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Unified Diff: webrtc/modules/audio_device/win/audio_device_core_win.cc

Issue 2985443002: Replace WEBRTC_TRACE logging in modules/audio_device/.. mac/ win/ (Closed)
Patch Set: Manual fixes. Created 3 years, 5 months ago
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Index: webrtc/modules/audio_device/win/audio_device_core_win.cc
diff --git a/webrtc/modules/audio_device/win/audio_device_core_win.cc b/webrtc/modules/audio_device/win/audio_device_core_win.cc
index 2b0a3adfd8ea44ad5f7d111ab96fcb6970da3831..783e9898ded6d817fa0b891b3d6066631dca8fa5 100644
--- a/webrtc/modules/audio_device/win/audio_device_core_win.cc
+++ b/webrtc/modules/audio_device/win/audio_device_core_win.cc
@@ -35,10 +35,11 @@
#include <strsafe.h>
#include <uuids.h>
+#include <iomanip>
+
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/platform_thread.h"
#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/system_wrappers/include/trace.h"
// Macro that calls a COM method returning HRESULT value.
#define EXIT_ON_ERROR(hres) do { if (FAILED(hres)) goto Exit; } while(0)
@@ -181,7 +182,7 @@ private:
bool AudioDeviceWindowsCore::CoreAudioIsSupported()
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
bool MMDeviceIsAvailable(false);
bool coreAudioIsSupported(false);
@@ -223,9 +224,9 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
dwlConditionMask);
if (isVistaRTMorXP != 0)
{
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
- "*** Windows Core Audio is only supported on Vista SP1 or later "
- "=> will revert to the Wave API ***");
+ LOG(LS_VERBOSE)
+ << "*** Windows Core Audio is only supported on Vista SP1 or later"
+ << " => will revert to the Wave API ***";
return false;
}
@@ -276,10 +277,12 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
if (FAILED(hr))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to create the required COM object", hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() CoCreateInstance(MMDeviceEnumerator) failed (hr=0x%x)", hr);
+ LOG(LS_ERROR) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " Failed to create the required COM object (hr="
+ << hr << ")";
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " CoCreateInstance(MMDeviceEnumerator) failed (hr="
+ << hr << ")";
const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
FORMAT_MESSAGE_IGNORE_INSERTS;
@@ -306,13 +309,14 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
StringCchCat(buf, MAXERRORLENGTH, errorText);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%S", buf);
+ LOG(LS_VERBOSE) << buf;
}
else
{
MMDeviceIsAvailable = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() CoCreateInstance(MMDeviceEnumerator) succeeded", hr);
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " CoCreateInstance(MMDeviceEnumerator) succeeded (hr=" << hr
+ << ")";
SAFE_RELEASE(pIMMD);
}
@@ -324,7 +328,7 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
{
coreAudioIsSupported = false;
- AudioDeviceWindowsCore* p = new AudioDeviceWindowsCore(-1);
+ AudioDeviceWindowsCore* p = new AudioDeviceWindowsCore();
if (p == NULL)
{
return false;
@@ -351,8 +355,10 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
}
if (ok)
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Recording for device id=%i", i);
+ LOG(LS_WARNING)
+ << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " Failed to use Core Audio Recording for device id="
+ << i;
}
}
@@ -369,8 +375,9 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
}
if (ok)
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1 ,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Playout for device id=%i", i);
+ LOG(LS_WARNING)
+ << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+ << " Failed to use Core Audio Playout for device id=" << i;
}
}
@@ -386,11 +393,12 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
if (coreAudioIsSupported)
{
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is supported ***");
+ LOG(LS_VERBOSE) << "*** Windows Core Audio is supported ***";
}
else
{
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is NOT supported => will revert to the Wave API ***");
+ LOG(LS_VERBOSE) << "*** Windows Core Audio is NOT supported"
+ << " => will revert to the Wave API ***";
}
return (coreAudioIsSupported);
@@ -404,9 +412,8 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
// AudioDeviceWindowsCore() - ctor
// ----------------------------------------------------------------------------
-AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id)
+AudioDeviceWindowsCore::AudioDeviceWindowsCore()
: _comInit(ScopedCOMInitializer::kMTA),
- _id(id),
_ptrAudioBuffer(NULL),
_ptrEnumerator(NULL),
_ptrRenderCollection(NULL),
@@ -472,7 +479,7 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id)
_inputDeviceIndex(0),
_outputDeviceIndex(0),
_newMicLevel(0) {
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+ LOG(LS_INFO) << __FUNCTION__ << " created";
assert(_comInit.succeeded());
// Try to load the Avrt DLL
@@ -482,9 +489,8 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id)
if (_avrtLibrary) {
// Handle is valid (should only happen if OS larger than vista & win7).
// Try to get the function addresses.
