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Fix win trybot errors due to r4729.

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The addition of logging.h in r4729 was causing the win trybot to fail
with "#pragma deprecated" errors in standard library headers. This
turned out to be due to including strsafe.h (via audio_device_config.h)
before sstream (via logging.h).

strsafe.h was only being included for the unused DEBUG_PRINT macro. I
removed all references to it.

This incidentally removes a bunch of other unneeded headers discovered
while trying to track the problem down.

This didn't show up in the commitbots; my guess is that the trybots are
using the VC10 toolchain and the commitbots the VC11 toolchain.

TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/2204004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4738 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2013-09-13 00:02:13 +00:00
parent 6a5cc9d899
commit 2553450959
7 changed files with 26 additions and 50 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/modules/audio_device/audio_device_buffer.cc
View File

@ -9,18 +9,16 @@
*/
#include "webrtc/modules/audio_device/audio_device_buffer.h"
#include <assert.h>
#include <string.h>
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/audio_device_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include <assert.h>
#include <stdlib.h>
#include <string.h>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
static const int kHighDelayThresholdMs = 300;

2
webrtc/modules/audio_device/audio_device_buffer.h
View File

@ -11,10 +11,8 @@
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/system_wrappers/interface/file_wrapper.h"
#include "webrtc/system_wrappers/interface/list_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {

20
webrtc/modules/audio_device/audio_device_config.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
// Enumerators
//
@ -26,19 +26,5 @@ enum { GET_MIC_VOLUME_INTERVAL_MS = 1000 };
#endif
#endif
#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
#include <windows.h>
#include <tchar.h>
#include <strsafe.h>
#define DEBUG_PRINT(...) \
{ \
TCHAR msg[256]; \
StringCchPrintf(msg, 256, __VA_ARGS__); \
OutputDebugString(msg); \
}
#else
#define DEBUG_PRINT(exp) ((void)0)
#endif
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_

1
webrtc/modules/audio_device/linux/audio_device_utility_linux.cc
View File

@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_device/audio_device_config.h" // DEBUG_PRINT()
#include "webrtc/modules/audio_device/linux/audio_device_utility_linux.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"

1
webrtc/modules/audio_device/mac/audio_device_utility_mac.cc
View File

@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_device/audio_device_config.h" // DEBUG_PRINT()
#include "webrtc/modules/audio_device/mac/audio_device_utility_mac.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"

23
webrtc/modules/audio_device/win/audio_device_utility_win.cc
View File

@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/win/audio_device_utility_win.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
@ -79,10 +78,8 @@ int32_t AudioDeviceUtilityWindows::Init()
{
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);
#else
// DEBUG_PRINTP("OS info: %s\n", szOS);
WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, " OS info: %s", szOS);
#endif
}
@ -125,7 +122,7 @@ BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
// Windows Server 2008 R2 6.1
// Windows Server 2008 6.0
// Windows Vista 6.0
// - - - - - - - - - - - - - - - - - - -
// - - - - - - - - - - - - - - - - - - -
// Windows Server 2003 R2 5.2
// Windows Server 2003 5.2
// Windows XP 5.1
@ -140,7 +137,7 @@ BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
// Windows Vista or Server 2008
if (osvi.wProductType == VER_NT_WORKSTATION)
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Vista "));
else
else
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 " ));
}
@ -149,7 +146,7 @@ BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
// Windows 7 or Server 2008 R2
if (osvi.wProductType == VER_NT_WORKSTATION)
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows 7 "));
else
else
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 R2 " ));
}
}
@ -164,7 +161,7 @@ BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows XP "));
if (osvi.wSuiteMask & VER_SUITE_PERSONAL)
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Home Edition" ));
else
else
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
}
@ -176,7 +173,7 @@ BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
{
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
}
else
else
{
if (osvi.wSuiteMask & VER_SUITE_DATACENTER)
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Datacenter Server" ));
@ -206,24 +203,24 @@ BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
pGNSI = (PGNSI) GetProcAddress(GetModuleHandle(TEXT("kernel32.dll")), "GetNativeSystemInfo");
if (NULL != pGNSI)
pGNSI(&si);
else
else
GetSystemInfo(&si);
// Add 64-bit or 32-bit for OS versions "later than" Vista
//
if (osvi.dwMajorVersion >= 6)
{
if ((si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_AMD64) ||
if ((si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_AMD64) ||
(si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_IA64))
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( ", 64-bit" ));
else if (si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_INTEL )
StringCchCat(pszOS, STRING_MAX_SIZE, TEXT(", 32-bit"));
}
return TRUE;
return TRUE;
}
else
{
{
return FALSE;
}
}

19
webrtc/modules/audio_device/win/audio_device_wave_win.cc
View File

@ -231,8 +231,8 @@ int32_t AudioDeviceWindowsWave::Init()
}
const char* threadName = "webrtc_audio_module_thread";
_ptrThread = ThreadWrapper::CreateThread(ThreadFunc,
this,
_ptrThread = ThreadWrapper::CreateThread(ThreadFunc,
this,
kRealtimePriority,
threadName);
if (_ptrThread == NULL)
@ -364,7 +364,7 @@ int32_t AudioDeviceWindowsWave::Terminate()
return -1;
}
_critSect.Enter();
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" volume getter thread is now closed");
SetEvent(_hShutdownSetVolumeEvent);
@ -417,7 +417,7 @@ DWORD AudioDeviceWindowsWave::DoGetCaptureVolumeThread()
while (1)
{
DWORD waitResult = WaitForSingleObject(waitObject,
DWORD waitResult = WaitForSingleObject(waitObject,
GET_MIC_VOLUME_INTERVAL_MS);
switch (waitResult)
{
@ -440,7 +440,7 @@ DWORD AudioDeviceWindowsWave::DoGetCaptureVolumeThread()
_critSect.Enter();
if (_ptrAudioBuffer)
{
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
_critSect.Leave();
}
@ -472,11 +472,11 @@ DWORD AudioDeviceWindowsWave::DoSetCaptureVolumeThread()
_critSect.Leave();
if (SetMicrophoneVolume(newMicLevel) == -1)
{
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" the required modification of the microphone volume failed");
}
}
}
return 0;
}
@ -1269,7 +1269,7 @@ int32_t AudioDeviceWindowsWave::MaxMicrophoneVolume(uint32_t& maxVolume) const
// (1) API GetLineControl() returns failure at querying the max Mic level.
// (2) API GetLineControl() returns maxVolume as zero in rare cases.
// Both cases show we don't have access to the mixer controls.
// We return -1 here to indicate that.
// We return -1 here to indicate that.
if (_maxMicVolume == 0)
{
return -1;
@ -2809,7 +2809,6 @@ int32_t AudioDeviceWindowsWave::GetPlayoutBufferDelay(uint32_t& writtenSamples,
// derive remaining amount (in ms) of data in the playout buffer
msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
// DEBUG_PRINTP("msecInPlayoutBuffer=%u\n", msecInPlayoutBuffer);
playedDifference = (long) (_playedSamplesOld - playedSamples);
@ -3393,7 +3392,7 @@ int32_t AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
// The VQE will only deliver non-zero microphone levels when a change is needed.
WEBRTC_TRACE(kTraceStream, kTraceUtility, _id,"AGC change of volume: => new=%u", newMicLevel);
// We store this outside of the audio buffer to avoid
// We store this outside of the audio buffer to avoid
// having it overwritten by the getter thread.
_newMicLevel = newMicLevel;
SetEvent(_hSetCaptureVolumeEvent);