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chromium / external / webrtc / 3c1e558449309be965815e1bf / . / webrtc / modules / audio_device / win / audio_device_core_win.cc
blob: 9d1d5322fb16833d77b4df44a6d5675517906d8e [file] [log] [blame]
/*
* 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
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#pragma warning(disable: 4995) // name was marked as #pragma deprecated
#if (_MSC_VER >= 1310) && (_MSC_VER < 1400)
// Reports the major and minor versions of the compiler.
// For example, 1310 for Microsoft Visual C++ .NET 2003. 1310 represents version 13 and a 1.0 point release.
// The Visual C++ 2005 compiler version is 1400.
// Type cl /? at the command line to see the major and minor versions of your compiler along with the build number.
#pragma message(">> INFO: Windows Core Audio is not supported in VS 2003")
#endif
#include "webrtc/modules/audio_device/audio_device_config.h"
#ifdef WEBRTC_WINDOWS_CORE_AUDIO_BUILD
#include "webrtc/modules/audio_device/win/audio_device_core_win.h"
#include <assert.h>
#include <string.h>
#include <windows.h>
#include <comdef.h>
#include <dmo.h>
#include <Functiondiscoverykeys_devpkey.h>
#include <mmsystem.h>
#include <strsafe.h>
#include <uuids.h>
#include "webrtc/base/logging.h"
#include "webrtc/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)
// Macro that continues to a COM error.
#define CONTINUE_ON_ERROR(hres) do { if (FAILED(hres)) goto Next; } while(0)
// Macro that releases a COM object if not NULL.
#define SAFE_RELEASE(p) do { if ((p)) { (p)->Release(); (p) = NULL; } } while(0)
#define ROUND(x) ((x) >=0 ? (int)((x) + 0.5) : (int)((x) - 0.5))
// REFERENCE_TIME time units per millisecond
#define REFTIMES_PER_MILLISEC 10000
typedef struct tagTHREADNAME_INFO
{
DWORD dwType; // must be 0x1000
LPCSTR szName; // pointer to name (in user addr space)
DWORD dwThreadID; // thread ID (-1=caller thread)
DWORD dwFlags; // reserved for future use, must be zero
} THREADNAME_INFO;
namespace webrtc {
namespace {
enum { COM_THREADING_MODEL = COINIT_MULTITHREADED };
enum
{
kAecCaptureStreamIndex = 0,
kAecRenderStreamIndex = 1
};
// An implementation of IMediaBuffer, as required for
// IMediaObject::ProcessOutput(). After consuming data provided by
// ProcessOutput(), call SetLength() to update the buffer availability.
//
// Example implementation:
// http://msdn.microsoft.com/en-us/library/dd376684(v=vs.85).aspx
class MediaBufferImpl : public IMediaBuffer
{
public:
explicit MediaBufferImpl(DWORD maxLength)
: _data(new BYTE[maxLength]),
_length(0),
_maxLength(maxLength),
_refCount(0)
{}
// IMediaBuffer methods.
STDMETHOD(GetBufferAndLength(BYTE** ppBuffer, DWORD* pcbLength))
{
if (!ppBuffer || !pcbLength)
{
return E_POINTER;
}
*ppBuffer = _data;
*pcbLength = _length;
return S_OK;
}
STDMETHOD(GetMaxLength(DWORD* pcbMaxLength))
{
if (!pcbMaxLength)
{
return E_POINTER;
}
*pcbMaxLength = _maxLength;
return S_OK;
}
STDMETHOD(SetLength(DWORD cbLength))
{
if (cbLength > _maxLength)
{
return E_INVALIDARG;
}
_length = cbLength;
return S_OK;
}
// IUnknown methods.
STDMETHOD_(ULONG, AddRef())
{
return InterlockedIncrement(&_refCount);
}
STDMETHOD(QueryInterface(REFIID riid, void** ppv))
{
if (!ppv)
{
return E_POINTER;
}
else if (riid != IID_IMediaBuffer && riid != IID_IUnknown)
{
return E_NOINTERFACE;
}
*ppv = static_cast<IMediaBuffer*>(this);
AddRef();
return S_OK;
}
STDMETHOD_(ULONG, Release())
{
LONG refCount = InterlockedDecrement(&_refCount);
if (refCount == 0)
{
delete this;
}
return refCount;
}
private:
~MediaBufferImpl()
{
delete [] _data;
}
BYTE* _data;
DWORD _length;
const DWORD _maxLength;
LONG _refCount;
};
} // namespace
// ============================================================================
// Static Methods
// ============================================================================
// ----------------------------------------------------------------------------
// CoreAudioIsSupported
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::CoreAudioIsSupported()
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%s", __FUNCTION__);
bool MMDeviceIsAvailable(false);
bool coreAudioIsSupported(false);
HRESULT hr(S_OK);
TCHAR buf[MAXERRORLENGTH];
TCHAR errorText[MAXERRORLENGTH];
// 1) Check if Windows version is Vista SP1 or later.
//
// CoreAudio is only available on Vista SP1 and later.
//
OSVERSIONINFOEX osvi;
DWORDLONG dwlConditionMask = 0;
int op = VER_LESS_EQUAL;
// Initialize the OSVERSIONINFOEX structure.
ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
osvi.dwMajorVersion = 6;
osvi.dwMinorVersion = 0;
osvi.wServicePackMajor = 0;
osvi.wServicePackMinor = 0;
osvi.wProductType = VER_NT_WORKSTATION;
// Initialize the condition mask.
VER_SET_CONDITION(dwlConditionMask, VER_MAJORVERSION, op);
VER_SET_CONDITION(dwlConditionMask, VER_MINORVERSION, op);
VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMAJOR, op);
VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMINOR, op);
VER_SET_CONDITION(dwlConditionMask, VER_PRODUCT_TYPE, VER_EQUAL);
DWORD dwTypeMask = VER_MAJORVERSION | VER_MINORVERSION |
VER_SERVICEPACKMAJOR | VER_SERVICEPACKMINOR |
VER_PRODUCT_TYPE;
// Perform the test.
BOOL isVistaRTMorXP = VerifyVersionInfo(&osvi, dwTypeMask,
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 ***");
return false;
}
// 2) Initializes the COM library for use by the calling thread.
// The COM init wrapper sets the thread's concurrency model to MTA,
// and creates a new apartment for the thread if one is required. The
// wrapper also ensures that each call to CoInitializeEx is balanced
// by a corresponding call to CoUninitialize.
//
ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
if (!comInit.succeeded()) {
// Things will work even if an STA thread is calling this method but we
// want to ensure that MTA is used and therefore return false here.
return false;
}
// 3) Check if the MMDevice API is available.
//
// The Windows Multimedia Device (MMDevice) API enables audio clients to
// discover audio endpoint devices, determine their capabilities, and create
// driver instances for those devices.
// Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
// The MMDevice API consists of several interfaces. The first of these is the
// IMMDeviceEnumerator interface. To access the interfaces in the MMDevice API,
// a client obtains a reference to the IMMDeviceEnumerator interface of a
// device-enumerator object by calling the CoCreateInstance function.
//
// Through the IMMDeviceEnumerator interface, the client can obtain references
// to the other interfaces in the MMDevice API. The MMDevice API implements
// the following interfaces:
//
// IMMDevice Represents an audio device.
// IMMDeviceCollection Represents a collection of audio devices.
// IMMDeviceEnumerator Provides methods for enumerating audio devices.
// IMMEndpoint Represents an audio endpoint device.
//
IMMDeviceEnumerator* pIMMD(NULL);
const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
hr = CoCreateInstance(
CLSID_MMDeviceEnumerator, // GUID value of MMDeviceEnumerator coclass
NULL,
CLSCTX_ALL,
IID_IMMDeviceEnumerator, // GUID value of the IMMDeviceEnumerator interface
(void**)&pIMMD );
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);
const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
FORMAT_MESSAGE_IGNORE_INSERTS;
const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
// Gets the system's human readable message string for this HRESULT.
// All error message in English by default.
DWORD messageLength = ::FormatMessageW(dwFlags,
0,
hr,
dwLangID,
errorText,
MAXERRORLENGTH,
NULL);
assert(messageLength <= MAXERRORLENGTH);
// Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
for (; messageLength && ::isspace(errorText[messageLength - 1]);
--messageLength)
{
errorText[messageLength - 1] = '\0';
}
StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
StringCchCat(buf, MAXERRORLENGTH, errorText);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%S", buf);
}
else
{
MMDeviceIsAvailable = true;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1,
"AudioDeviceWindowsCore::CoreAudioIsSupported() CoCreateInstance(MMDeviceEnumerator) succeeded", hr);
SAFE_RELEASE(pIMMD);
}
// 4) Verify that we can create and initialize our Core Audio class.
//
// Also, perform a limited "API test" to ensure that Core Audio is supported for all devices.
