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

BUG=
TEST=

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1624 4adac7df-926f-26a2-2b94-8c16560cd09d

src/build/common.gypi

@@ -147,12 +147,11 @@
           'WEBRTC_TARGET_PC',
         ],
         # TODO(andrew): remove this block when possible.
-        'msvs_disabled_warnings': [4389], # Signed/unsigned mismatch.
+        # 4389: Signed/unsigned mismatch.
-        'msvs_settings': {
+        # 4373: MSVC legacy warning for ignoring const / volatile in
-          'VCCLCompilerTool': {
+        # signatures. TODO(phoglund): get rid of 4373 supression when
-            'WarnAsError': 'false',
+        # http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
-          },
+        'msvs_disabled_warnings': [4389, 4373],
-        },
       }],
     ], # conditions
   }, # target_defaults

src/common_audio/signal_processing/signal_processing_unittest.cc

@@ -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));
 

src/modules/audio_device/main/source/win/audio_device_utility_win.cc

@@ -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);

src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc

@@ -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));

src/modules/rtp_rtcp/source/rtcp_sender_test.cc

@@ -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));

src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc

@@ -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_N_EQ(buffer_[0], hdr_info_->nonReference ? 1 : 0);
-  EXPECT_BIT_S_EQ(buffer_[0], frag_start);
+  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);

src/modules/rtp_rtcp/source/rtp_sender_test.cc

@@ -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

src/modules/rtp_rtcp/source/transmission_bucket.cc

@@ -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;

src/modules/video_coding/codecs/vp8/main/source/vp8.cc

@@ -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(

src/video_engine/main/test/WindowsTest/ChannelDlg.cc

@@ -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;

src/voice_engine/main/test/auto_test/standard/volume_test.cc

@@ -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 left = 0.0f;
-  float right = 0.0;
+  float right = 0.0f;
 
   EXPECT_EQ(0, voe_volume_control_->GetOutputVolumePan(channel_, left, right));
-  EXPECT_FLOAT_EQ(0.1, left);
+  EXPECT_FLOAT_EQ(0.1f, left);
-  EXPECT_FLOAT_EQ(0.8, right);
+  EXPECT_FLOAT_EQ(0.8f, right);
 }
 
 #endif  // !WEBRTC_ANDROID && !MAC_IPHONE