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::AudioDeviceWindowsCore() The Avrt "
- "DLL module is now loaded");
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+ << " The Avrt DLL module is now loaded";
_PAvRevertMmThreadCharacteristics =
(PAvRevertMmThreadCharacteristics)GetProcAddress(
@@ -497,15 +503,12 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id)
if (_PAvRevertMmThreadCharacteristics &&
_PAvSetMmThreadCharacteristicsA && _PAvSetMmThreadPriority) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::AudioDeviceWindowsCore() "
- "AvRevertMmThreadCharacteristics() is OK");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::AudioDeviceWindowsCore() "
- "AvSetMmThreadCharacteristicsA() is OK");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::AudioDeviceWindowsCore() "
- "AvSetMmThreadPriority() is OK");
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+ << " AvRevertMmThreadCharacteristics() is OK";
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+ << " AvSetMmThreadCharacteristicsA() is OK";
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+ << " AvSetMmThreadPriority() is OK";
_winSupportAvrt = true;
}
}
@@ -573,7 +576,7 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id)
AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+ LOG(LS_INFO) << __FUNCTION__ << " destroyed";
Terminate();
@@ -630,13 +633,15 @@ AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
BOOL freeOK = FreeLibrary(_avrtLibrary);
if (!freeOK)
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::~AudioDeviceWindowsCore() failed to free the loaded Avrt DLL module correctly");
+ LOG(LS_WARNING)
+ << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
+ << " failed to free the loaded Avrt DLL module correctly";
}
else
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::~AudioDeviceWindowsCore() the Avrt DLL module is now unloaded");
+ LOG(LS_WARNING)
+ << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
+ << " the Avrt DLL module is now unloaded";
}
}
}
@@ -766,7 +771,8 @@ int32_t AudioDeviceWindowsCore::InitSpeaker()
int16_t nDevices = PlayoutDevices();
if (_outputDeviceIndex > (nDevices - 1))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
+ LOG(LS_ERROR) << "current device selection is invalid => unable to"
+ << " initialize";
return -1;
}
}
@@ -789,7 +795,7 @@ int32_t AudioDeviceWindowsCore::InitSpeaker()
if (ret != 0 || (_ptrDeviceOut == NULL))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the rendering enpoint device");
+ LOG(LS_ERROR) << "failed to initialize the rendering enpoint device";
SAFE_RELEASE(_ptrDeviceOut);
return -1;
}
@@ -801,8 +807,7 @@ int32_t AudioDeviceWindowsCore::InitSpeaker()
(void**)&pManager);
if (ret != 0 || pManager == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize the render manager");
+ LOG(LS_ERROR) << "failed to initialize the render manager";
SAFE_RELEASE(pManager);
return -1;
}
@@ -811,8 +816,7 @@ int32_t AudioDeviceWindowsCore::InitSpeaker()
ret = pManager->GetSimpleAudioVolume(NULL, FALSE, &_ptrRenderSimpleVolume);
if (ret != 0 || _ptrRenderSimpleVolume == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize the render simple volume");
+ LOG(LS_ERROR) << "failed to initialize the render simple volume";
SAFE_RELEASE(pManager);
SAFE_RELEASE(_ptrRenderSimpleVolume);
return -1;
@@ -848,7 +852,8 @@ int32_t AudioDeviceWindowsCore::InitMicrophone()
int16_t nDevices = RecordingDevices();
if (_inputDeviceIndex > (nDevices - 1))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
+ LOG(LS_ERROR) << "current device selection is invalid => unable to"
+ << " initialize";
return -1;
}
}
@@ -871,7 +876,7 @@ int32_t AudioDeviceWindowsCore::InitMicrophone()
if (ret != 0 || (_ptrDeviceIn == NULL))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the capturing enpoint device");
+ LOG(LS_ERROR) << "failed to initialize the capturing enpoint device";
SAFE_RELEASE(_ptrDeviceIn);
return -1;
}
@@ -883,8 +888,7 @@ int32_t AudioDeviceWindowsCore::InitMicrophone()
reinterpret_cast<void **>(&_ptrCaptureVolume));
if (ret != 0 || _ptrCaptureVolume == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize the capture volume");
+ LOG(LS_ERROR) << "failed to initialize the capture volume";
SAFE_RELEASE(_ptrCaptureVolume);
return -1;
}
@@ -1548,7 +1552,8 @@ Exit:
int32_t AudioDeviceWindowsCore::SetMicrophoneVolume(uint32_t volume)
{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::SetMicrophoneVolume(volume=%u)", volume);
+ LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::SetMicrophoneVolume(volume="
+ << volume << ")";
{
rtc::CritScope lock(&_critSect);
@@ -1634,7 +1639,7 @@ Exit:
int32_t AudioDeviceWindowsCore::MaxMicrophoneVolume(uint32_t& maxVolume) const
{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
if (!_microphoneIsInitialized)
{
@@ -1714,7 +1719,8 @@ int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index)
if (index < 0 || index > (nDevices-1))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
+ LOG(LS_ERROR) << "device index is out of range [0," << (nDevices-1)
+ << "]";
return -1;
}
@@ -1742,7 +1748,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index)
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
}
_usingOutputDeviceIndex = true;
@@ -1801,7 +1807,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutDevice(AudioDeviceModule::WindowsDevic
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
}
_usingOutputDeviceIndex = false;
@@ -1828,7 +1834,7 @@ int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
{
defaultCommunicationDevice = true;
index = 0;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
+ LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
}
if ((index > (nDevices-1)) || (name == NULL))
@@ -1864,7 +1870,9 @@ int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
// Convert the endpoint device's friendly-name to UTF-8
if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ LOG(LS_ERROR)
+ << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
}
@@ -1883,7 +1891,9 @@ int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
// Convert the endpoint device's ID string to UTF-8
if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ LOG(LS_ERROR)
+ << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
}
@@ -1908,7 +1918,7 @@ int32_t AudioDeviceWindowsCore::RecordingDeviceName(
{
defaultCommunicationDevice = true;
index = 0;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
+ LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
}
if ((index > (nDevices-1)) || (name == NULL))
@@ -1944,7 +1954,9 @@ int32_t AudioDeviceWindowsCore::RecordingDeviceName(
// Convert the endpoint device's friendly-name to UTF-8
if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ LOG(LS_ERROR)
+ << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
}
@@ -1963,7 +1975,9 @@ int32_t AudioDeviceWindowsCore::RecordingDeviceName(
// Convert the endpoint device's ID string to UTF-8
if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
+ LOG(LS_ERROR)
+ << "WideCharToMultiByte(CP_UTF8) failed with error code "
+ << GetLastError();
}
}
@@ -2004,7 +2018,8 @@ int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index)
if (index < 0 || index > (nDevices-1))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
+ LOG(LS_ERROR) << "device index is out of range [0," << (nDevices-1)
+ << "]";
return -1;
}
@@ -2032,7 +2047,7 @@ int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index)
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
}
_usingInputDeviceIndex = true;
@@ -2091,7 +2106,7 @@ int32_t AudioDeviceWindowsCore::SetRecordingDevice(AudioDeviceModule::WindowsDev
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
+ LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
}
_usingInputDeviceIndex = false;
@@ -2173,7 +2188,7 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
// Initialize the speaker (devices might have been added or removed)
if (InitSpeaker() == -1)
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
+ LOG(LS_WARNING) << "InitSpeaker() failed";
}
// Ensure that the updated rendering endpoint device is valid
@@ -2211,20 +2226,22 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
hr = _ptrClientOut->GetMixFormat(&pWfxOut);
if (SUCCEEDED(hr))
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current rendering mix format:");
+ LOG(LS_VERBOSE) << "Audio Engine's current rendering mix format:";
// format type
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxOut->wFormatTag, pWfxOut->wFormatTag);
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
+ << pWfxOut->wFormatTag << std::dec << " ("
+ << pWfxOut->wFormatTag << ")";
// number of channels (i.e. mono, stereo...)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxOut->nChannels);