//
if (MMDeviceIsAvailable)
{
coreAudioIsSupported = false;
AudioDeviceWindowsCore* p = new AudioDeviceWindowsCore(-1);
if (p == NULL)
{
return false;
}
int ok(0);
int temp_ok(0);
bool available(false);
if (p->Init() != InitStatus::OK) {
ok |= -1;
}
int16_t numDevsRec = p->RecordingDevices();
for (uint16_t i = 0; i < numDevsRec; i++)
{
ok |= p->SetRecordingDevice(i);
temp_ok = p->RecordingIsAvailable(available);
ok |= temp_ok;
ok |= (available == false);
if (available)
{
ok |= p->InitMicrophone();
}
if (ok)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
"AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Recording for device id=%i", i);
}
}
int16_t numDevsPlay = p->PlayoutDevices();
for (uint16_t i = 0; i < numDevsPlay; i++)
{
ok |= p->SetPlayoutDevice(i);
temp_ok = p->PlayoutIsAvailable(available);
ok |= temp_ok;
ok |= (available == false);
if (available)
{
ok |= p->InitSpeaker();
}
if (ok)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1 ,
"AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Playout for device id=%i", i);
}
}
ok |= p->Terminate();
if (ok == 0)
{
coreAudioIsSupported = true;
}
delete p;
}
if (coreAudioIsSupported)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is supported ***");
}
else
{
WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is NOT supported => will revert to the Wave API ***");
}
return (coreAudioIsSupported);
}
// ============================================================================
// Construction & Destruction
// ============================================================================
// ----------------------------------------------------------------------------
// AudioDeviceWindowsCore() - ctor
// ----------------------------------------------------------------------------
AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id) :
_comInit(ScopedCOMInitializer::kMTA),
_id(id),
_ptrAudioBuffer(NULL),
_ptrEnumerator(NULL),
_ptrRenderCollection(NULL),
_ptrCaptureCollection(NULL),
_ptrDeviceOut(NULL),
_ptrDeviceIn(NULL),
_ptrClientOut(NULL),
_ptrClientIn(NULL),
_ptrRenderClient(NULL),
_ptrCaptureClient(NULL),
_ptrCaptureVolume(NULL),
_ptrRenderSimpleVolume(NULL),
_dmo(NULL),
_mediaBuffer(NULL),
_builtInAecEnabled(false),
_playAudioFrameSize(0),
_playSampleRate(0),
_playBlockSize(0),
_playChannels(2),
_sndCardPlayDelay(0),
_sndCardRecDelay(0),
_writtenSamples(0),
_readSamples(0),
_playAcc(0),
_recAudioFrameSize(0),
_recSampleRate(0),
_recBlockSize(0),
_recChannels(2),
_avrtLibrary(NULL),
_winSupportAvrt(false),
_hRenderSamplesReadyEvent(NULL),
_hPlayThread(NULL),
_hCaptureSamplesReadyEvent(NULL),
_hRecThread(NULL),
_hShutdownRenderEvent(NULL),
_hShutdownCaptureEvent(NULL),
_hRenderStartedEvent(NULL),
_hCaptureStartedEvent(NULL),
_hGetCaptureVolumeThread(NULL),
_hSetCaptureVolumeThread(NULL),
_hSetCaptureVolumeEvent(NULL),
_hMmTask(NULL),
_initialized(false),
_recording(false),
_playing(false),
_recIsInitialized(false),
_playIsInitialized(false),
_speakerIsInitialized(false),
_microphoneIsInitialized(false),
_AGC(false),
_playWarning(0),
_playError(0),
_recWarning(0),
_recError(0),
_playBufType(AudioDeviceModule::kAdaptiveBufferSize),
_playBufDelay(80),
_playBufDelayFixed(80),
_usingInputDeviceIndex(false),
_usingOutputDeviceIndex(false),
_inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_outputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_inputDeviceIndex(0),
_outputDeviceIndex(0),
_newMicLevel(0)
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
assert(_comInit.succeeded());
// Try to load the Avrt DLL
if (!_avrtLibrary)
{
// Get handle to the Avrt DLL module.
_avrtLibrary = LoadLibrary(TEXT("Avrt.dll"));
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");
_PAvRevertMmThreadCharacteristics = (PAvRevertMmThreadCharacteristics)GetProcAddress(_avrtLibrary, "AvRevertMmThreadCharacteristics");
_PAvSetMmThreadCharacteristicsA = (PAvSetMmThreadCharacteristicsA)GetProcAddress(_avrtLibrary, "AvSetMmThreadCharacteristicsA");
_PAvSetMmThreadPriority = (PAvSetMmThreadPriority)GetProcAddress(_avrtLibrary, "AvSetMmThreadPriority");
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");
_winSupportAvrt = true;
}
}
}
// Create our samples ready events - we want auto reset events that start in the not-signaled state.
// The state of an auto-reset event object remains signaled until a single waiting thread is released,
// at which time the system automatically sets the state to nonsignaled. If no threads are waiting,
// the event object's state remains signaled.
// (Except for _hShutdownCaptureEvent, which is used to shutdown multiple threads).
_hRenderSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hCaptureSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hShutdownRenderEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hShutdownCaptureEvent = CreateEvent(NULL, TRUE, FALSE, NULL);
_hRenderStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hCaptureStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_perfCounterFreq.QuadPart = 1;
_perfCounterFactor = 0.0;
_avgCPULoad = 0.0;
// list of number of channels to use on recording side
_recChannelsPrioList[0] = 2; // stereo is prio 1
_recChannelsPrioList[1] = 1; // mono is prio 2
_recChannelsPrioList[2] = 4; // quad is prio 3
// list of number of channels to use on playout side
_playChannelsPrioList[0] = 2; // stereo is prio 1
_playChannelsPrioList[1] = 1; // mono is prio 2
HRESULT hr;
// We know that this API will work since it has already been verified in
// CoreAudioIsSupported, hence no need to check for errors here as well.
// Retrive the IMMDeviceEnumerator API (should load the MMDevAPI.dll)
// TODO(henrika): we should probably move this allocation to Init() instead
// and deallocate in Terminate() to make the implementation more symmetric.
CoCreateInstance(
__uuidof(MMDeviceEnumerator),
NULL,
CLSCTX_ALL,
__uuidof(IMMDeviceEnumerator),
reinterpret_cast<void**>(&_ptrEnumerator));
assert(NULL != _ptrEnumerator);
// DMO initialization for built-in WASAPI AEC.
{
IMediaObject* ptrDMO = NULL;
hr = CoCreateInstance(CLSID_CWMAudioAEC,
NULL,
CLSCTX_INPROC_SERVER,
IID_IMediaObject,
reinterpret_cast<void**>(&ptrDMO));
if (FAILED(hr) || ptrDMO == NULL)
{
// Since we check that _dmo is non-NULL in EnableBuiltInAEC(), the
// feature is prevented from being enabled.
_builtInAecEnabled = false;
_TraceCOMError(hr);
}
_dmo = ptrDMO;
SAFE_RELEASE(ptrDMO);
}
}
// ----------------------------------------------------------------------------
// AudioDeviceWindowsCore() - dtor
// ----------------------------------------------------------------------------
AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
Terminate();
// The IMMDeviceEnumerator is created during construction. Must release
// it here and not in Terminate() since we don't recreate it in Init().
SAFE_RELEASE(_ptrEnumerator);
_ptrAudioBuffer = NULL;
if (NULL != _hRenderSamplesReadyEvent)
{
CloseHandle(_hRenderSamplesReadyEvent);
_hRenderSamplesReadyEvent = NULL;
}
if (NULL != _hCaptureSamplesReadyEvent)
{
CloseHandle(_hCaptureSamplesReadyEvent);
_hCaptureSamplesReadyEvent = NULL;
}
if (NULL != _hRenderStartedEvent)
{
CloseHandle(_hRenderStartedEvent);
_hRenderStartedEvent = NULL;
}
if (NULL != _hCaptureStartedEvent)
{
CloseHandle(_hCaptureStartedEvent);
_hCaptureStartedEvent = NULL;
}
if (NULL != _hShutdownRenderEvent)
{
CloseHandle(_hShutdownRenderEvent);
_hShutdownRenderEvent = NULL;
}
if (NULL != _hShutdownCaptureEvent)
{
CloseHandle(_hShutdownCaptureEvent);
_hShutdownCaptureEvent = NULL;
}
if (NULL != _hSetCaptureVolumeEvent)
{
CloseHandle(_hSetCaptureVolumeEvent);
_hSetCaptureVolumeEvent = NULL;
}
if (_avrtLibrary)
{
BOOL freeOK = FreeLibrary(_avrtLibrary);
if (!freeOK)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"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");
}
}
}
// ============================================================================
// API
// ============================================================================
// ----------------------------------------------------------------------------
// AttachAudioBuffer
// ----------------------------------------------------------------------------
void AudioDeviceWindowsCore::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
{
_ptrAudioBuffer = audioBuffer;
// Inform the AudioBuffer about default settings for this implementation.
// Set all values to zero here since the actual settings will be done by
// InitPlayout and InitRecording later.
_ptrAudioBuffer->SetRecordingSampleRate(0);
_ptrAudioBuffer->SetPlayoutSampleRate(0);
_ptrAudioBuffer->SetRecordingChannels(0);
_ptrAudioBuffer->SetPlayoutChannels(0);
}
// ----------------------------------------------------------------------------
// ActiveAudioLayer
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
{
audioLayer = AudioDeviceModule::kWindowsCoreAudio;
return 0;
}
// ----------------------------------------------------------------------------
// Init
// ----------------------------------------------------------------------------
AudioDeviceGeneric::InitStatus AudioDeviceWindowsCore::Init() {
rtc::CritScope lock(&_critSect);
if (_initialized) {
return InitStatus::OK;
}
_playWarning = 0;
_playError = 0;
_recWarning = 0;
_recError = 0;
// Enumerate all audio rendering and capturing endpoint devices.
// Note that, some of these will not be able to select by the user.
// The complete collection is for internal use only.
