Ensures that we can build using VS 2012 on Windows.
See more details at https://code.google.com/p/webrtc/issues/detail?id=1146& TBR=Niklas BUG=1146 Review URL: https://webrtc-codereview.appspot.com/939028 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3162 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
@@ -160,7 +160,7 @@ Channel::CalcStatistics(
|
|||||||
_lastPayloadType = rtpInfo.header.payloadType;
|
_lastPayloadType = rtpInfo.header.payloadType;
|
||||||
|
|
||||||
bool newPayload = true;
|
bool newPayload = true;
|
||||||
ACMTestPayloadStats* currentPayloadStr;
|
ACMTestPayloadStats* currentPayloadStr = NULL;
|
||||||
for(n = 0; n < MAX_NUM_PAYLOADS; n++)
|
for(n = 0; n < MAX_NUM_PAYLOADS; n++)
|
||||||
{
|
{
|
||||||
if(rtpInfo.header.payloadType == _payloadStats[n].payloadType)
|
if(rtpInfo.header.payloadType == _payloadStats[n].payloadType)
|
||||||
|
|||||||
@@ -3471,6 +3471,8 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
|
|||||||
|
|
||||||
_Lock();
|
_Lock();
|
||||||
|
|
||||||
|
IAudioClock* clock = NULL;
|
||||||
|
|
||||||
// Get size of rendering buffer (length is expressed as the number of audio frames the buffer can hold).
|
// Get size of rendering buffer (length is expressed as the number of audio frames the buffer can hold).
|
||||||
// This value is fixed during the rendering session.
|
// This value is fixed during the rendering session.
|
||||||
//
|
//
|
||||||
@@ -3525,7 +3527,6 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
|
|||||||
|
|
||||||
_writtenSamples += bufferLength;
|
_writtenSamples += bufferLength;
|
||||||
|
|
||||||
IAudioClock* clock = NULL;
|
|
||||||
hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
|
hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
|
||||||
if (FAILED(hr)) {
|
if (FAILED(hr)) {
|
||||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
|
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
|
||||||
@@ -3629,7 +3630,7 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
|
|||||||
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
|
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
|
||||||
goto Exit;
|
goto Exit;
|
||||||
}
|
}
|
||||||
if (nSamples != _playBlockSize)
|
if (nSamples != static_cast<WebRtc_Word32>(_playBlockSize))
|
||||||
{
|
{
|
||||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "nSamples(%d) != _playBlockSize(%d)", nSamples, _playBlockSize);
|
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "nSamples(%d) != _playBlockSize(%d)", nSamples, _playBlockSize);
|
||||||
}
|
}
|
||||||
@@ -3852,7 +3853,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
|
|||||||
// TODO(andrew): verify that this is always satisfied. It might
|
// TODO(andrew): verify that this is always satisfied. It might
|
||||||
// be that ProcessOutput will try to return more than 10 ms if
|
// be that ProcessOutput will try to return more than 10 ms if
|
||||||
// we fail to call it frequently enough.
|
// we fail to call it frequently enough.
|
||||||
assert(kSamplesProduced == _recBlockSize);
|
assert(kSamplesProduced == static_cast<int>(_recBlockSize));
|
||||||
assert(sizeof(BYTE) == sizeof(WebRtc_Word8));
|
assert(sizeof(BYTE) == sizeof(WebRtc_Word8));
|
||||||
_ptrAudioBuffer->SetRecordedBuffer(
|
_ptrAudioBuffer->SetRecordedBuffer(
|
||||||
reinterpret_cast<WebRtc_Word8*>(data),
|
reinterpret_cast<WebRtc_Word8*>(data),
|
||||||
@@ -4904,6 +4905,10 @@ WebRtc_Word32 AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dat
|
|||||||
|
|
||||||
HRESULT hr = S_OK;
|
HRESULT hr = S_OK;
|
||||||
IMMDeviceCollection *pCollection = NULL;
|
IMMDeviceCollection *pCollection = NULL;
|
||||||
|
IMMDevice *pEndpoint = NULL;
|
||||||
|
IPropertyStore *pProps = NULL;
|
||||||
|
IAudioEndpointVolume* pEndpointVolume = NULL;
|
||||||
|
LPWSTR pwszID = NULL;
|
||||||
|
|
||||||
// Generate a collection of audio endpoint devices in the system.
|
// Generate a collection of audio endpoint devices in the system.
|
||||||
// Get states for *all* endpoint devices.
|
// Get states for *all* endpoint devices.
|
||||||
@@ -4917,11 +4922,7 @@ WebRtc_Word32 AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dat
|
|||||||
|
|
||||||
// use the IMMDeviceCollection interface...
|
// use the IMMDeviceCollection interface...
|
||||||
|
|
||||||
UINT count;
|
UINT count = 0;
|
||||||
IMMDevice *pEndpoint = NULL;
|
|
||||||
IPropertyStore *pProps = NULL;
|
|
||||||
IAudioEndpointVolume* pEndpointVolume = NULL;
|
|
||||||
LPWSTR pwszID = NULL;
|
|
||||||
|
|
||||||
// Retrieve a count of the devices in the device collection.
|
// Retrieve a count of the devices in the device collection.
|
||||||
hr = pCollection->GetCount(&count);
|
hr = pCollection->GetCount(&count);
|
||||||
|
|||||||
@@ -194,8 +194,8 @@ WebRtc_Word32 DeviceInfoDS::GetDeviceInfo(
|
|||||||
(_wcsnicmp(varName.bstrVal, (L"Google Camera Adapter"),21)
|
(_wcsnicmp(varName.bstrVal, (L"Google Camera Adapter"),21)
|
||||||
!= 0))
|
!= 0))
|
||||||
{
|
{
|
||||||
// Found a valid device
|
// Found a valid device.
|
||||||
if (index == deviceNumber) // This is the device we are interested in.
|
if (index == static_cast<int>(deviceNumber))
|
||||||
{
|
{
|
||||||
int convResult = 0;
|
int convResult = 0;
|
||||||
if (deviceNameLength > 0)
|
if (deviceNameLength > 0)
|
||||||
|
|||||||
Reference in New Issue
Block a user