+ LOG(LS_VERBOSE) << "nChannels : " << pWfxOut->nChannels;
// sample rate
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxOut->nSamplesPerSec);
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxOut->nSamplesPerSec;
// for buffer estimation
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxOut->nAvgBytesPerSec);
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxOut->nAvgBytesPerSec;
// block size of data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", pWfxOut->nBlockAlign);
+ LOG(LS_VERBOSE) << "nBlockAlign : " << pWfxOut->nBlockAlign;
// number of bits per sample of mono data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", pWfxOut->wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", pWfxOut->cbSize);
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxOut->wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << pWfxOut->cbSize;
}
// Set wave format
@@ -2295,19 +2312,21 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
_devicePlayBlockSize = Wfx.nSamplesPerSec/100;
_playChannels = Wfx.nChannels;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "VoE selected this rendering format:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", Wfx.wFormatTag, Wfx.wFormatTag);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", Wfx.nChannels);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", Wfx.nSamplesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec : %d", Wfx.nAvgBytesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", Wfx.nBlockAlign);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", Wfx.wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", Wfx.cbSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Additional settings:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playAudioFrameSize: %d", _playAudioFrameSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "_playBlockSizeInFrames : %d", _playBlockSizeInFrames);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playChannels : %d", _playChannels);
+ LOG(LS_VERBOSE) << "VoE selected this rendering format:";
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
+ << Wfx.wFormatTag << std::dec << " (" << Wfx.wFormatTag
+ << ")";
+ LOG(LS_VERBOSE) << "nChannels : " << Wfx.nChannels;
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << Wfx.nSamplesPerSec;
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec : " << Wfx.nAvgBytesPerSec;
+ LOG(LS_VERBOSE) << "nBlockAlign : " << Wfx.nBlockAlign;
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << Wfx.wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << Wfx.cbSize;
+ LOG(LS_VERBOSE) << "Additional settings:";
+ LOG(LS_VERBOSE) << "_playAudioFrameSize: " << _playAudioFrameSize;
+ LOG(LS_VERBOSE) << "_playBlockSizeInFrames : "
+ << _playBlockSizeInFrames;
+ LOG(LS_VERBOSE) << "_playChannels : " << _playChannels;
}
// Create a rendering stream.
@@ -2347,7 +2366,7 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
if (FAILED(hr))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
+ LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
}
EXIT_ON_ERROR(hr);
@@ -2362,7 +2381,8 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
// We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
// has been created, hence the AudioDeviceBuffer does not exist.
// It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceBuffer must be attached before streaming can start");
+ LOG(LS_VERBOSE)
+ << "AudioDeviceBuffer must be attached before streaming can start";
}
// Get the actual size of the shared (endpoint buffer).
@@ -2372,8 +2392,9 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
&bufferFrameCount);
if (SUCCEEDED(hr))
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
- bufferFrameCount, bufferFrameCount*_playAudioFrameSize);
+ LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => "
+ << bufferFrameCount << " (<=> "
+ << bufferFrameCount*_playAudioFrameSize << " bytes)";
}
// Set the event handle that the system signals when an audio buffer is ready
@@ -2395,7 +2416,7 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
CoTaskMemFree(pWfxOut);
CoTaskMemFree(pWfxClosestMatch);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render side is now initialized");
+ LOG(LS_VERBOSE) << "render side is now initialized";
return 0;
Exit:
@@ -2470,8 +2491,8 @@ int32_t AudioDeviceWindowsCore::InitRecordingDMO()
else
{
// Refer to InitRecording() for comments.
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioDeviceBuffer must be attached before streaming can start");
+ LOG(LS_VERBOSE)
+ << "AudioDeviceBuffer must be attached before streaming can start";
}
_mediaBuffer = new MediaBufferImpl(_recBlockSize * _recAudioFrameSize);
@@ -2485,8 +2506,7 @@ int32_t AudioDeviceWindowsCore::InitRecordingDMO()
}
_recIsInitialized = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Capture side is now initialized");
+ LOG(LS_VERBOSE) << "Capture side is now initialized";
return 0;
}
@@ -2524,7 +2544,7 @@ int32_t AudioDeviceWindowsCore::InitRecording()
// Initialize the microphone (devices might have been added or removed)
if (InitMicrophone() == -1)
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
+ LOG(LS_WARNING) << "InitMicrophone() failed";
}
// Ensure that the updated capturing endpoint device is valid
@@ -2558,20 +2578,22 @@ int32_t AudioDeviceWindowsCore::InitRecording()
hr = _ptrClientIn->GetMixFormat(&pWfxIn);
if (SUCCEEDED(hr))
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current capturing mix format:");
+ LOG(LS_VERBOSE) << "Audio Engine's current capturing mix format:";
// format type
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxIn->wFormatTag, pWfxIn->wFormatTag);
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
+ << pWfxIn->wFormatTag << std::dec << " ("
+ << pWfxIn->wFormatTag << ")";
// number of channels (i.e. mono, stereo...)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxIn->nChannels);
+ LOG(LS_VERBOSE) << "nChannels : " << pWfxIn->nChannels;
// sample rate
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxIn->nSamplesPerSec);
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxIn->nSamplesPerSec;
// for buffer estimation
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxIn->nAvgBytesPerSec);
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxIn->nAvgBytesPerSec;
// block size of data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", pWfxIn->nBlockAlign);
+ LOG(LS_VERBOSE) << "nBlockAlign : " << pWfxIn->nBlockAlign;
// number of bits per sample of mono data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", pWfxIn->wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", pWfxIn->cbSize);
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxIn->wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << pWfxIn->cbSize;
}
// Set wave format
@@ -2638,19 +2660,20 @@ int32_t AudioDeviceWindowsCore::InitRecording()
_recBlockSize = Wfx.Format.nSamplesPerSec/100;
_recChannels = Wfx.Format.nChannels;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "VoE selected this capturing format:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", Wfx.Format.wFormatTag,
- Wfx.Format.wFormatTag);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", Wfx.Format.nChannels);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", Wfx.Format.nSamplesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec : %d", Wfx.Format.nAvgBytesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", Wfx.Format.nBlockAlign);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", Wfx.Format.wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", Wfx.Format.cbSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Additional settings:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recAudioFrameSize: %d", _recAudioFrameSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recBlockSize : %d", _recBlockSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recChannels : %d", _recChannels);
+ LOG(LS_VERBOSE) << "VoE selected this capturing format:";
+ LOG(LS_VERBOSE) << "wFormatTag : 0x" << std::hex
+ << Wfx.Format.wFormatTag << std::dec
+ << " (" << Wfx.Format.wFormatTag << ")";
+ LOG(LS_VERBOSE) << "nChannels : " << Wfx.Format.nChannels;
+ LOG(LS_VERBOSE) << "nSamplesPerSec : " << Wfx.Format.nSamplesPerSec;
+ LOG(LS_VERBOSE) << "nAvgBytesPerSec : " << Wfx.Format.nAvgBytesPerSec;
+ LOG(LS_VERBOSE) << "nBlockAlign : " << Wfx.Format.nBlockAlign;
+ LOG(LS_VERBOSE) << "wBitsPerSample : " << Wfx.Format.wBitsPerSample;
+ LOG(LS_VERBOSE) << "cbSize : " << Wfx.Format.cbSize;
+ LOG(LS_VERBOSE) << "Additional settings:";
+ LOG(LS_VERBOSE) << "_recAudioFrameSize: " << _recAudioFrameSize;
+ LOG(LS_VERBOSE) << "_recBlockSize : " << _recBlockSize;
+ LOG(LS_VERBOSE) << "_recChannels : " << _recChannels;
}
// Create a capturing stream.