_EnumerateEndpointDevicesAll(eRender);
_EnumerateEndpointDevicesAll(eCapture);
_initialized = true;
return InitStatus::OK;
}
// ----------------------------------------------------------------------------
// Terminate
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::Terminate()
{
rtc::CritScope lock(&_critSect);
if (!_initialized) {
return 0;
}
_initialized = false;
_speakerIsInitialized = false;
_microphoneIsInitialized = false;
_playing = false;
_recording = false;
SAFE_RELEASE(_ptrRenderCollection);
SAFE_RELEASE(_ptrCaptureCollection);
SAFE_RELEASE(_ptrDeviceOut);
SAFE_RELEASE(_ptrDeviceIn);
SAFE_RELEASE(_ptrClientOut);
SAFE_RELEASE(_ptrClientIn);
SAFE_RELEASE(_ptrRenderClient);
SAFE_RELEASE(_ptrCaptureClient);
SAFE_RELEASE(_ptrCaptureVolume);
SAFE_RELEASE(_ptrRenderSimpleVolume);
return 0;
}
// ----------------------------------------------------------------------------
// Initialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::Initialized() const
{
return (_initialized);
}
// ----------------------------------------------------------------------------
// InitSpeaker
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::InitSpeaker()
{
rtc::CritScope lock(&_critSect);
if (_playing)
{
return -1;
}
if (_ptrDeviceOut == NULL)
{
return -1;
}
if (_usingOutputDeviceIndex)
{
int16_t nDevices = PlayoutDevices();
if (_outputDeviceIndex > (nDevices - 1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
return -1;
}
}
int32_t ret(0);
SAFE_RELEASE(_ptrDeviceOut);
if (_usingOutputDeviceIndex)
{
// Refresh the selected rendering endpoint device using current index
ret = _GetListDevice(eRender, _outputDeviceIndex, &_ptrDeviceOut);
}
else
{
ERole role;
(_outputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
// Refresh the selected rendering endpoint device using role
ret = _GetDefaultDevice(eRender, role, &_ptrDeviceOut);
}
if (ret != 0 || (_ptrDeviceOut == NULL))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the rendering enpoint device");
SAFE_RELEASE(_ptrDeviceOut);
return -1;
}
IAudioSessionManager* pManager = NULL;
ret = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager),
CLSCTX_ALL,
NULL,
(void**)&pManager);
if (ret != 0 || pManager == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" failed to initialize the render manager");
SAFE_RELEASE(pManager);
return -1;
}
SAFE_RELEASE(_ptrRenderSimpleVolume);
ret = pManager->GetSimpleAudioVolume(NULL, FALSE, &_ptrRenderSimpleVolume);
if (ret != 0 || _ptrRenderSimpleVolume == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" failed to initialize the render simple volume");
SAFE_RELEASE(pManager);
SAFE_RELEASE(_ptrRenderSimpleVolume);
return -1;
}
SAFE_RELEASE(pManager);
_speakerIsInitialized = true;
return 0;
}
// ----------------------------------------------------------------------------
// InitMicrophone
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::InitMicrophone()
{
rtc::CritScope lock(&_critSect);
if (_recording)
{
return -1;
}
if (_ptrDeviceIn == NULL)
{
return -1;
}
if (_usingInputDeviceIndex)
{
int16_t nDevices = RecordingDevices();
if (_inputDeviceIndex > (nDevices - 1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
return -1;
}
}
int32_t ret(0);
SAFE_RELEASE(_ptrDeviceIn);
if (_usingInputDeviceIndex)
{
// Refresh the selected capture endpoint device using current index
ret = _GetListDevice(eCapture, _inputDeviceIndex, &_ptrDeviceIn);
}
else
{
ERole role;
(_inputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
// Refresh the selected capture endpoint device using role
ret = _GetDefaultDevice(eCapture, role, &_ptrDeviceIn);
}
if (ret != 0 || (_ptrDeviceIn == NULL))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the capturing enpoint device");
SAFE_RELEASE(_ptrDeviceIn);
return -1;
}
SAFE_RELEASE(_ptrCaptureVolume);
ret = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume),
CLSCTX_ALL,
NULL,
reinterpret_cast<void **>(&_ptrCaptureVolume));
if (ret != 0 || _ptrCaptureVolume == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" failed to initialize the capture volume");
SAFE_RELEASE(_ptrCaptureVolume);
return -1;
}
_microphoneIsInitialized = true;
return 0;
}
// ----------------------------------------------------------------------------
// SpeakerIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::SpeakerIsInitialized() const
{
return (_speakerIsInitialized);
}
// ----------------------------------------------------------------------------
// MicrophoneIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const
{
return (_microphoneIsInitialized);
}
// ----------------------------------------------------------------------------
// SpeakerVolumeIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available)
{
rtc::CritScope lock(&_critSect);
if (_ptrDeviceOut == NULL)
{
return -1;
}
HRESULT hr = S_OK;
IAudioSessionManager* pManager = NULL;
ISimpleAudioVolume* pVolume = NULL;
hr = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL, NULL, (void**)&pManager);
EXIT_ON_ERROR(hr);
hr = pManager->GetSimpleAudioVolume(NULL, FALSE, &pVolume);
EXIT_ON_ERROR(hr);
float volume(0.0f);
hr = pVolume->GetMasterVolume(&volume);
if (FAILED(hr))
{
available = false;
}
available = true;
SAFE_RELEASE(pManager);
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pManager);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// SetSpeakerVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetSpeakerVolume(uint32_t volume)
{
{
rtc::CritScope lock(&_critSect);
if (!_speakerIsInitialized)
{
return -1;
}
if (_ptrDeviceOut == NULL)
{
return -1;
}
}
if (volume < (uint32_t)MIN_CORE_SPEAKER_VOLUME ||
volume > (uint32_t)MAX_CORE_SPEAKER_VOLUME)
{
return -1;
}
HRESULT hr = S_OK;
// scale input volume to valid range (0.0 to 1.0)
const float fLevel = (float)volume/MAX_CORE_SPEAKER_VOLUME;
_volumeMutex.Enter();
hr = _ptrRenderSimpleVolume->SetMasterVolume(fLevel,NULL);
_volumeMutex.Leave();
EXIT_ON_ERROR(hr);
return 0;
Exit:
_TraceCOMError(hr);
return -1;
}
// ----------------------------------------------------------------------------
// SpeakerVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SpeakerVolume(uint32_t& volume) const
{
{
rtc::CritScope lock(&_critSect);
if (!_speakerIsInitialized)
{
return -1;
}
if (_ptrDeviceOut == NULL)
{
return -1;
}
}
HRESULT hr = S_OK;
float fLevel(0.0f);
_volumeMutex.Enter();
hr = _ptrRenderSimpleVolume->GetMasterVolume(&fLevel);
_volumeMutex.Leave();
EXIT_ON_ERROR(hr);
// scale input volume range [0.0,1.0] to valid output range
volume = static_cast<uint32_t> (fLevel*MAX_CORE_SPEAKER_VOLUME);
return 0;
Exit:
_TraceCOMError(hr);
return -1;
}
// ----------------------------------------------------------------------------
// SetWaveOutVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight)
{
return -1;
}
// ----------------------------------------------------------------------------
// WaveOutVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::WaveOutVolume(uint16_t& volumeLeft, uint16_t& volumeRight) const
{
return -1;
}
// ----------------------------------------------------------------------------
// MaxSpeakerVolume
//
// The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
// silence and 1.0 indicates full volume (no attenuation).
// We add our (webrtc-internal) own max level to match the Wave API and
// how it is used today in VoE.
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MaxSpeakerVolume(uint32_t& maxVolume) const
{
if (!_speakerIsInitialized)
{
return -1;
}
maxVolume = static_cast<uint32_t> (MAX_CORE_SPEAKER_VOLUME);
return 0;
}
// ----------------------------------------------------------------------------
// MinSpeakerVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MinSpeakerVolume(uint32_t& minVolume) const
{
if (!_speakerIsInitialized)
{
return -1;
}
minVolume = static_cast<uint32_t> (MIN_CORE_SPEAKER_VOLUME);
return 0;
}
// ----------------------------------------------------------------------------
// SpeakerVolumeStepSize
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SpeakerVolumeStepSize(uint16_t& stepSize) const
{
if (!_speakerIsInitialized)
{
return -1;
}
stepSize = CORE_SPEAKER_VOLUME_STEP_SIZE;
return 0;
}
// ----------------------------------------------------------------------------
// SpeakerMuteIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available)
{
rtc::CritScope lock(&_critSect);
if (_ptrDeviceOut == NULL)
{
return -1;
}
HRESULT hr = S_OK;
IAudioEndpointVolume* pVolume = NULL;
// Query the speaker system mute state.
hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume),
CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
EXIT_ON_ERROR(hr);
BOOL mute;
hr = pVolume->GetMute(&mute);
if (FAILED(hr))
available = false;
else
available = true;
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// SetSpeakerMute
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetSpeakerMute(bool enable)
{
rtc::CritScope lock(&_critSect);
if (!_speakerIsInitialized)
{
return -1;
}
if (_ptrDeviceOut == NULL)
{
return -1;
}
HRESULT hr = S_OK;
IAudioEndpointVolume* pVolume = NULL;
// Set the speaker system mute state.
hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
EXIT_ON_ERROR(hr);
const BOOL mute(enable);
hr = pVolume->SetMute(mute, NULL);
EXIT_ON_ERROR(hr);
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// SpeakerMute
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SpeakerMute(bool& enabled) const
{
if (!_speakerIsInitialized)
{
return -1;
}
if (_ptrDeviceOut == NULL)
{
return -1;
}
HRESULT hr = S_OK;
IAudioEndpointVolume* pVolume = NULL;
// Query the speaker system mute state.
hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
EXIT_ON_ERROR(hr);
BOOL mute;
hr = pVolume->GetMute(&mute);
EXIT_ON_ERROR(hr);
enabled = (mute == TRUE) ? true : false;
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneMuteIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available)
{
rtc::CritScope lock(&_critSect);
if (_ptrDeviceIn == NULL)
{
return -1;
}
HRESULT hr = S_OK;
IAudioEndpointVolume* pVolume = NULL;
// Query the microphone system mute state.
hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
EXIT_ON_ERROR(hr);
BOOL mute;
hr = pVolume->GetMute(&mute);
if (FAILED(hr))
available = false;
else
available = true;
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// SetMicrophoneMute
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetMicrophoneMute(bool enable)
{
if (!_microphoneIsInitialized)
{
return -1;
}
if (_ptrDeviceIn == NULL)
{
return -1;
}
HRESULT hr = S_OK;
IAudioEndpointVolume* pVolume = NULL;
// Set the microphone system mute state.
hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
EXIT_ON_ERROR(hr);
const BOOL mute(enable);
hr = pVolume->SetMute(mute, NULL);
EXIT_ON_ERROR(hr);
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneMute
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MicrophoneMute(bool& enabled) const
{
if (!_microphoneIsInitialized)
{
return -1;
}
HRESULT hr = S_OK;
IAudioEndpointVolume* pVolume = NULL;
// Query the microphone system mute state.
hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
EXIT_ON_ERROR(hr);
BOOL mute;
hr = pVolume->GetMute(&mute);
EXIT_ON_ERROR(hr);
enabled = (mute == TRUE) ? true : false;
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneBoostIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MicrophoneBoostIsAvailable(bool& available)
{
available = false;
return 0;
}
// ----------------------------------------------------------------------------
// SetMicrophoneBoost
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetMicrophoneBoost(bool enable)
{
if (!_microphoneIsInitialized)
{
return -1;
}
return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneBoost
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MicrophoneBoost(bool& enabled) const
{
if (!_microphoneIsInitialized)
{
return -1;
}
return -1;
}
// ----------------------------------------------------------------------------
// StereoRecordingIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available)
{
available = true;
return 0;
}
// ----------------------------------------------------------------------------
// SetStereoRecording
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetStereoRecording(bool enable)
{
rtc::CritScope lock(&_critSect);
if (enable)
{
_recChannelsPrioList[0] = 2; // try stereo first
_recChannelsPrioList[1] = 1;
_recChannels = 2;
}
else
{
_recChannelsPrioList[0] = 1; // try mono first
_recChannelsPrioList[1] = 2;
_recChannels = 1;
}
return 0;
}
// ----------------------------------------------------------------------------
// StereoRecording
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StereoRecording(bool& enabled) const
{
if (_recChannels == 2)
enabled = true;
else
enabled = false;
return 0;
}
// ----------------------------------------------------------------------------
// StereoPlayoutIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available)
{
available = true;
return 0;
}
// ----------------------------------------------------------------------------
// SetStereoPlayout
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetStereoPlayout(bool enable)
{
rtc::CritScope lock(&_critSect);
if (enable)
{
_playChannelsPrioList[0] = 2; // try stereo first
_playChannelsPrioList[1] = 1;
_playChannels = 2;
}
else
{
_playChannelsPrioList[0] = 1; // try mono first
_playChannelsPrioList[1] = 2;
_playChannels = 1;
}
return 0;
}
// ----------------------------------------------------------------------------
// StereoPlayout
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const
{
if (_playChannels == 2)
enabled = true;
else
enabled = false;
return 0;
}
// ----------------------------------------------------------------------------
// SetAGC
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetAGC(bool enable)
{
rtc::CritScope lock(&_critSect);
_AGC = enable;
return 0;
}
// ----------------------------------------------------------------------------
// AGC
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::AGC() const
{
rtc::CritScope lock(&_critSect);
return _AGC;
}
// ----------------------------------------------------------------------------
// MicrophoneVolumeIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available)
{
rtc::CritScope lock(&_critSect);
if (_ptrDeviceIn == NULL)
{
return -1;
}
HRESULT hr = S_OK;
IAudioEndpointVolume* pVolume = NULL;
hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
EXIT_ON_ERROR(hr);
float volume(0.0f);
hr = pVolume->GetMasterVolumeLevelScalar(&volume);
if (FAILED(hr))
{
available = false;
}
available = true;
SAFE_RELEASE(pVolume);
return 0;
Exit:
_TraceCOMError(hr);
SAFE_RELEASE(pVolume);
return -1;
}
// ----------------------------------------------------------------------------
// SetMicrophoneVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetMicrophoneVolume(uint32_t volume)
{
WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::SetMicrophoneVolume(volume=%u)", volume);
{
rtc::CritScope lock(&_critSect);
if (!_microphoneIsInitialized)
{
return -1;
}
if (_ptrDeviceIn == NULL)
{
return -1;
}
}
if (volume < static_cast<uint32_t>(MIN_CORE_MICROPHONE_VOLUME) ||
volume > static_cast<uint32_t>(MAX_CORE_MICROPHONE_VOLUME))
{
return -1;
}
HRESULT hr = S_OK;
// scale input volume to valid range (0.0 to 1.0)
const float fLevel = static_cast<float>(volume)/MAX_CORE_MICROPHONE_VOLUME;
_volumeMutex.Enter();
_ptrCaptureVolume->SetMasterVolumeLevelScalar(fLevel, NULL);
_volumeMutex.Leave();
EXIT_ON_ERROR(hr);
return 0;
Exit:
_TraceCOMError(hr);
return -1;
}
// ----------------------------------------------------------------------------
// MicrophoneVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MicrophoneVolume(uint32_t& volume) const
{
{
rtc::CritScope lock(&_critSect);
if (!_microphoneIsInitialized)
{
return -1;
}
if (_ptrDeviceIn == NULL)
{
return -1;
}
}
HRESULT hr = S_OK;
float fLevel(0.0f);
volume = 0;
_volumeMutex.Enter();
hr = _ptrCaptureVolume->GetMasterVolumeLevelScalar(&fLevel);
_volumeMutex.Leave();
EXIT_ON_ERROR(hr);
// scale input volume range [0.0,1.0] to valid output range
volume = static_cast<uint32_t> (fLevel*MAX_CORE_MICROPHONE_VOLUME);
return 0;
Exit:
_TraceCOMError(hr);
return -1;
}
// ----------------------------------------------------------------------------
// MaxMicrophoneVolume
//
// The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
// silence and 1.0 indicates full volume (no attenuation).
// We add our (webrtc-internal) own max level to match the Wave API and
// how it is used today in VoE.
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MaxMicrophoneVolume(uint32_t& maxVolume) const
{
WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
if (!_microphoneIsInitialized)
{
return -1;
}
maxVolume = static_cast<uint32_t> (MAX_CORE_MICROPHONE_VOLUME);
return 0;
}
// ----------------------------------------------------------------------------
// MinMicrophoneVolume
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MinMicrophoneVolume(uint32_t& minVolume) const
{
if (!_microphoneIsInitialized)
{
return -1;
}
minVolume = static_cast<uint32_t> (MIN_CORE_MICROPHONE_VOLUME);
return 0;
}
// ----------------------------------------------------------------------------
// MicrophoneVolumeStepSize
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::MicrophoneVolumeStepSize(uint16_t& stepSize) const
{
if (!_microphoneIsInitialized)
{
return -1;
}
stepSize = CORE_MICROPHONE_VOLUME_STEP_SIZE;
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDevices
// ----------------------------------------------------------------------------
int16_t AudioDeviceWindowsCore::PlayoutDevices()
{
rtc::CritScope lock(&_critSect);
if (_RefreshDeviceList(eRender) != -1)
{
return (_DeviceListCount(eRender));
}
return -1;
}
// ----------------------------------------------------------------------------
// SetPlayoutDevice I (II)
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index)
{
if (_playIsInitialized)
{
return -1;
}
// Get current number of available rendering endpoint devices and refresh the rendering collection.
UINT nDevices = PlayoutDevices();
if (index < 0 || index > (nDevices-1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
return -1;
}
rtc::CritScope lock(&_critSect);
HRESULT hr(S_OK);
assert(_ptrRenderCollection != NULL);
// Select an endpoint rendering device given the specified index
SAFE_RELEASE(_ptrDeviceOut);
hr = _ptrRenderCollection->Item(
index,
&_ptrDeviceOut);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceOut);
return -1;
}
WCHAR szDeviceName[MAX_PATH];
const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
}
_usingOutputDeviceIndex = true;
_outputDeviceIndex = index;
return 0;
}
// ----------------------------------------------------------------------------
// SetPlayoutDevice II (II)
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
{
if (_playIsInitialized)
{
return -1;
}
ERole role(eCommunications);
if (device == AudioDeviceModule::kDefaultDevice)
{
role = eConsole;
}
else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
{
role = eCommunications;
}
rtc::CritScope lock(&_critSect);
// Refresh the list of rendering endpoint devices
_RefreshDeviceList(eRender);
HRESULT hr(S_OK);
assert(_ptrEnumerator != NULL);
// Select an endpoint rendering device given the specified role
SAFE_RELEASE(_ptrDeviceOut);
hr = _ptrEnumerator->GetDefaultAudioEndpoint(
eRender,
role,
&_ptrDeviceOut);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceOut);
return -1;
}
WCHAR szDeviceName[MAX_PATH];
const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
}
_usingOutputDeviceIndex = false;
_outputDevice = device;
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDeviceName
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize])
{
bool defaultCommunicationDevice(false);
const int16_t nDevices(PlayoutDevices()); // also updates the list of devices
// Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
if (index == (uint16_t)(-1))
{
defaultCommunicationDevice = true;
index = 0;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
}
if ((index > (nDevices-1)) || (name == NULL))
{
return -1;
}
memset(name, 0, kAdmMaxDeviceNameSize);
if (guid != NULL)
{
memset(guid, 0, kAdmMaxGuidSize);
}
rtc::CritScope lock(&_critSect);
int32_t ret(-1);
WCHAR szDeviceName[MAX_PATH];
const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
// Get the endpoint device's friendly-name
if (defaultCommunicationDevice)
{
ret = _GetDefaultDeviceName(eRender, eCommunications, szDeviceName, bufferLen);
}
else
{
ret = _GetListDeviceName(eRender, index, szDeviceName, bufferLen);
}
if (ret == 0)
{
// 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());
}
}
// Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
if (defaultCommunicationDevice)
{
ret = _GetDefaultDeviceID(eRender, eCommunications, szDeviceName, bufferLen);
}
else
{
ret = _GetListDeviceID(eRender, index, szDeviceName, bufferLen);
}
if (guid != NULL && ret == 0)
{
// 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());
}
}
return ret;
}
// ----------------------------------------------------------------------------
// RecordingDeviceName
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize])
{
bool defaultCommunicationDevice(false);
const int16_t nDevices(RecordingDevices()); // also updates the list of devices
// Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
if (index == (uint16_t)(-1))
{
defaultCommunicationDevice = true;
index = 0;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
}
if ((index > (nDevices-1)) || (name == NULL))
{
return -1;
}
memset(name, 0, kAdmMaxDeviceNameSize);
if (guid != NULL)
{
memset(guid, 0, kAdmMaxGuidSize);
}
rtc::CritScope lock(&_critSect);
int32_t ret(-1);
WCHAR szDeviceName[MAX_PATH];
const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
// Get the endpoint device's friendly-name
if (defaultCommunicationDevice)
{
ret = _GetDefaultDeviceName(eCapture, eCommunications, szDeviceName, bufferLen);
}
else
{
ret = _GetListDeviceName(eCapture, index, szDeviceName, bufferLen);
}
if (ret == 0)
{
// 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());
}
}
// Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
if (defaultCommunicationDevice)
{
ret = _GetDefaultDeviceID(eCapture, eCommunications, szDeviceName, bufferLen);
}
else
{
ret = _GetListDeviceID(eCapture, index, szDeviceName, bufferLen);
}
if (guid != NULL && ret == 0)
{
// 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());
}
}
return ret;
}
// ----------------------------------------------------------------------------
// RecordingDevices
// ----------------------------------------------------------------------------
int16_t AudioDeviceWindowsCore::RecordingDevices()
{
rtc::CritScope lock(&_critSect);
if (_RefreshDeviceList(eCapture) != -1)
{
return (_DeviceListCount(eCapture));
}
return -1;
}
// ----------------------------------------------------------------------------
// SetRecordingDevice I (II)
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index)
{
if (_recIsInitialized)
{
return -1;
}
// Get current number of available capture endpoint devices and refresh the capture collection.
