diff --git a/src/build/common.gypi b/src/build/common.gypi index 297ab2619..bd2f5120a 100644 --- a/src/build/common.gypi +++ b/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. - '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 diff --git a/src/common_audio/signal_processing/signal_processing_unittest.cc b/src/common_audio/signal_processing/signal_processing_unittest.cc index 606b2694b..f6d11233d 100644 --- a/src/common_audio/signal_processing/signal_processing_unittest.cc +++ b/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)); diff --git a/src/modules/audio_device/main/source/win/audio_device_utility_win.cc b/src/modules/audio_device/main/source/win/audio_device_utility_win.cc index cd9c7ad12..3ef85ea19 100644 --- a/src/modules/audio_device/main/source/win/audio_device_utility_win.cc +++ 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); diff --git a/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc index aa1f7a499..7f3eb2ec5 100644 --- a/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc +++ b/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)); diff --git a/src/modules/rtp_rtcp/source/rtcp_sender_test.cc b/src/modules/rtp_rtcp/source/rtcp_sender_test.cc index 96478a42c..81e8665dc 100644 --- a/src/modules/rtp_rtcp/source/rtcp_sender_test.cc +++ b/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)); diff --git a/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc index 35d9109ec..59e76ee87 100644 --- a/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc +++ b/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_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); diff --git a/src/modules/rtp_rtcp/source/rtp_sender_test.cc b/src/modules/rtp_rtcp/source/rtp_sender_test.cc index 90cfb72ac..4e8c78c10 100644 --- a/src/modules/rtp_rtcp/source/rtp_sender_test.cc +++ b/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 diff --git a/src/modules/rtp_rtcp/source/transmission_bucket.cc b/src/modules/rtp_rtcp/source/transmission_bucket.cc index 4ba280b5b..e79d227a4 100644 --- a/src/modules/rtp_rtcp/source/transmission_bucket.cc +++ b/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; diff --git a/src/modules/video_coding/codecs/vp8/main/source/vp8.cc b/src/modules/video_coding/codecs/vp8/main/source/vp8.cc index 0f29843f9..599d3a53f 100644 --- a/src/modules/video_coding/codecs/vp8/main/source/vp8.cc +++ b/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( diff --git a/src/video_engine/main/test/WindowsTest/ChannelDlg.cc b/src/video_engine/main/test/WindowsTest/ChannelDlg.cc index 0f5c5c506..cccfa8ee2 100644 --- a/src/video_engine/main/test/WindowsTest/ChannelDlg.cc +++ b/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;iGetChannelOutputVolumeScaling( 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