@@ -2666,7 +2689,7 @@ int32_t AudioDeviceWindowsCore::InitRecording()
if (hr != S_OK)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
+ LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
}
EXIT_ON_ERROR(hr);
@@ -2681,7 +2704,8 @@ int32_t AudioDeviceWindowsCore::InitRecording()
// We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
// has been created, hence the AudioDeviceBuffer does not exist.
// It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceBuffer must be attached before streaming can start");
+ LOG(LS_VERBOSE)
+ << "AudioDeviceBuffer must be attached before streaming can start";
}
// Get the actual size of the shared (endpoint buffer).
@@ -2691,8 +2715,9 @@ int32_t AudioDeviceWindowsCore::InitRecording()
&bufferFrameCount);
if (SUCCEEDED(hr))
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
- bufferFrameCount, bufferFrameCount*_recAudioFrameSize);
+ LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => "
+ << bufferFrameCount << " (<=> "
+ << bufferFrameCount*_recAudioFrameSize << " bytes)";
}
// Set the event handle that the system signals when an audio buffer is ready
@@ -2714,7 +2739,7 @@ int32_t AudioDeviceWindowsCore::InitRecording()
CoTaskMemFree(pWfxIn);
CoTaskMemFree(pWfxClosestMatch);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "capture side is now initialized");
+ LOG(LS_VERBOSE) << "capture side is now initialized";
return 0;
Exit:
@@ -2762,9 +2787,9 @@ int32_t AudioDeviceWindowsCore::StartRecording()
{
// The DMO won't provide us captured output data unless we
// give it render data to process.
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Playout must be started before recording when using the "
- "built-in AEC");
+ LOG(LS_ERROR)
+ << "Playout must be started before recording when using"
+ << " the built-in AEC";
return -1;
}
}
@@ -2778,8 +2803,7 @@ int32_t AudioDeviceWindowsCore::StartRecording()
NULL);
if (_hRecThread == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to create the recording thread");
+ LOG(LS_ERROR) << "failed to create the recording thread";
return -1;
}
@@ -2795,8 +2819,7 @@ int32_t AudioDeviceWindowsCore::StartRecording()
NULL);
if (_hGetCaptureVolumeThread == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume getter thread");
+ LOG(LS_ERROR) << "failed to create the volume getter thread";
return -1;
}
@@ -2809,8 +2832,7 @@ int32_t AudioDeviceWindowsCore::StartRecording()
NULL);
if (_hSetCaptureVolumeThread == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume setter thread");
+ LOG(LS_ERROR) << "failed to create the volume setter thread";
return -1;
}
} // critScoped
@@ -2818,12 +2840,10 @@ int32_t AudioDeviceWindowsCore::StartRecording()
DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
if (ret != WAIT_OBJECT_0)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "capturing did not start up properly");
+ LOG(LS_VERBOSE) << "capturing did not start up properly";
return -1;
}
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "capture audio stream has now started...");
+ LOG(LS_VERBOSE) << "capture audio stream has now started...";
_avgCPULoad = 0.0f;
_playAcc = 0;
@@ -2849,8 +2869,8 @@ int32_t AudioDeviceWindowsCore::StopRecording()
if (_hRecThread == NULL)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "no capturing stream is active => close down WASAPI only");
+ LOG(LS_VERBOSE)
+ << "no capturing stream is active => close down WASAPI only";
SAFE_RELEASE(_ptrClientIn);
SAFE_RELEASE(_ptrCaptureClient);
_recIsInitialized = false;
@@ -2860,8 +2880,7 @@ int32_t AudioDeviceWindowsCore::StopRecording()
}
// Stop the driving thread...
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "closing down the webrtc_core_audio_capture_thread...");
+ LOG(LS_VERBOSE) << "closing down the webrtc_core_audio_capture_thread...";
// Manual-reset event; it will remain signalled to stop all capture threads.