UINT nDevices = RecordingDevices();
if (index < 0 || index > (nDevices-1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
return -1;
}
rtc::CritScope lock(&_critSect);
HRESULT hr(S_OK);
assert(_ptrCaptureCollection != NULL);
// Select an endpoint capture device given the specified index
SAFE_RELEASE(_ptrDeviceIn);
hr = _ptrCaptureCollection->Item(
index,
&_ptrDeviceIn);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceIn);
return -1;
}
WCHAR szDeviceName[MAX_PATH];
const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
}
_usingInputDeviceIndex = true;
_inputDeviceIndex = index;
return 0;
}
// ----------------------------------------------------------------------------
// SetRecordingDevice II (II)
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
{
if (_recIsInitialized)
{
return -1;
}
ERole role(eCommunications);
if (device == AudioDeviceModule::kDefaultDevice)
{
role = eConsole;
}
else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
{
role = eCommunications;
}
rtc::CritScope lock(&_critSect);
// Refresh the list of capture endpoint devices
_RefreshDeviceList(eCapture);
HRESULT hr(S_OK);
assert(_ptrEnumerator != NULL);
// Select an endpoint capture device given the specified role
SAFE_RELEASE(_ptrDeviceIn);
hr = _ptrEnumerator->GetDefaultAudioEndpoint(
eCapture,
role,
&_ptrDeviceIn);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(_ptrDeviceIn);
return -1;
}
WCHAR szDeviceName[MAX_PATH];
const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
// Get the endpoint device's friendly-name
if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
}
_usingInputDeviceIndex = false;
_inputDevice = device;
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::PlayoutIsAvailable(bool& available)
{
available = false;
// Try to initialize the playout side
int32_t res = InitPlayout();
// Cancel effect of initialization
StopPlayout();
if (res != -1)
{
available = true;
}
return 0;
}
// ----------------------------------------------------------------------------
// RecordingIsAvailable
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::RecordingIsAvailable(bool& available)
{
available = false;
// Try to initialize the recording side
int32_t res = InitRecording();
// Cancel effect of initialization
StopRecording();
if (res != -1)
{
available = true;
}
return 0;
}
// ----------------------------------------------------------------------------
// InitPlayout
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::InitPlayout()
{
rtc::CritScope lock(&_critSect);
if (_playing)
{
return -1;
}
if (_playIsInitialized)
{
return 0;
}
if (_ptrDeviceOut == NULL)
{
return -1;
}
// Initialize the speaker (devices might have been added or removed)
if (InitSpeaker() == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
}
// Ensure that the updated rendering endpoint device is valid
if (_ptrDeviceOut == NULL)
{
return -1;
}
if (_builtInAecEnabled && _recIsInitialized)
{
// Ensure the correct render device is configured in case
// InitRecording() was called before InitPlayout().
if (SetDMOProperties() == -1)
{
return -1;
}
}
HRESULT hr = S_OK;
WAVEFORMATEX* pWfxOut = NULL;
WAVEFORMATEX Wfx = WAVEFORMATEX();
WAVEFORMATEX* pWfxClosestMatch = NULL;
// Create COM object with IAudioClient interface.
SAFE_RELEASE(_ptrClientOut);
hr = _ptrDeviceOut->Activate(
__uuidof(IAudioClient),
CLSCTX_ALL,
NULL,
(void**)&_ptrClientOut);
EXIT_ON_ERROR(hr);
// Retrieve the stream format that the audio engine uses for its internal
// processing (mixing) of shared-mode streams.
hr = _ptrClientOut->GetMixFormat(&pWfxOut);
if (SUCCEEDED(hr))
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current rendering mix format:");
// format type
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxOut->wFormatTag, pWfxOut->wFormatTag);
// number of channels (i.e. mono, stereo...)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxOut->nChannels);
// sample rate
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxOut->nSamplesPerSec);
// for buffer estimation
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxOut->nAvgBytesPerSec);
// block size of data
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", 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);
}
// Set wave format
Wfx.wFormatTag = WAVE_FORMAT_PCM;
Wfx.wBitsPerSample = 16;
Wfx.cbSize = 0;
const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
hr = S_FALSE;
// Iterate over frequencies and channels, in order of priority
for (unsigned int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
{
for (unsigned int chan = 0; chan < sizeof(_playChannelsPrioList)/sizeof(_playChannelsPrioList[0]); chan++)
{
Wfx.nChannels = _playChannelsPrioList[chan];
Wfx.nSamplesPerSec = freqs[freq];
Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
// If the method succeeds and the audio endpoint device supports the specified stream format,
// it returns S_OK. If the method succeeds and provides a closest match to the specified format,
// it returns S_FALSE.
hr = _ptrClientOut->IsFormatSupported(
AUDCLNT_SHAREMODE_SHARED,
&Wfx,
&pWfxClosestMatch);
if (hr == S_OK)
{
break;
}
else
{
if (pWfxClosestMatch)
{
LOG(INFO) << "nChannels=" << Wfx.nChannels <<
", nSamplesPerSec=" << Wfx.nSamplesPerSec <<
" is not supported. Closest match: " <<
"nChannels=" << pWfxClosestMatch->nChannels <<
", nSamplesPerSec=" << pWfxClosestMatch->nSamplesPerSec;
CoTaskMemFree(pWfxClosestMatch);
pWfxClosestMatch = NULL;
}
else
{
LOG(INFO) << "nChannels=" << Wfx.nChannels <<
", nSamplesPerSec=" << Wfx.nSamplesPerSec <<
" is not supported. No closest match.";
}
}
}
if (hr == S_OK)
break;
}
// TODO(andrew): what happens in the event of failure in the above loop?
// Is _ptrClientOut->Initialize expected to fail?
// Same in InitRecording().
if (hr == S_OK)
{
_playAudioFrameSize = Wfx.nBlockAlign;
_playBlockSize = Wfx.nSamplesPerSec/100;
_playSampleRate = Wfx.nSamplesPerSec;
_devicePlaySampleRate = Wfx.nSamplesPerSec; // The device itself continues to run at 44.1 kHz.
_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, "_playBlockSize : %d", _playBlockSize);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playChannels : %d", _playChannels);
}
// Create a rendering stream.
//
// ****************************************************************************
// For a shared-mode stream that uses event-driven buffering, the caller must
// set both hnsPeriodicity and hnsBufferDuration to 0. The Initialize method
// determines how large a buffer to allocate based on the scheduling period
// of the audio engine. Although the client's buffer processing thread is
// event driven, the basic buffer management process, as described previously,
// is unaltered.
// Each time the thread awakens, it should call IAudioClient::GetCurrentPadding
// to determine how much data to write to a rendering buffer or read from a capture
// buffer. In contrast to the two buffers that the Initialize method allocates
// for an exclusive-mode stream that uses event-driven buffering, a shared-mode
// stream requires a single buffer.
// ****************************************************************************
//
REFERENCE_TIME hnsBufferDuration = 0; // ask for minimum buffer size (default)
if (_devicePlaySampleRate == 44100)
{
// Ask for a larger buffer size (30ms) when using 44.1kHz as render rate.
// There seems to be a larger risk of underruns for 44.1 compared
// with the default rate (48kHz). When using default, we set the requested
// buffer duration to 0, which sets the buffer to the minimum size
// required by the engine thread. The actual buffer size can then be
// read by GetBufferSize() and it is 20ms on most machines.
hnsBufferDuration = 30*10000;
}
hr = _ptrClientOut->Initialize(
AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
AUDCLNT_STREAMFLAGS_EVENTCALLBACK, // processing of the audio buffer by the client will be event driven
hnsBufferDuration, // requested buffer capacity as a time value (in 100-nanosecond units)
0, // periodicity
&Wfx, // selected wave format
NULL); // session GUID
if (FAILED(hr))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
}
EXIT_ON_ERROR(hr);
if (_ptrAudioBuffer)
{
// Update the audio buffer with the selected parameters
_ptrAudioBuffer->SetPlayoutSampleRate(_playSampleRate);
_ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
}
else
{
// 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");
}
// Get the actual size of the shared (endpoint buffer).
// Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
UINT bufferFrameCount(0);
hr = _ptrClientOut->GetBufferSize(
&bufferFrameCount);
if (SUCCEEDED(hr))
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
bufferFrameCount, bufferFrameCount*_playAudioFrameSize);
}
// Set the event handle that the system signals when an audio buffer is ready
// to be processed by the client.
hr = _ptrClientOut->SetEventHandle(
_hRenderSamplesReadyEvent);
EXIT_ON_ERROR(hr);
// Get an IAudioRenderClient interface.
SAFE_RELEASE(_ptrRenderClient);
hr = _ptrClientOut->GetService(
__uuidof(IAudioRenderClient),
(void**)&_ptrRenderClient);
EXIT_ON_ERROR(hr);
// Mark playout side as initialized
_playIsInitialized = true;
CoTaskMemFree(pWfxOut);
CoTaskMemFree(pWfxClosestMatch);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render side is now initialized");
return 0;
Exit:
_TraceCOMError(hr);
CoTaskMemFree(pWfxOut);
CoTaskMemFree(pWfxClosestMatch);
SAFE_RELEASE(_ptrClientOut);
SAFE_RELEASE(_ptrRenderClient);
return -1;
}
// Capture initialization when the built-in AEC DirectX Media Object (DMO) is
// used. Called from InitRecording(), most of which is skipped over. The DMO
// handles device initialization itself.
