diff --git a/DEPS b/DEPS index d2850e499c..494e5d344c 100644 --- a/DEPS +++ b/DEPS @@ -6,7 +6,7 @@ vars = { 'extra_gyp_flag': '-Dextra_gyp_flag=0', 'chromium_git': 'https://chromium.googlesource.com', - 'chromium_revision': '7a4fb8d231fde514f4e687f949cfa951df4cfb7c', + 'chromium_revision': 'f527e8602d8b4cb18c973d143cb6e33cb4ea6922', } # NOTE: Prefer revision numbers to tags for svn deps. Use http rather than diff --git a/third_party/winsdk_samples/winsdk_samples.gyp b/third_party/winsdk_samples/winsdk_samples.gyp index 12bc265ec7..6de0dead7c 100644 --- a/third_party/winsdk_samples/winsdk_samples.gyp +++ b/third_party/winsdk_samples/winsdk_samples.gyp @@ -109,6 +109,17 @@ ], }, }, + 'direct_dependent_settings': { + 'msvs_settings': { + 'VCCLCompilerTool': { + 'AdditionalOptions': [ + # Disable warnings failing when compiling with Clang on Windows. + # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 + '-Wno-ignored-qualifiers', + ], + }, + }, + }, },], ], # conditions. }, diff --git a/webrtc/base/base.gyp b/webrtc/base/base.gyp index c9182632ab..3d2ef629d1 100644 --- a/webrtc/base/base.gyp +++ b/webrtc/base/base.gyp @@ -552,6 +552,7 @@ 'AdditionalOptions': [ # Disable warnings failing when compiling with Clang on Windows. # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 + '-Wno-sign-compare', '-Wno-missing-braces', ], }, diff --git a/webrtc/base/base_tests.gyp b/webrtc/base/base_tests.gyp index 5d73d50756..cf56bcedf5 100644 --- a/webrtc/base/base_tests.gyp +++ b/webrtc/base/base_tests.gyp @@ -141,6 +141,7 @@ # Disable warnings failing when compiling with Clang on Windows. # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 '-Wno-missing-braces', + '-Wno-sign-compare', '-Wno-unused-const-variable', ], }, diff --git a/webrtc/modules/audio_device/test/audio_device_test_api.cc b/webrtc/modules/audio_device/test/audio_device_test_api.cc index 26a2dcd8c4..f2861ec3e5 100644 --- a/webrtc/modules/audio_device/test/audio_device_test_api.cc +++ b/webrtc/modules/audio_device/test/audio_device_test_api.cc @@ -915,7 +915,7 @@ TEST_F(AudioDeviceAPITest, SpeakerVolumeTests) { EXPECT_EQ(0, audio_device_->MaxSpeakerVolume(&maxVolume)); EXPECT_EQ(0, audio_device_->MinSpeakerVolume(&minVolume)); EXPECT_EQ(0, audio_device_->SpeakerVolumeStepSize(&stepSize)); - for (vol = minVolume; vol < (int)maxVolume; vol += 20*stepSize) { + for (vol = minVolume; vol < (unsigned int)maxVolume; vol += 20*stepSize) { EXPECT_EQ(0, audio_device_->SetSpeakerVolume(vol)); EXPECT_EQ(0, audio_device_->SpeakerVolume(&volume)); CheckVolume(volume, vol); @@ -1067,7 +1067,7 @@ TEST_F(AudioDeviceAPITest, MAYBE_MicrophoneVolumeTests) { EXPECT_EQ(0, audio_device_->MaxMicrophoneVolume(&maxVolume)); EXPECT_EQ(0, audio_device_->MinMicrophoneVolume(&minVolume)); EXPECT_EQ(0, audio_device_->MicrophoneVolumeStepSize(&stepSize)); - for (vol = minVolume; vol < (int)maxVolume; vol += 10*stepSize) + for (vol = minVolume; vol < (unsigned int)maxVolume; vol += 10*stepSize) { EXPECT_EQ(0, audio_device_->SetMicrophoneVolume(vol)); EXPECT_EQ(0, audio_device_->MicrophoneVolume(&volume)); diff --git a/webrtc/modules/audio_device/win/audio_device_core_win.cc b/webrtc/modules/audio_device/win/audio_device_core_win.cc index 4b75922109..811a04d2fe 100644 --- a/webrtc/modules/audio_device/win/audio_device_core_win.cc +++ b/webrtc/modules/audio_device/win/audio_device_core_win.cc @@ -2237,9 +2237,9 @@ int32_t AudioDeviceWindowsCore::InitPlayout() hr = S_FALSE; // Iterate over frequencies and channels, in order of priority - for (int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++) + for (unsigned int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++) { - for (int chan = 0; chan < sizeof(_playChannelsPrioList)/sizeof(_playChannelsPrioList[0]); chan++) + for (unsigned int chan = 0; chan < sizeof(_playChannelsPrioList)/sizeof(_playChannelsPrioList[0]); chan++) { Wfx.nChannels = _playChannelsPrioList[chan]; Wfx.nSamplesPerSec = freqs[freq]; @@ -2574,9 +2574,9 @@ int32_t AudioDeviceWindowsCore::InitRecording() hr = S_FALSE; // Iterate over frequencies and channels, in order of priority - for (int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++) + for (unsigned int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++) { - for (int chan = 0; chan < sizeof(_recChannelsPrioList)/sizeof(_recChannelsPrioList[0]); chan++) + for (unsigned int chan = 0; chan < sizeof(_recChannelsPrioList)/sizeof(_recChannelsPrioList[0]); chan++) { Wfx.nChannels = _recChannelsPrioList[chan]; Wfx.nSamplesPerSec = freqs[freq]; @@ -5078,7 +5078,7 @@ char* AudioDeviceWindowsCore::WideToUTF8(const TCHAR* src) const { const size_t kStrLen = sizeof(_str); memset(_str, 0, kStrLen); // Get required size (in