SetEvent(_hShutdownCaptureEvent);
@@ -2869,42 +2888,37 @@ int32_t AudioDeviceWindowsCore::StopRecording()
DWORD ret = WaitForSingleObject(_hRecThread, 2000);
if (ret != WAIT_OBJECT_0)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to close down webrtc_core_audio_capture_thread");
+ LOG(LS_ERROR)
+ << "failed to close down webrtc_core_audio_capture_thread";
err = -1;
}
else
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "webrtc_core_audio_capture_thread is now closed");
+ LOG(LS_VERBOSE) << "webrtc_core_audio_capture_thread is now closed";
}
ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
if (ret != WAIT_OBJECT_0)
{
// the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume getter thread");
+ LOG(LS_ERROR) << "failed to close down volume getter thread";
err = -1;
}
else
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume getter thread is now closed");
+ LOG(LS_VERBOSE) << "volume getter thread is now closed";
}
ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
if (ret != WAIT_OBJECT_0)
{
// the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume setter thread");
+ LOG(LS_ERROR) << "failed to close down volume setter thread";
err = -1;
}
else
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume setter thread is now closed");
+ LOG(LS_VERBOSE) << "volume setter thread is now closed";
}
_Lock();
@@ -3012,8 +3026,7 @@ int32_t AudioDeviceWindowsCore::StartPlayout()
NULL);
if (_hPlayThread == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to create the playout thread");
+ LOG(LS_ERROR) << "failed to create the playout thread";
return -1;
}
@@ -3024,14 +3037,12 @@ int32_t AudioDeviceWindowsCore::StartPlayout()
DWORD ret = WaitForSingleObject(_hRenderStartedEvent, 1000);
if (ret != WAIT_OBJECT_0)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "rendering did not start up properly");
+ LOG(LS_VERBOSE) << "rendering did not start up properly";
return -1;
}
_playing = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "rendering audio stream has now started...");
+ LOG(LS_VERBOSE) << "rendering audio stream has now started...";
return 0;
}
@@ -3053,8 +3064,8 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
if (_hPlayThread == NULL)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "no rendering stream is active => close down WASAPI only");
+ LOG(LS_VERBOSE)
+ << "no rendering stream is active => close down WASAPI only";
SAFE_RELEASE(_ptrClientOut);
SAFE_RELEASE(_ptrRenderClient);
_playIsInitialized = false;
@@ -3063,8 +3074,8 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
}
// stop the driving thread...
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "closing down the webrtc_core_audio_render_thread...");
+ LOG(LS_VERBOSE)
+ << "closing down the webrtc_core_audio_render_thread...";
SetEvent(_hShutdownRenderEvent);
} // critScoped
@@ -3072,8 +3083,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
if (ret != WAIT_OBJECT_0)
{
// the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to close down webrtc_core_audio_render_thread");
+ LOG(LS_ERROR) << "failed to close down webrtc_core_audio_render_thread";
CloseHandle(_hPlayThread);
_hPlayThread = NULL;
_playIsInitialized = false;
@@ -3083,8 +3093,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
{
rtc::CritScope critScoped(&_critSect);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "webrtc_core_audio_render_thread is now closed");
+ LOG(LS_VERBOSE) << "webrtc_core_audio_render_thread is now closed";
// to reset this event manually at each time we finish with it,
// in case that the render thread has exited before StopPlayout(),
@@ -3108,9 +3117,9 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
// We still permit the playout to shutdown, and trace a warning.
// Otherwise, VoE can get into a state which will never permit
// playout to stop properly.
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "Recording should be stopped before playout when using the "
- "built-in AEC");
+ LOG(LS_WARNING)
+ << "Recording should be stopped before playout when using the"
+ << " built-in AEC";
}
// Reset the playout delay value.
@@ -3347,8 +3356,8 @@ DWORD AudioDeviceWindowsCore::DoGetCaptureVolumeThread()
case WAIT_TIMEOUT: // timeout notification
break;
default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on get volume thread");
+ LOG(LS_WARNING)
+ << "unknown wait termination on get volume thread";
return 1;
}
}
@@ -3368,8 +3377,8 @@ DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread()
case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
break;
default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on set volume thread");
+ LOG(LS_WARNING)
+ << "unknown wait termination on set volume thread";
return 1;
}
@@ -3379,8 +3388,8 @@ DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread()
if (SetMicrophoneVolume(newMicLevel) == -1)
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " the required modification of the microphone volume failed");
+ LOG(LS_WARNING)
+ << "the required modification of the microphone volume failed";
}
}
}
@@ -3404,8 +3413,7 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
// Initialize COM as MTA in this thread.
ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
if (!comInit.succeeded()) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to initialize COM in render thread");
+ LOG(LS_ERROR) << "failed to initialize COM in render thread";
return 1;
}
@@ -3421,13 +3429,16 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
{
if (FALSE == _PAvSetMmThreadPriority(hMmTask, AVRT_PRIORITY_CRITICAL))
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to boost play-thread using MMCSS");
+ LOG(LS_WARNING) << "failed to boost play-thread using MMCSS";
}
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render thread is now registered with MMCSS (taskIndex=%d)", taskIndex);
+ LOG(LS_VERBOSE)
+ << "render thread is now registered with MMCSS (taskIndex="
+ << taskIndex << ")";
}
else
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to enable MMCSS on render thread (err=%d)", GetLastError());
+ LOG(LS_WARNING) << "failed to enable MMCSS on render thread (err="
+ << GetLastError() << ")";
_TraceCOMError(GetLastError());
}
}
@@ -3442,14 +3453,14 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
UINT32 bufferLength = 0;
hr = _ptrClientOut->GetBufferSize(&bufferLength);
EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] size of buffer : %u", bufferLength);
+ LOG(LS_VERBOSE) << "[REND] size of buffer : " << bufferLength;
// Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
//
REFERENCE_TIME latency;
_ptrClientOut->GetStreamLatency(&latency);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] max stream latency : %u (%3.2f ms)",
- (DWORD)latency, (double)(latency/10000.0));
+ LOG(LS_VERBOSE) << "[REND] max stream latency : " << (DWORD)latency
+ << " (" << (double)(latency/10000.0) << " ms)";
// Get the length of the periodic interval separating successive processing passes by
// the audio engine on the data in the endpoint buffer.
@@ -3463,8 +3474,8 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
REFERENCE_TIME devPeriod = 0;
REFERENCE_TIME devPeriodMin = 0;
_ptrClientOut->GetDevicePeriod(&devPeriod, &devPeriodMin);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] device period : %u (%3.2f ms)",
- (DWORD)devPeriod, (double)(devPeriod/10000.0));
+ LOG(LS_VERBOSE) << "[REND] device period : " << (DWORD)devPeriod
+ << " (" << (double)(devPeriod/10000.0) << " ms)";
// Derive initial rendering delay.
// Example: 10*(960/480) + 15 = 20 + 15 = 35ms
@@ -3473,11 +3484,10 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
(int)((latency + devPeriod) / 10000);
_sndCardPlayDelay = playout_delay;
_writtenSamples = 0;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "[REND] initial delay : %u", playout_delay);
+ LOG(LS_VERBOSE) << "[REND] initial delay : " << playout_delay;
double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_devicePlayBlockSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
+ LOG(LS_VERBOSE) << "[REND] endpointBufferSizeMS : " << endpointBufferSizeMS;
// Before starting the stream, fill the rendering buffer with silence.
//
@@ -3492,8 +3502,8 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
if (FAILED(hr)) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "failed to get IAudioClock interface from the IAudioClient");
+ LOG(LS_WARNING)
+ << "failed to get IAudioClock interface from the IAudioClient";
}
// Start up the rendering audio stream.
