1
webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
View File

@ -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

4
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
View File

@ -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.

1
webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
View File

@ -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:

1
webrtc/video_engine/test/libvietest/testbed/fake_network_pipe.cc
View File

@ -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) {

3
webrtc/voice_engine/test/auto_test/standard/network_test.cc
View File

@ -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 {
};

10
webrtc/voice_engine/test/auto_test/voe_extended_test.cc
View File

@ -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;