// Reference: http://msdn.microsoft.com/en-us/library/ff819492(v=vs.85).aspx
int32_t AudioDeviceWindowsCore::InitRecordingDMO()
{
assert(_builtInAecEnabled);
assert(_dmo != NULL);
if (SetDMOProperties() == -1)
{
return -1;
}
DMO_MEDIA_TYPE mt = {0};
HRESULT hr = MoInitMediaType(&mt, sizeof(WAVEFORMATEX));
if (FAILED(hr))
{
MoFreeMediaType(&mt);
_TraceCOMError(hr);
return -1;
}
mt.majortype = MEDIATYPE_Audio;
mt.subtype = MEDIASUBTYPE_PCM;
mt.formattype = FORMAT_WaveFormatEx;
// Supported formats
// nChannels: 1 (in AEC-only mode)
// nSamplesPerSec: 8000, 11025, 16000, 22050
// wBitsPerSample: 16
WAVEFORMATEX* ptrWav = reinterpret_cast<WAVEFORMATEX*>(mt.pbFormat);
ptrWav->wFormatTag = WAVE_FORMAT_PCM;
ptrWav->nChannels = 1;
// 16000 is the highest we can support with our resampler.
ptrWav->nSamplesPerSec = 16000;
ptrWav->nAvgBytesPerSec = 32000;
ptrWav->nBlockAlign = 2;
ptrWav->wBitsPerSample = 16;
ptrWav->cbSize = 0;
// Set the VoE format equal to the AEC output format.
_recAudioFrameSize = ptrWav->nBlockAlign;
_recSampleRate = ptrWav->nSamplesPerSec;
_recBlockSize = ptrWav->nSamplesPerSec / 100;
_recChannels = ptrWav->nChannels;
// Set the DMO output format parameters.
hr = _dmo->SetOutputType(kAecCaptureStreamIndex, &mt, 0);
MoFreeMediaType(&mt);
if (FAILED(hr))
{
_TraceCOMError(hr);
return -1;
}
if (_ptrAudioBuffer)
{
_ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
_ptrAudioBuffer->SetRecordingChannels(_recChannels);
}
else
{
// Refer to InitRecording() for comments.
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"AudioDeviceBuffer must be attached before streaming can start");
}
_mediaBuffer = new MediaBufferImpl(_recBlockSize * _recAudioFrameSize);
// Optional, but if called, must be after media types are set.
hr = _dmo->AllocateStreamingResources();
if (FAILED(hr))
{
_TraceCOMError(hr);
return -1;
}
_recIsInitialized = true;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Capture side is now initialized");
return 0;
}
// ----------------------------------------------------------------------------
// InitRecording
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::InitRecording()
{
rtc::CritScope lock(&_critSect);
if (_recording)
{
return -1;
}
if (_recIsInitialized)
{
return 0;
}
if (QueryPerformanceFrequency(&_perfCounterFreq) == 0)
{
return -1;
}
_perfCounterFactor = 10000000.0 / (double)_perfCounterFreq.QuadPart;
if (_ptrDeviceIn == NULL)
{
return -1;
}
// Initialize the microphone (devices might have been added or removed)
if (InitMicrophone() == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
}
// Ensure that the updated capturing endpoint device is valid
if (_ptrDeviceIn == NULL)
{
return -1;
}
if (_builtInAecEnabled)
{
// The DMO will configure the capture device.
return InitRecordingDMO();
}
HRESULT hr = S_OK;
WAVEFORMATEX* pWfxIn = NULL;
WAVEFORMATEXTENSIBLE Wfx = WAVEFORMATEXTENSIBLE();
WAVEFORMATEX* pWfxClosestMatch = NULL;
// Create COM object with IAudioClient interface.
SAFE_RELEASE(_ptrClientIn);
hr = _ptrDeviceIn->Activate(
__uuidof(IAudioClient),
CLSCTX_ALL,
NULL,
(void**)&_ptrClientIn);
EXIT_ON_ERROR(hr);
// Retrieve the stream format that the audio engine uses for its internal
// processing (mixing) of shared-mode streams.
hr = _ptrClientIn->GetMixFormat(&pWfxIn);
if (SUCCEEDED(hr))
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current capturing mix format:");
// format type
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxIn->wFormatTag, pWfxIn->wFormatTag);
// number of channels (i.e. mono, stereo...)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxIn->nChannels);
// sample rate
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxIn->nSamplesPerSec);
// for buffer estimation
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxIn->nAvgBytesPerSec);
// block size of data
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", 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);
}
// Set wave format
Wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
Wfx.Format.wBitsPerSample = 16;
Wfx.Format.cbSize = 22;
Wfx.dwChannelMask = 0;
Wfx.Samples.wValidBitsPerSample = Wfx.Format.wBitsPerSample;
Wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
const int freqs[6] = {48000, 44100, 16000, 96000, 32000, 8000};
hr = S_FALSE;
// Iterate over frequencies and channels, in order of priority
for (unsigned int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
{
for (unsigned int chan = 0; chan < sizeof(_recChannelsPrioList)/sizeof(_recChannelsPrioList[0]); chan++)
{
Wfx.Format.nChannels = _recChannelsPrioList[chan];
Wfx.Format.nSamplesPerSec = freqs[freq];
Wfx.Format.nBlockAlign = Wfx.Format.nChannels *
Wfx.Format.wBitsPerSample / 8;
Wfx.Format.nAvgBytesPerSec = Wfx.Format.nSamplesPerSec *
Wfx.Format.nBlockAlign;
// If the method succeeds and the audio endpoint device supports the specified stream format,
// it returns S_OK. If the method succeeds and provides a closest match to the specified format,
// it returns S_FALSE.
hr = _ptrClientIn->IsFormatSupported(
AUDCLNT_SHAREMODE_SHARED,
(WAVEFORMATEX*)&Wfx,
&pWfxClosestMatch);
if (hr == S_OK)
{
break;
}
else
{
if (pWfxClosestMatch)
{
LOG(INFO) << "nChannels=" << Wfx.Format.nChannels <<
", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec <<
" is not supported. Closest match: " <<
"nChannels=" << pWfxClosestMatch->nChannels <<
", nSamplesPerSec=" << pWfxClosestMatch->nSamplesPerSec;
CoTaskMemFree(pWfxClosestMatch);
pWfxClosestMatch = NULL;
}
else
{
LOG(INFO) << "nChannels=" << Wfx.Format.nChannels <<
", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec <<
" is not supported. No closest match.";
}
}
}
if (hr == S_OK)
break;
}
if (hr == S_OK)
{
_recAudioFrameSize = Wfx.Format.nBlockAlign;
_recSampleRate = Wfx.Format.nSamplesPerSec;
_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);
}
// Create a capturing stream.
hr = _ptrClientIn->Initialize(
AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
AUDCLNT_STREAMFLAGS_EVENTCALLBACK | // processing of the audio buffer by the client will be event driven
AUDCLNT_STREAMFLAGS_NOPERSIST, // volume and mute settings for an audio session will not persist across system restarts
0, // required for event-driven shared mode
0, // periodicity
(WAVEFORMATEX*)&Wfx, // selected wave format
NULL); // session GUID
if (hr != S_OK)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
}
EXIT_ON_ERROR(hr);
if (_ptrAudioBuffer)
{
// Update the audio buffer with the selected parameters
_ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
_ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
}
else
{
// 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");
}
// Get the actual size of the shared (endpoint buffer).
// Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
UINT bufferFrameCount(0);
hr = _ptrClientIn->GetBufferSize(
&bufferFrameCount);
if (SUCCEEDED(hr))
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
bufferFrameCount, bufferFrameCount*_recAudioFrameSize);
}
// Set the event handle that the system signals when an audio buffer is ready
// to be processed by the client.
hr = _ptrClientIn->SetEventHandle(
_hCaptureSamplesReadyEvent);
EXIT_ON_ERROR(hr);
// Get an IAudioCaptureClient interface.
SAFE_RELEASE(_ptrCaptureClient);
hr = _ptrClientIn->GetService(
__uuidof(IAudioCaptureClient),
(void**)&_ptrCaptureClient);
EXIT_ON_ERROR(hr);
// Mark capture side as initialized
_recIsInitialized = true;
CoTaskMemFree(pWfxIn);
CoTaskMemFree(pWfxClosestMatch);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "capture side is now initialized");
return 0;
Exit:
_TraceCOMError(hr);
CoTaskMemFree(pWfxIn);
CoTaskMemFree(pWfxClosestMatch);
SAFE_RELEASE(_ptrClientIn);
SAFE_RELEASE(_ptrCaptureClient);
return -1;
}
// ----------------------------------------------------------------------------
// StartRecording
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StartRecording()
{
if (!_recIsInitialized)
{
return -1;
}
if (_hRecThread != NULL)
{
return 0;
}
if (_recording)
{
return 0;
}
{
rtc::CritScope critScoped(&_critSect);
// Create thread which will drive the capturing
LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
if (_builtInAecEnabled)
{
// Redirect to the DMO polling method.
lpStartAddress = WSAPICaptureThreadPollDMO;
if (!_playing)
{
// 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");
return -1;
}
}
assert(_hRecThread == NULL);
_hRecThread = CreateThread(NULL,
0,
lpStartAddress,
this,
0,
NULL);
if (_hRecThread == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"failed to create the recording thread");
return -1;
}
// Set thread priority to highest possible
SetThreadPriority(_hRecThread, THREAD_PRIORITY_TIME_CRITICAL);
assert(_hGetCaptureVolumeThread == NULL);
_hGetCaptureVolumeThread = CreateThread(NULL,
0,
GetCaptureVolumeThread,
this,
0,
NULL);
if (_hGetCaptureVolumeThread == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" failed to create the volume getter thread");
return -1;
}
assert(_hSetCaptureVolumeThread == NULL);
_hSetCaptureVolumeThread = CreateThread(NULL,
0,
SetCaptureVolumeThread,
this,
0,
NULL);
if (_hSetCaptureVolumeThread == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" failed to create the volume setter thread");
return -1;
}
} // critScoped
DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
if (ret != WAIT_OBJECT_0)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"capturing did not start up properly");
return -1;
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"capture audio stream has now started...");
_avgCPULoad = 0.0f;
_playAcc = 0;
_recording = true;
return 0;
}
// ----------------------------------------------------------------------------
// StopRecording
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StopRecording()
{
int32_t err = 0;
if (!_recIsInitialized)
{
return 0;
}
_Lock();
if (_hRecThread == NULL)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"no capturing stream is active => close down WASAPI only");
SAFE_RELEASE(_ptrClientIn);
SAFE_RELEASE(_ptrCaptureClient);
_recIsInitialized = false;
_recording = false;
_UnLock();
return 0;
}
// Stop the driving thread...