@@ -3520,10 +3530,10 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
case WAIT_OBJECT_0 + 1: // _hRenderSamplesReadyEvent
break;
case WAIT_TIMEOUT: // timeout notification
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "render event timed out after 0.5 seconds");
+ LOG(LS_WARNING) << "render event timed out after 0.5 seconds";
goto Exit;
default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on render side");
+ LOG(LS_WARNING) << "unknown wait termination on render side";
goto Exit;
}
@@ -3536,8 +3546,8 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
{
_UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "output state has been modified during unlocked period");
+ LOG(LS_ERROR)
+ << "output state has been modified during unlocked period";
goto Exit;
}
@@ -3548,7 +3558,6 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
// Derive the amount of available space in the output buffer
uint32_t framesAvailable = bufferLength - padding;
- // WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "#avaliable audio frames = %u", framesAvailable);
// Do we have 10 ms available in the render buffer?
if (framesAvailable < _playBlockSizeInFrames) {
@@ -3580,8 +3589,7 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
if (nSamples == -1) {
_UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "failed to read data from render client");
+ LOG(LS_ERROR) << "failed to read data from render client";
goto Exit;
}
@@ -3589,15 +3597,17 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
{
_UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
+ LOG(LS_ERROR)
+ << "output state has been modified during unlocked"
+ << " period";
goto Exit;
}
if (nSamples !=
static_cast<int32_t>(_playBlockSizeInSamples)) {
- WEBRTC_TRACE(
- kTraceWarning, kTraceAudioDevice, _id,
- "nSamples(%d) != _playBlockSizeInSamples(%d)",
- nSamples, _playBlockSizeInSamples);
+ LOG(LS_WARNING)
+ << "nSamples(" << nSamples
+ << ") != _playBlockSizeInSamples("
+ << _playBlockSizeInSamples << ")";
}
// Get the actual (stored) data
@@ -3675,11 +3685,13 @@ Exit:
}
// Trigger callback from module process thread
_playError = 1;
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: rendering thread has ended pre-maturely");
+ LOG(LS_ERROR)
+ << "kPlayoutError message posted: rendering thread has ended"
+ << " pre-maturely";
}
else
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Rendering thread is now terminated properly");
+ LOG(LS_VERBOSE) << "_Rendering thread is now terminated properly";
}
_UnLock();
@@ -3703,18 +3715,16 @@ DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority()
{
if (!_PAvSetMmThreadPriority(_hMmTask, AVRT_PRIORITY_CRITICAL))
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "failed to boost rec-thread using MMCSS");
+ LOG(LS_WARNING) << "failed to boost rec-thread using MMCSS";
}
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "capture thread is now registered with MMCSS (taskIndex=%d)",
- taskIndex);
+ LOG(LS_VERBOSE)
+ << "capture thread is now registered with MMCSS (taskIndex="
+ << taskIndex << ")";
}
else
{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "failed to enable MMCSS on capture thread (err=%d)",
- GetLastError());
+ LOG(LS_WARNING) << "failed to enable MMCSS on capture thread (err="
+ << GetLastError() << ")";
_TraceCOMError(GetLastError());
}
}
@@ -3743,8 +3753,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
// Initialize COM as MTA in this thread.
ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
if (!comInit.succeeded()) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to initialize COM in polling DMO thread");
+ LOG(LS_ERROR) << "failed to initialize COM in polling DMO thread";
return 1;
}
@@ -3772,8 +3781,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
case WAIT_TIMEOUT: // timeout notification
break;
default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "Unknown wait termination on capture side");
+ LOG(LS_WARNING) << "Unknown wait termination on capture side";
hr = -1; // To signal an error callback.
keepRecording = false;
break;
@@ -3864,14 +3872,12 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
{
// Trigger callback from module process thread
_recError = 1;
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
- "kRecordingError message posted: capturing thread has ended "
- "prematurely");
+ LOG(LS_ERROR) << "kRecordingError message posted: capturing thread has"
+ << " ended prematurely";
}
else
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Capturing thread is now terminated properly");
+ LOG(LS_VERBOSE) << "Capturing thread is now terminated properly";
}
return hr;
@@ -3901,8 +3907,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
// Initialize COM as MTA in this thread.
ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
if (!comInit.succeeded()) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to initialize COM in capture thread");
+ LOG(LS_ERROR) << "failed to initialize COM in capture thread";
return 1;
}
@@ -3920,13 +3925,13 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
UINT32 bufferLength = 0;
if (_ptrClientIn == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "input state has been modified before capture loop starts.");
+ LOG(LS_ERROR)
+ << "input state has been modified before capture loop starts.";
return 1;
}
hr = _ptrClientIn->GetBufferSize(&bufferLength);
EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] size of buffer : %u", bufferLength);
+ LOG(LS_VERBOSE) << "[CAPT] size of buffer : " << bufferLength;
// Allocate memory for sync buffer.
// It is used for compensation between native 44.1 and internal 44.0 and
@@ -3938,14 +3943,15 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
{
return (DWORD)E_POINTER;
}
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] size of sync buffer : %u [bytes]", syncBufferSize);
+ LOG(LS_VERBOSE) << "[CAPT] size of sync buffer : " << syncBufferSize
+ << " [bytes]";
// Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
//
REFERENCE_TIME latency;
_ptrClientIn->GetStreamLatency(&latency);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] max stream latency : %u (%3.2f ms)",
- (DWORD)latency, (double)(latency / 10000.0));
+ LOG(LS_VERBOSE) << "[CAPT] max stream latency : " << (DWORD)latency
+ << " (" << (double)(latency / 10000.0) << " ms)";
// Get the length of the periodic interval separating successive processing passes by
// the audio engine on the data in the endpoint buffer.
@@ -3953,14 +3959,14 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
REFERENCE_TIME devPeriod = 0;
REFERENCE_TIME devPeriodMin = 0;
_ptrClientIn->GetDevicePeriod(&devPeriod, &devPeriodMin);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] device period : %u (%3.2f ms)",
- (DWORD)devPeriod, (double)(devPeriod / 10000.0));
+ LOG(LS_VERBOSE) << "[CAPT] device period : " << (DWORD)devPeriod
+ << " (" << (double)(devPeriod / 10000.0) << " ms)";
double extraDelayMS = (double)((latency + devPeriod) / 10000.0);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] extraDelayMS : %3.2f", extraDelayMS);
+ LOG(LS_VERBOSE) << "[CAPT] extraDelayMS : " << extraDelayMS;
double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_recBlockSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
+ LOG(LS_VERBOSE) << "[CAPT] endpointBufferSizeMS : " << endpointBufferSizeMS;
// Start up the capturing stream.
