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Removed warnings on Windows and enabled warnings-as-errors on Windows.

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TEST=

Review URL: https://webrtc-codereview.appspot.com/377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1624 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org 2012-02-07 14:56:45 +00:00
parent 87885e8409
commit 78088c2f36
11 changed files with 36 additions and 37 deletions
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11
src/build/common.gypi
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@ -147,12 +147,11 @@
'WEBRTC_TARGET_PC',
],
# TODO(andrew): remove this block when possible.
'msvs_disabled_warnings': [4389], # Signed/unsigned mismatch.
'msvs_settings': {
'VCCLCompilerTool': {
'WarnAsError': 'false',
},
},
# 4389: Signed/unsigned mismatch.
# 4373: MSVC legacy warning for ignoring const / volatile in
# signatures. TODO(phoglund): get rid of 4373 supression when
# http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
'msvs_disabled_warnings': [4389, 4373],
}],
], # conditions
}, # target_defaults

4
src/common_audio/signal_processing/signal_processing_unittest.cc
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@ -41,12 +41,12 @@ TEST_F(SplTest, MacroTest) {
EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
EXPECT_EQ(-2147483645u, WEBRTC_SPL_UMUL(a, b));
EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
b = WEBRTC_SPL_WORD16_MAX >> 1;
EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_RSFT16(a, b));
EXPECT_EQ(1073627139u, WEBRTC_SPL_UMUL_16_16(a, b));
EXPECT_EQ(16382u, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
EXPECT_EQ(-49149u, WEBRTC_SPL_UMUL_32_16(a, b));
EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));

4
src/modules/audio_device/main/source/win/audio_device_utility_win.cc
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@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -79,7 +79,7 @@ WebRtc_Word32 AudioDeviceUtilityWindows::Init()
if (WideCharToMultiByte(CP_UTF8, 0, szOS, -1, os, STRING_MAX_SIZE, NULL, NULL) == 0)
{
DWORD err = GetLastError();
sprintf(os, "Could not get OS info");
strncpy(os, "Could not get OS info", STRING_MAX_SIZE);
}
// DEBUG_PRINTP("OS info: %s\n", os);
WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, " OS info: %s", os);

4
src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -39,7 +39,7 @@ class TestTransport : public Transport {
(WebRtc_Word32)packetLength,
true); // Allow non-compound RTCP
EXPECT_EQ(true, rtcpParser.IsValid());
EXPECT_TRUE(rtcpParser.IsValid());
RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
&rtcpParser));

2
src/modules/rtp_rtcp/source/rtcp_sender_test.cc
View File

@ -72,7 +72,7 @@ class TestTransport : public Transport,
(WebRtc_Word32)packet_len,
true); // Allow non-compound RTCP
EXPECT_EQ(true, rtcpParser.IsValid());
EXPECT_TRUE(rtcpParser.IsValid());
RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
&rtcpParser));

10
src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -139,8 +139,8 @@ void RtpFormatVp8TestHelper::CheckHeader(bool frag_start) {
EXPECT_BIT_X_EQ(buffer_[0], 0);
}
EXPECT_BIT_N_EQ(buffer_[0], hdr_info_->nonReference);
EXPECT_BIT_S_EQ(buffer_[0], frag_start);
EXPECT_BIT_N_EQ(buffer_[0], hdr_info_->nonReference ? 1 : 0);
EXPECT_BIT_S_EQ(buffer_[0], frag_start ? 1 : 0);
// Check partition index.
if (!sloppy_partitioning_) {
@ -201,11 +201,11 @@ void RtpFormatVp8TestHelper::CheckTIDAndKeyIdx() {
if (hdr_info_->temporalIdx != kNoTemporalIdx) {
EXPECT_BIT_T_EQ(buffer_[1], 1);
EXPECT_TID_EQ(buffer_[payload_start_], hdr_info_->temporalIdx);
EXPECT_BIT_Y_EQ(buffer_[payload_start_], hdr_info_->layerSync);
EXPECT_BIT_Y_EQ(buffer_[payload_start_], hdr_info_->layerSync ? 1 : 0);
} else {
EXPECT_BIT_T_EQ(buffer_[1], 0);
EXPECT_TID_EQ(buffer_[payload_start_], 0);
EXPECT_BIT_Y_EQ(buffer_[payload_start_], false);
EXPECT_BIT_Y_EQ(buffer_[payload_start_], 0);
}
if (hdr_info_->keyIdx != kNoKeyIdx) {
EXPECT_BIT_K_EQ(buffer_[1], 1);

