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@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
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parent
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commit
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DEPS
webrtc
build
modules
audio_device/linux
audio_processing/test
rtp_rtcp/source
video_engine/test/libvietest/testbed
voice_engine/test/auto_test
2
DEPS
2
DEPS
@ -11,7 +11,7 @@ vars = {
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"googlecode_url": "http://%s.googlecode.com/svn",
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"sourceforge_url": "http://svn.code.sf.net/p/%(repo)s/code",
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"chromium_trunk" : "http://src.chromium.org/svn/trunk",
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"chromium_revision": "226126",
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"chromium_revision": "228675",
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# A small subset of WebKit is needed for the Android Python test framework.
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"webkit_trunk": "http://src.chromium.org/blink/trunk",
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@ -311,6 +311,12 @@
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}],
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],
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}],
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['clang==1', {
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'cflags!': [
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# TODO(kjellander): Remove when Chromium's common.gypi enables it.
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'-Wno-unused-const-variable',
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],
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}],
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], # conditions
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'direct_dependent_settings': {
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'include_dirs': [
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@ -55,7 +55,6 @@ static const unsigned int ALSA_PLAYOUT_LATENCY = 40*1000; // in us
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static const unsigned int ALSA_CAPTURE_FREQ = 48000;
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static const unsigned int ALSA_CAPTURE_CH = 2;
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static const unsigned int ALSA_CAPTURE_LATENCY = 40*1000; // in us
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static const unsigned int ALSA_PLAYOUT_WAIT_TIMEOUT = 5; // in ms
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static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
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#define FUNC_GET_NUM_OF_DEVICE 0
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@ -55,12 +55,12 @@ namespace {
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// be set to true with the command-line switch --write_ref_data.
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#ifdef WEBRTC_AUDIOPROC_BIT_EXACT
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bool write_ref_data = false;
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const int kChannels[] = {1, 2};
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const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
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#endif
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const int kSampleRates[] = {8000, 16000, 32000};
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const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
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const int kChannels[] = {1, 2};
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const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
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#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
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// AECM doesn't support super-wb.
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@ -20,7 +20,6 @@ const int kPacketSize1 = 100;
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const int kPacketSize2 = 300;
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const uint32_t kSsrc1 = 1;
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const uint32_t kSsrc2 = 2;
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const uint32_t kSsrc3 = 3;
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class ReceiveStatisticsTest : public ::testing::Test {
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public:
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@ -19,7 +19,6 @@
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namespace webrtc {
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const int kNetworkProcessMaxWaitTime = 10;
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const double kPi = 3.14159265;
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static int GaussianRandom(int mean_delay_ms, int standard_deviation_ms) {
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@ -15,9 +15,6 @@
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#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
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#include "webrtc/voice_engine/test/auto_test/voe_test_interface.h"
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static const int kDefaultRtpPort = 8000;
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static const int kDefaultRtcpPort = 8001;
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class NetworkTest : public AfterStreamingFixture {
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};
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@ -29,16 +29,6 @@ using namespace test;
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namespace voetest {
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// Set this flag to ensure that test packets are transmitted to
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// RemoteIP::RemotePort during tests of SetSendToS and SetSendGQos. Requires
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// receiver at the remote side and Wireshark with a proper ip.src filter.
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#define _SEND_TO_REMOTE_IP_
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#ifdef _SEND_TO_REMOTE_IP_
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const int RemotePort = 12345; // transmit to this UDP port
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const char* RemoteIP = "192.168.200.1"; // transmit to this IP address
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#endif
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#ifdef WEBRTC_ANDROID
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// Global pointers
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extern void* globalJavaVM;
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