//
@@ -3987,10 +3993,10 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
case WAIT_OBJECT_0 + 1: // _hCaptureSamplesReadyEvent
break;
case WAIT_TIMEOUT: // timeout notification
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "capture event timed out after 0.5 seconds");
+ LOG(LS_WARNING) << "capture event timed out after 0.5 seconds";
goto Exit;
default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on capture side");
+ LOG(LS_WARNING) << "unknown wait termination on capture side";
goto Exit;
}
@@ -4009,8 +4015,8 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
{
_UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "input state has been modified during unlocked period");
+ LOG(LS_ERROR)
+ << "input state has been modified during unlocked period";
goto Exit;
}
@@ -4034,7 +4040,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
{
// Treat all of the data in the packet as silence and ignore the actual data values.
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "AUDCLNT_BUFFERFLAGS_SILENT");
+ LOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_SILENT";
pData = NULL;
}
@@ -4100,7 +4106,9 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
{
_UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
+ LOG(LS_ERROR)
+ << "input state has been modified during"
+ << " unlocked period";
goto Exit;
}
}
@@ -4118,7 +4126,8 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
{
// The VQE will only deliver non-zero microphone levels when a change is needed.
// Set this new mic level (received from the observer as return value in the callback).
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AGC change of volume: new=%u", newMicLevel);
+ LOG(LS_VERBOSE) << "AGC change of volume: new="
+ << newMicLevel;
// We store this outside of the audio buffer to avoid
// having it overwritten by the getter thread.
_newMicLevel = newMicLevel;
@@ -4133,8 +4142,9 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
// of the failed GetBuffer calls. If GetBuffer returns this error repeatedly, the client
// can start a new processing loop after shutting down the current client by calling
// IAudioClient::Stop, IAudioClient::Reset, and releasing the audio client.
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IAudioCaptureClient::GetBuffer returned AUDCLNT_E_BUFFER_ERROR, hr = 0x%08X", hr);
+ LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer returned"
+ << " AUDCLNT_E_BUFFER_ERROR, hr = 0x"
+ << std::hex << hr << std::dec;
goto Exit;
}
@@ -4179,11 +4189,13 @@ Exit:
// Trigger callback from module process thread
_recError = 1;
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: capturing thread has ended pre-maturely");
+ LOG(LS_ERROR)
+ << "kRecordingError message posted: capturing thread has ended"
+ << " pre-maturely";
}
else
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Capturing thread is now terminated properly");
+ LOG(LS_VERBOSE) << "_Capturing thread is now terminated properly";
}
SAFE_RELEASE(_ptrClientIn);
@@ -4204,15 +4216,15 @@ int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable)
if (_recIsInitialized)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Attempt to set Windows AEC with recording already initialized");
+ LOG(LS_ERROR)
+ << "Attempt to set Windows AEC with recording already initialized";
return -1;
}
if (_dmo == NULL)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Built-in AEC DMO was not initialized properly at create time");
+ LOG(LS_ERROR)
+ << "Built-in AEC DMO was not initialized properly at create time";
return -1;
}
@@ -4328,9 +4340,8 @@ int AudioDeviceWindowsCore::SetDMOProperties()
DWORD devIndex = static_cast<uint32_t>(outDevIndex << 16) +
static_cast<uint32_t>(0x0000ffff & inDevIndex);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Capture device index: %d, render device index: %d",
- inDevIndex, outDevIndex);
+ LOG(LS_VERBOSE) << "Capture device index: " << inDevIndex
+ << ", render device index: " << outDevIndex;
if (SetVtI4Property(ps,
MFPKEY_WMAAECMA_DEVICE_INDEXES,
devIndex) == -1)
@@ -4387,7 +4398,7 @@ int AudioDeviceWindowsCore::SetVtI4Property(IPropertyStore* ptrPS,
int32_t AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr = S_OK;
IMMDeviceCollection *pCollection = NULL;
@@ -4430,7 +4441,7 @@ int32_t AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir)
int16_t AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr = S_OK;
UINT count = 0;
@@ -4468,7 +4479,7 @@ int16_t AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir)
int32_t AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
@@ -4507,7 +4518,7 @@ int32_t AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir, int index, LPW
int32_t AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
@@ -4546,7 +4557,7 @@ int32_t AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir, ERole role,
int32_t AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
@@ -4585,7 +4596,7 @@ int32_t AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir, int index, LPWST
int32_t AudioDeviceWindowsCore::_GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
@@ -4615,7 +4626,7 @@ int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
ERole role,
int* index)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr = S_OK;
WCHAR szDefaultDeviceID[MAX_PATH] = {0};
@@ -4641,8 +4652,7 @@ int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
if (!collection)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Device collection not valid");
+ LOG(LS_ERROR) << "Device collection not valid";
return -1;
}
@@ -4687,8 +4697,7 @@ int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
if (*index == -1)
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Unable to find collection index for default device");
+ LOG(LS_ERROR) << "Unable to find collection index for default device";
return -1;
}
@@ -4703,7 +4712,7 @@ int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
LPWSTR pszBuffer,
int bufferLen)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
static const WCHAR szDefault[] = L"<Device not available>";
@@ -4719,8 +4728,8 @@ int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
if (FAILED(hr))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IMMDevice::OpenPropertyStore failed, hr = 0x%08X", hr);
+ LOG(LS_ERROR) << "IMMDevice::OpenPropertyStore failed, hr = 0x"
+ << std::hex << hr << std::dec;
}
}
@@ -4733,24 +4742,24 @@ int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
if (FAILED(hr))
{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IPropertyStore::GetValue failed, hr = 0x%08X", hr);
+ LOG(LS_ERROR) << "IPropertyStore::GetValue failed, hr = 0x"
+ << std::hex << hr << std::dec;
}
}
if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt))
{
hr = E_FAIL;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IPropertyStore::GetValue returned no value, hr = 0x%08X", hr);
+ LOG(LS_ERROR) << "IPropertyStore::GetValue returned no value,"
+ << " hr = 0x" << std::hex << hr << std::dec;
}
if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt))
{
// The returned value is not a wide null terminated string.