bytes) to be able to complete the conversion. - int required_size = WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0); + unsigned int required_size = (unsigned int)WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0); if (required_size <= kStrLen) { // Process the entire input string, including the terminating null char. diff --git a/webrtc/modules/audio_device/win/audio_device_wave_win.cc b/webrtc/modules/audio_device/win/audio_device_wave_win.cc index 6f4d7df397..8079051184 100644 --- a/webrtc/modules/audio_device/win/audio_device_wave_win.cc +++ b/webrtc/modules/audio_device/win/audio_device_wave_win.cc @@ -2737,7 +2737,7 @@ int32_t AudioDeviceWindowsWave::GetPlayoutBufferDelay(uint32_t& writtenSamples, msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs); } } - else if ((_writtenSamplesOld > POW2(31)) && (writtenSamples < 96000)) + else if ((_writtenSamplesOld > (unsigned long)POW2(31)) && (writtenSamples < 96000)) { // Wrap around as expected after having used all 32 bits. (But we still // test if the wrap around happened earlier which it should not) @@ -2754,7 +2754,7 @@ int32_t AudioDeviceWindowsWave::GetPlayoutBufferDelay(uint32_t& writtenSamples, msecInPlayoutBuffer = (int)((writtenSamples + POW2(i + 1) - playedSamples)/nSamplesPerMs); } - else if ((writtenSamples < 96000) && (playedSamples > POW2(31))) + else if ((writtenSamples < 96000) && (playedSamples > (unsigned long)POW2(31))) { // Wrap around has, as expected, happened for written_sampels before // playedSampels so we have to adjust for this until also playedSampels @@ -2953,7 +2953,7 @@ int32_t AudioDeviceWindowsWave::GetRecordingBufferDelay(uint32_t& readSamples, u if((_wrapCounter>200)){ // Do nothing, handled later } - else if((_rec_samples_old > POW2(31)) && (recSamples < 96000)) { + else if((_rec_samples_old > (unsigned long)POW2(31)) && (recSamples < 96000)) { WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 2 (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples); // Wrap around as expected after having used all 32 bits. _read_samples_old = readSamples; @@ -2962,7 +2962,7 @@ int32_t AudioDeviceWindowsWave::GetRecordingBufferDelay(uint32_t& readSamples, u return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs); - } else if((recSamples < 96000) && (readSamples > POW2(31))) { + } else if((recSamples < 96000) && (readSamples > (unsigned long)POW2(31))) { WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 3 (readSamples %d recSamples %d)",readSamples, recSamples); // Wrap around has, as expected, happened for rec_sampels before // readSampels so we have to adjust for this until also readSampels diff --git a/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc b/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc index 368b54c746..ae2d00faeb 100644 --- a/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc +++ b/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc @@ -2709,7 +2709,7 @@ char* AudioMixerManager::WideToUTF8(const TCHAR* src) const { const size_t kStrLen = sizeof(_str); memset(_str, 0, kStrLen); // Get required size (in bytes) to be able to complete the conversion. - int required_size = WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0); + unsigned int required_size = (unsigned int)WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0); if (required_size <= kStrLen) { // Process the entire input string, including the terminating null char. diff --git a/webrtc/modules/video_capture/windows/video_capture_ds.cc b/webrtc/modules/video_capture/windows/video_capture_ds.cc index b69e50121d..dcef2d4706 100644 --- a/webrtc/modules/video_capture/windows/video_capture_ds.cc +++ b/webrtc/modules/video_capture/windows/video_capture_ds.cc @@ -395,7 +395,7 @@ HRESULT VideoCaptureDS::ConnectDVCamera() hr = _graphBuilder->ConnectDirect(_outputDvPin, _inputSendPin, NULL); if (hr != S_OK) { - if (hr == 0x80070004) + if (hr == (long)0x80070004) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "Failed to connect the capture device, busy"); diff --git a/webrtc/modules/video_render/windows/video_render_direct3d9.cc b/webrtc/modules/video_render/windows/video_render_direct3d9.cc index 83835aebb8..b59b944e48 100644 --- a/webrtc/modules/video_render/windows/video_render_direct3d9.cc +++ b/webrtc/modules/video_render/windows/video_render_direct3d9.cc @@ -1125,12 +1125,6 @@ int32_t VideoRenderDirect3D9::SetBitmap(const void* bitMap, int32_t VideoRenderDirect3D9::GetGraphicsMemory(uint64_t& totalMemory, uint64_t& availableMemory) { - if (_totalMemory == -1 || _availableMemory == -1) - { - totalMemory = 0; - availableMemory = 0; - return -1; - } totalMemory = _totalMemory; availableMemory = _availableMemory; return 0;