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"closing down the webrtc_core_audio_capture_thread...");
// Manual-reset event; it will remain signalled to stop all capture threads.
SetEvent(_hShutdownCaptureEvent);
_UnLock();
DWORD ret = WaitForSingleObject(_hRecThread, 2000);
if (ret != WAIT_OBJECT_0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"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");
}
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");
err = -1;
}
else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" 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");
err = -1;
}
else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" volume setter thread is now closed");
}
_Lock();
ResetEvent(_hShutdownCaptureEvent); // Must be manually reset.
// Ensure that the thread has released these interfaces properly.
assert(err == -1 || _ptrClientIn == NULL);
assert(err == -1 || _ptrCaptureClient == NULL);
_recIsInitialized = false;
_recording = false;
// These will create thread leaks in the result of an error,
// but we can at least resume the call.
CloseHandle(_hRecThread);
_hRecThread = NULL;
CloseHandle(_hGetCaptureVolumeThread);
_hGetCaptureVolumeThread = NULL;
CloseHandle(_hSetCaptureVolumeThread);
_hSetCaptureVolumeThread = NULL;
if (_builtInAecEnabled)
{
assert(_dmo != NULL);
// This is necessary. Otherwise the DMO can generate garbage render
// audio even after rendering has stopped.
HRESULT hr = _dmo->FreeStreamingResources();
if (FAILED(hr))
{
_TraceCOMError(hr);
err = -1;
}
}
// Reset the recording delay value.
_sndCardRecDelay = 0;
_UnLock();
return err;
}
// ----------------------------------------------------------------------------
// RecordingIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::RecordingIsInitialized() const
{
return (_recIsInitialized);
}
// ----------------------------------------------------------------------------
// Recording
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::Recording() const
{
return (_recording);
}
// ----------------------------------------------------------------------------
// PlayoutIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::PlayoutIsInitialized() const
{
return (_playIsInitialized);
}
// ----------------------------------------------------------------------------
// StartPlayout
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StartPlayout()
{
if (!_playIsInitialized)
{
return -1;
}
if (_hPlayThread != NULL)
{
return 0;
}
if (_playing)
{
return 0;
}
{
rtc::CritScope critScoped(&_critSect);
// Create thread which will drive the rendering.
assert(_hPlayThread == NULL);
_hPlayThread = CreateThread(
NULL,
0,
WSAPIRenderThread,
this,
0,
NULL);
if (_hPlayThread == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"failed to create the playout thread");
return -1;
}
// Set thread priority to highest possible.
SetThreadPriority(_hPlayThread, THREAD_PRIORITY_TIME_CRITICAL);
} // critScoped
DWORD ret = WaitForSingleObject(_hRenderStartedEvent, 1000);
if (ret != WAIT_OBJECT_0)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"rendering did not start up properly");
return -1;
}
_playing = true;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"rendering audio stream has now started...");
return 0;
}
// ----------------------------------------------------------------------------
// StopPlayout
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::StopPlayout()
{
if (!_playIsInitialized)
{
return 0;
}
{
rtc::CritScope critScoped(&_critSect) ;
if (_hPlayThread == NULL)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"no rendering stream is active => close down WASAPI only");
SAFE_RELEASE(_ptrClientOut);
SAFE_RELEASE(_ptrRenderClient);
_playIsInitialized = false;
_playing = false;
return 0;
}
// stop the driving thread...
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"closing down the webrtc_core_audio_render_thread...");
SetEvent(_hShutdownRenderEvent);
} // critScoped
DWORD ret = WaitForSingleObject(_hPlayThread, 2000);
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");
CloseHandle(_hPlayThread);
_hPlayThread = NULL;
_playIsInitialized = false;
_playing = false;
return -1;
}
{
rtc::CritScope critScoped(&_critSect);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"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(),
// this event might be caught by the new render thread within same VoE instance.
ResetEvent(_hShutdownRenderEvent);
SAFE_RELEASE(_ptrClientOut);
SAFE_RELEASE(_ptrRenderClient);
_playIsInitialized = false;
_playing = false;
CloseHandle(_hPlayThread);
_hPlayThread = NULL;
if (_builtInAecEnabled && _recording)
{
// The DMO won't provide us captured output data unless we
// give it render data to process.
//
// 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");
}
// Reset the playout delay value.
_sndCardPlayDelay = 0;
} // critScoped
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDelay
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const
{
rtc::CritScope critScoped(&_critSect);
delayMS = static_cast<uint16_t>(_sndCardPlayDelay);
return 0;
}
// ----------------------------------------------------------------------------
// RecordingDelay
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::RecordingDelay(uint16_t& delayMS) const
{
rtc::CritScope critScoped(&_critSect);
delayMS = static_cast<uint16_t>(_sndCardRecDelay);
return 0;
}
// ----------------------------------------------------------------------------
// Playing
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::Playing() const
{
return (_playing);
}
// ----------------------------------------------------------------------------
// SetPlayoutBuffer
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, uint16_t sizeMS)
{
rtc::CritScope lock(&_critSect);
_playBufType = type;
if (type == AudioDeviceModule::kFixedBufferSize)
{
_playBufDelayFixed = sizeMS;
}
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutBuffer
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::PlayoutBuffer(AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const
{
rtc::CritScope lock(&_critSect);
type = _playBufType;
if (type == AudioDeviceModule::kFixedBufferSize)
{
sizeMS = _playBufDelayFixed;
}
else
{
// Use same value as for PlayoutDelay
sizeMS = static_cast<uint16_t>(_sndCardPlayDelay);
}
return 0;
}
// ----------------------------------------------------------------------------
// CPULoad
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::CPULoad(uint16_t& load) const
{
load = static_cast<uint16_t> (100*_avgCPULoad);
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutWarning
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::PlayoutWarning() const
{
return ( _playWarning > 0);
}
// ----------------------------------------------------------------------------
// PlayoutError
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::PlayoutError() const
{
return ( _playError > 0);
}
// ----------------------------------------------------------------------------
// RecordingWarning
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::RecordingWarning() const
{
return ( _recWarning > 0);
}
// ----------------------------------------------------------------------------
// RecordingError
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsCore::RecordingError() const
{
return ( _recError > 0);
}
// ----------------------------------------------------------------------------
// ClearPlayoutWarning
// ----------------------------------------------------------------------------
void AudioDeviceWindowsCore::ClearPlayoutWarning()
{
_playWarning = 0;
}
// ----------------------------------------------------------------------------
// ClearPlayoutError
// ----------------------------------------------------------------------------
void AudioDeviceWindowsCore::ClearPlayoutError()
{
_playError = 0;
}
// ----------------------------------------------------------------------------
// ClearRecordingWarning
// ----------------------------------------------------------------------------
void AudioDeviceWindowsCore::ClearRecordingWarning()
{
_recWarning = 0;
}
// ----------------------------------------------------------------------------
// ClearRecordingError
// ----------------------------------------------------------------------------
void AudioDeviceWindowsCore::ClearRecordingError()
{
_recError = 0;
}
// ============================================================================
// Private Methods
// ============================================================================
// ----------------------------------------------------------------------------
// [static] WSAPIRenderThread
// ----------------------------------------------------------------------------
DWORD WINAPI AudioDeviceWindowsCore::WSAPIRenderThread(LPVOID context)
{
return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
DoRenderThread();
}
// ----------------------------------------------------------------------------
// [static] WSAPICaptureThread
// ----------------------------------------------------------------------------
DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThread(LPVOID context)
{
return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
DoCaptureThread();
}
DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThreadPollDMO(LPVOID context)
{
return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
DoCaptureThreadPollDMO();
}
DWORD WINAPI AudioDeviceWindowsCore::GetCaptureVolumeThread(LPVOID context)
{
return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
DoGetCaptureVolumeThread();
}
DWORD WINAPI AudioDeviceWindowsCore::SetCaptureVolumeThread(LPVOID context)
{
return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
DoSetCaptureVolumeThread();
}
DWORD AudioDeviceWindowsCore::DoGetCaptureVolumeThread()
{
HANDLE waitObject = _hShutdownCaptureEvent;
while (1)
{
if (AGC())
{
uint32_t currentMicLevel = 0;
if (MicrophoneVolume(currentMicLevel) == 0)
{
// This doesn't set the system volume, just stores it.
_Lock();
if (_ptrAudioBuffer)
{
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
_UnLock();
}
}
DWORD waitResult = WaitForSingleObject(waitObject,
GET_MIC_VOLUME_INTERVAL_MS);
switch (waitResult)
{
case WAIT_OBJECT_0: // _hShutdownCaptureEvent
return 0;
case WAIT_TIMEOUT: // timeout notification
break;
default: // unexpected error
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" unknown wait termination on get volume thread");
return 1;
}
}
}
DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread()
{
HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hSetCaptureVolumeEvent};
while (1)
{
DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
switch (waitResult)
{
case WAIT_OBJECT_0: // _hShutdownCaptureEvent
return 0;
case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
break;
default: // unexpected error
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" unknown wait termination on set volume thread");
return 1;
}
_Lock();
uint32_t newMicLevel = _newMicLevel;
_UnLock();
if (SetMicrophoneVolume(newMicLevel) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" the required modification of the microphone volume failed");
}
}
}
// ----------------------------------------------------------------------------
// DoRenderThread
// ----------------------------------------------------------------------------
DWORD AudioDeviceWindowsCore::DoRenderThread()
{
bool keepPlaying = true;
HANDLE waitArray[2] = {_hShutdownRenderEvent, _hRenderSamplesReadyEvent};
HRESULT hr = S_OK;
HANDLE hMmTask = NULL;
LARGE_INTEGER t1;
LARGE_INTEGER t2;
int32_t time(0);
// 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");
return 1;
}
rtc::SetCurrentThreadName("webrtc_core_audio_render_thread");
// Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority.