hr = E_UNEXPECTED;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IPropertyStore::GetValue returned unexpected type, hr = 0x%08X", hr);
+ LOG(LS_ERROR) << "IPropertyStore::GetValue returned unexpected"
+ << " type, hr = 0x" << std::hex << hr << std::dec;
}
if (SUCCEEDED(hr) && (varName.pwszVal != NULL))
@@ -4776,7 +4785,7 @@ int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
int32_t AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
static const WCHAR szDefault[] = L"<Device not available>";
@@ -4812,7 +4821,7 @@ int32_t AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffe
int32_t AudioDeviceWindowsCore::_GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
HRESULT hr(S_OK);
@@ -4873,7 +4882,7 @@ int32_t AudioDeviceWindowsCore::_GetListDevice(EDataFlow dir, int index, IMMDevi
int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow) const
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+ LOG(LS_VERBOSE) << __FUNCTION__;
assert(_ptrEnumerator != NULL);
@@ -4902,9 +4911,11 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
hr = pCollection->GetCount(&count);
EXIT_ON_ERROR(hr);
if (dataFlow == eRender)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#rendering endpoint devices (counting all): %u", count);
+ LOG(LS_VERBOSE) << "#rendering endpoint devices (counting all): "
+ << count;
else if (dataFlow == eCapture)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#capturing endpoint devices (counting all): %u", count);
+ LOG(LS_VERBOSE) << "#capturing endpoint devices (counting all): "
+ << count;
if (count == 0)
{
@@ -4914,7 +4925,7 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
// Each loop prints the name of an endpoint device.
for (ULONG i = 0; i < count; i++)
{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Endpoint %d:", i);
+ LOG(LS_VERBOSE) << "Endpoint " << i << ":";
// Get pointer to endpoint number i.
// Output: IMMDevice interface.
@@ -4928,7 +4939,7 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
// Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
hr = pEndpoint->GetId(&pwszID);
CONTINUE_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "ID string : %S", pwszID);
+ LOG(LS_VERBOSE) << "ID string : " << pwszID;
// Retrieve an interface to the device's property store.
// Output: IPropertyStore interface.
@@ -4949,20 +4960,24 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
PKEY_Device_FriendlyName,
&varName);
CONTINUE_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", varName.pwszVal);
+ LOG(LS_VERBOSE) << "friendly name: \"" << varName.pwszVal << "\"";
// Get the endpoint's current device state
DWORD dwState;
hr = pEndpoint->GetState(&dwState);
CONTINUE_ON_ERROR(hr);
if (dwState & DEVICE_STATE_ACTIVE)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : *ACTIVE*", dwState);
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : *ACTIVE*";
if (dwState & DEVICE_STATE_DISABLED)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : DISABLED", dwState);
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : DISABLED";
if (dwState & DEVICE_STATE_NOTPRESENT)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : NOTPRESENT", dwState);
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : NOTPRESENT";
if (dwState & DEVICE_STATE_UNPLUGGED)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : UNPLUGGED", dwState);
+ LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
+ << ") : UNPLUGGED";
// Check the hardware volume capabilities.
DWORD dwHwSupportMask = 0;
@@ -4973,20 +4988,23 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
CONTINUE_ON_ERROR(hr);
if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
// The audio endpoint device supports a hardware volume control
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_VOLUME", dwHwSupportMask);
+ LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
+ << std::dec << ") : HARDWARE_SUPPORT_VOLUME";
if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_MUTE)
// The audio endpoint device supports a hardware mute control
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_MUTE", dwHwSupportMask);
+ LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
+ << std::dec << ") : HARDWARE_SUPPORT_MUTE";
if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_METER)
// The audio endpoint device supports a hardware peak meter
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_METER", dwHwSupportMask);
+ LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
+ << std::dec << ") : HARDWARE_SUPPORT_METER";
// Check the channel count (#channels in the audio stream that enters or leaves the audio endpoint device)
UINT nChannelCount(0);
hr = pEndpointVolume->GetChannelCount(
&nChannelCount);
CONTINUE_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#channels : %u", nChannelCount);
+ LOG(LS_VERBOSE) << "#channels : " << nChannelCount;
if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
{
@@ -4999,8 +5017,9 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
&fLevelMaxDB,
&fVolumeIncrementDB);
CONTINUE_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "volume range : %4.2f (min), %4.2f (max), %4.2f (inc) [dB]",
- fLevelMinDB, fLevelMaxDB, fVolumeIncrementDB);
+ LOG(LS_VERBOSE) << "volume range : " << fLevelMinDB << " (min), "
+ << fLevelMaxDB << " (max), " << fVolumeIncrementDB
+ << " (inc) [dB]";
// The volume range from vmin = fLevelMinDB to vmax = fLevelMaxDB is divided
// into n uniform intervals of size vinc = fVolumeIncrementDB, where
@@ -5008,7 +5027,7 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
// The values vmin, vmax, and vinc are measured in decibels. The client can set
// the volume level to one of n + 1 discrete values in the range from vmin to vmax.
int n = (int)((fLevelMaxDB-fLevelMinDB)/fVolumeIncrementDB);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#intervals : %d", n);
+ LOG(LS_VERBOSE) << "#intervals : " << n;
// Get information about the current step in the volume range.
// This method represents the volume level of the audio stream that enters or leaves
@@ -5022,12 +5041,12 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow)
&nStep,
&nStepCount);
CONTINUE_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "volume steps : %d (nStep), %d (nStepCount)", nStep, nStepCount);
+ LOG(LS_VERBOSE) << "volume steps : " << nStep << " (nStep), "
+ << nStepCount << " (nStepCount)";
}
Next:
if (FAILED(hr)) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Error when logging device information");
+ LOG(LS_VERBOSE) << "Error when logging device information";
}
CoTaskMemFree(pwszID);
pwszID = NULL;
@@ -5082,11 +5101,10 @@ void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const
errorText[messageLength - 1] = '\0';
}
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Core Audio method failed (hr=0x%x)", hr);
+ LOG(LS_ERROR) << "Core Audio method failed (hr=" << hr << ")";
StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
StringCchCat(buf, MAXERRORLENGTH, errorText);
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
+ LOG(LS_ERROR) << WideToUTF8(buf);
}
// ----------------------------------------------------------------------------
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