2
src/modules/rtp_rtcp/source/rtp_sender_test.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source

6
src/modules/rtp_rtcp/source/transmission_bucket.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -57,7 +57,7 @@ void TransmissionBucket::UpdateBytesPerInterval(
const uint16_t target_bitrate_kbps) {
webrtc::CriticalSectionScoped cs(*critsect_);
const float kMargin = 1.05;
const float kMargin = 1.05f;
uint32_t bytes_per_interval =
kMargin * (target_bitrate_kbps * delta_time_ms / 8);
@ -93,7 +93,7 @@ int32_t TransmissionBucket::GetNextPacket() {
return seq_num;
}
const float kFrameComplete = 0.80;
const float kFrameComplete = 0.80f;
if (num_bytes * kFrameComplete > bytes_rem_total_) {
// Packet does not fit.
return -1;

2
src/modules/video_coding/codecs/vp8/main/source/vp8.cc
View File

@ -402,7 +402,7 @@ void VP8Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
vp8Info->pictureId = picture_id_;
vp8Info->simulcastIdx = 0;
vp8Info->keyIdx = kNoKeyIdx; // TODO(hlundin) populate this
vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE);
vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) != 0;
#if WEBRTC_LIBVPX_VERSION >= 971
if (temporal_layers_) {
temporal_layers_->PopulateCodecSpecific(

4
src/video_engine/main/test/WindowsTest/ChannelDlg.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -205,7 +205,7 @@ BOOL CDXChannelDlg::OnInitDialog()
m_ctrlMinFrameRate.SetCurSel(25);
// Codec sizes
for(VideoSize i=VideoSize::UNDEFINED;i<VideoSize::NUMBER_OF_VIDEO_SIZE;i=VideoSize(i+1))
for(VideoSize i=UNDEFINED;i<NUMBER_OF_VIDEO_SIZE;i=VideoSize(i+1))
{
char sizeStr[64];
int width=0;

24
src/voice_engine/main/test/auto_test/standard/volume_test.cc
View File

@ -69,22 +69,22 @@ TEST_F(VolumeTest, ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAcgOff) {
}
TEST_F(VolumeTest, ChannelScalingIsOneByDefault) {
float scaling = -1.0;
float scaling = -1.0f;
EXPECT_EQ(0, voe_volume_control_->GetChannelOutputVolumeScaling(
channel_, scaling));
EXPECT_FLOAT_EQ(1.0, scaling);
EXPECT_FLOAT_EQ(1.0f, scaling);
}
TEST_F(VolumeTest, ManualCanSetChannelScaling) {
EXPECT_EQ(0, voe_volume_control_->SetChannelOutputVolumeScaling(
channel_, 0.1));
channel_, 0.1f));
float scaling = 1.0;
float scaling = 1.0f;
EXPECT_EQ(0, voe_volume_control_->GetChannelOutputVolumeScaling(
channel_, scaling));
EXPECT_FLOAT_EQ(0.1, scaling);
EXPECT_FLOAT_EQ(0.1f, scaling);
TEST_LOG("Channel scaling set to 0.1: audio should be barely audible.\n");
Sleep(2000);
@ -211,24 +211,24 @@ TEST_F(VolumeTest, ChannelsAreNotPannedByDefault) {
TEST_F(VolumeTest, ManualTestChannelPanning) {
TEST_LOG("Panning left.\n");
EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.8, 0.1));
EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.8f, 0.1f));
Sleep(1000);
TEST_LOG("Back to center.\n");
EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 1.0, 1.0));
EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 1.0f, 1.0f));
Sleep(1000);
TEST_LOG("Panning right.\n");
EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.1, 0.8));
EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.1f, 0.8f));
Sleep(1000);
// To finish, verify that the getter works.
float left = 0.0;
float right = 0.0;
float left = 0.0f;
float right = 0.0f;
EXPECT_EQ(0, voe_volume_control_->GetOutputVolumePan(channel_, left, right));
EXPECT_FLOAT_EQ(0.1, left);
EXPECT_FLOAT_EQ(0.8, right);
EXPECT_FLOAT_EQ(0.1f, left);
EXPECT_FLOAT_EQ(0.8f, right);
}
#endif // !WEBRTC_ANDROID && !MAC_IPHONE