//
if (_winSupportAvrt)
{
DWORD taskIndex(0);
hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
if (hMmTask)
{
if (FALSE == _PAvSetMmThreadPriority(hMmTask, AVRT_PRIORITY_CRITICAL))
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to boost play-thread using MMCSS");
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render thread is now registered with MMCSS (taskIndex=%d)", taskIndex);
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to enable MMCSS on render thread (err=%d)", GetLastError());
_TraceCOMError(GetLastError());
}
}
_Lock();
IAudioClock* clock = NULL;
// 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.
//
UINT32 bufferLength = 0;
hr = _ptrClientOut->GetBufferSize(&bufferLength);
EXIT_ON_ERROR(hr);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] size of buffer : %u", 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));
// Get the length of the periodic interval separating successive processing passes by
// the audio engine on the data in the endpoint buffer.
//
// The period between processing passes by the audio engine is fixed for a particular
// audio endpoint device and represents the smallest processing quantum for the audio engine.
// This period plus the stream latency between the buffer and endpoint device represents
// the minimum possible latency that an audio application can achieve.
// Typical value: 100000 <=> 0.01 sec = 10ms.
//
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));
// Derive initial rendering delay.
// Example: 10*(960/480) + 15 = 20 + 15 = 35ms
//
int playout_delay = 10 * (bufferLength / _playBlockSize) +
(int)((latency + devPeriod) / 10000);
_sndCardPlayDelay = playout_delay;
_writtenSamples = 0;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"[REND] initial delay : %u", playout_delay);
double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_devicePlayBlockSize);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
// Before starting the stream, fill the rendering buffer with silence.
//
BYTE *pData = NULL;
hr = _ptrRenderClient->GetBuffer(bufferLength, &pData);
EXIT_ON_ERROR(hr);
hr = _ptrRenderClient->ReleaseBuffer(bufferLength, AUDCLNT_BUFFERFLAGS_SILENT);
EXIT_ON_ERROR(hr);
_writtenSamples += bufferLength;
hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
if (FAILED(hr)) {
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"failed to get IAudioClock interface from the IAudioClient");
}
// Start up the rendering audio stream.
hr = _ptrClientOut->Start();
EXIT_ON_ERROR(hr);
_UnLock();
// Set event which will ensure that the calling thread modifies the playing state to true.
//
SetEvent(_hRenderStartedEvent);
// >> ------------------ THREAD LOOP ------------------
while (keepPlaying)
{
// Wait for a render notification event or a shutdown event
DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
switch (waitResult)
{
case WAIT_OBJECT_0 + 0: // _hShutdownRenderEvent
keepPlaying = false;
break;
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");
goto Exit;
default: // unexpected error
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on render side");
goto Exit;
}
while (keepPlaying)
{
_Lock();
// Sanity check to ensure that essential states are not modified
// during the unlocked period.
if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
{
_UnLock();
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"output state has been modified during unlocked period");
goto Exit;
}
// Get the number of frames of padding (queued up to play) in the endpoint buffer.
UINT32 padding = 0;
hr = _ptrClientOut->GetCurrentPadding(&padding);
EXIT_ON_ERROR(hr);
// 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 < _playBlockSize)
{
// Not enough space in render buffer to store next render packet.
_UnLock();
break;
}
// Write n*10ms buffers to the render buffer
const uint32_t n10msBuffers = (framesAvailable / _playBlockSize);
for (uint32_t n = 0; n < n10msBuffers; n++)
{
// Get pointer (i.e., grab the buffer) to next space in the shared render buffer.
hr = _ptrRenderClient->GetBuffer(_playBlockSize, &pData);
EXIT_ON_ERROR(hr);
QueryPerformanceCounter(&t1); // measure time: START
if (_ptrAudioBuffer)
{
// Request data to be played out (#bytes = _playBlockSize*_audioFrameSize)
_UnLock();
int32_t nSamples =
_ptrAudioBuffer->RequestPlayoutData(_playBlockSize);
_Lock();
if (nSamples == -1)
{
_UnLock();
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"failed to read data from render client");
goto Exit;
}
// Sanity check to ensure that essential states are not modified during the unlocked period
if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
{
_UnLock();
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
goto Exit;
}
if (nSamples != static_cast<int32_t>(_playBlockSize))
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "nSamples(%d) != _playBlockSize(%d)", nSamples, _playBlockSize);
}
// Get the actual (stored) data
nSamples = _ptrAudioBuffer->GetPlayoutData((int8_t*)pData);
}
QueryPerformanceCounter(&t2); // measure time: STOP
time = (int)(t2.QuadPart-t1.QuadPart);
_playAcc += time;
DWORD dwFlags(0);
hr = _ptrRenderClient->ReleaseBuffer(_playBlockSize, dwFlags);
// See http://msdn.microsoft.com/en-us/library/dd316605(VS.85).aspx
// for more details regarding AUDCLNT_E_DEVICE_INVALIDATED.
EXIT_ON_ERROR(hr);
_writtenSamples += _playBlockSize;
}
// Check the current delay on the playout side.
if (clock) {
UINT64 pos = 0;
UINT64 freq = 1;
clock->GetPosition(&pos, NULL);
clock->GetFrequency(&freq);
playout_delay = ROUND((double(_writtenSamples) /
_devicePlaySampleRate - double(pos) / freq) * 1000.0);
_sndCardPlayDelay = playout_delay;
}
_UnLock();
}
}
// ------------------ THREAD LOOP ------------------ <<
SleepMs(static_cast<DWORD>(endpointBufferSizeMS+0.5));
hr = _ptrClientOut->Stop();
Exit:
SAFE_RELEASE(clock);
if (FAILED(hr))
{
_ptrClientOut->Stop();
_UnLock();
_TraceCOMError(hr);
}
if (_winSupportAvrt)
{
if (NULL != hMmTask)
{
_PAvRevertMmThreadCharacteristics(hMmTask);
}
}
_Lock();
if (keepPlaying)
{
if (_ptrClientOut != NULL)
{
hr = _ptrClientOut->Stop();
if (FAILED(hr))
{
_TraceCOMError(hr);
}
hr = _ptrClientOut->Reset();
if (FAILED(hr))
{
_TraceCOMError(hr);
}
}
// Trigger callback from module process thread
_playError = 1;
WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: rendering thread has ended pre-maturely");
}
else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Rendering thread is now terminated properly");
}
_UnLock();
return (DWORD)hr;
}
DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority()
{
_hMmTask = NULL;
rtc::SetCurrentThreadName("webrtc_core_audio_capture_thread");
// Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
// priority.
if (_winSupportAvrt)
{
DWORD taskIndex(0);
_hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
if (_hMmTask)
{
if (!_PAvSetMmThreadPriority(_hMmTask, AVRT_PRIORITY_CRITICAL))
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"failed to boost rec-thread using MMCSS");
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"capture thread is now registered with MMCSS (taskIndex=%d)",
taskIndex);
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"failed to enable MMCSS on capture thread (err=%d)",
GetLastError());
_TraceCOMError(GetLastError());
}
}
return S_OK;
}
void AudioDeviceWindowsCore::RevertCaptureThreadPriority()
{
if (_winSupportAvrt)
{
if (NULL != _hMmTask)
{
_PAvRevertMmThreadCharacteristics(_hMmTask);
}
}
_hMmTask = NULL;
}
DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
{
assert(_mediaBuffer != NULL);
bool keepRecording = true;
// 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");
return 1;
}
HRESULT hr = InitCaptureThreadPriority();
if (FAILED(hr))
{
return hr;
}
// Set event which will ensure that the calling thread modifies the
// recording state to true.
SetEvent(_hCaptureStartedEvent);
// >> ---------------------------- THREAD LOOP ----------------------------
while (keepRecording)
{
// Poll the DMO every 5 ms.
// (The same interval used in the Wave implementation.)
DWORD waitResult = WaitForSingleObject(_hShutdownCaptureEvent, 5);
switch (waitResult)
{
case WAIT_OBJECT_0: // _hShutdownCaptureEvent
keepRecording = false;
break;
case WAIT_TIMEOUT: // timeout notification
break;
default: // unexpected error
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"Unknown wait termination on capture side");
hr = -1; // To signal an error callback.
keepRecording = false;
break;
}
while (keepRecording)
{
rtc::CritScope critScoped(&_critSect);
DWORD dwStatus = 0;
{
DMO_OUTPUT_DATA_BUFFER dmoBuffer = {0};
dmoBuffer.pBuffer = _mediaBuffer;
dmoBuffer.pBuffer->AddRef();
// Poll the DMO for AEC processed capture data. The DMO will
// copy available data to |dmoBuffer|, and should only return
// 10 ms frames. The value of |dwStatus| should be ignored.
hr = _dmo->ProcessOutput(0, 1, &dmoBuffer, &dwStatus);
SAFE_RELEASE(dmoBuffer.pBuffer);
dwStatus = dmoBuffer.dwStatus;
}
if (FAILED(hr))
{
_TraceCOMError(hr);
keepRecording = false;
assert(false);
break;
}
ULONG bytesProduced = 0;
BYTE* data;
// Get a pointer to the data buffer. This should be valid until
// the next call to ProcessOutput.
hr = _mediaBuffer->GetBufferAndLength(&data, &bytesProduced);
if (FAILED(hr))
{
_TraceCOMError(hr);
keepRecording = false;
assert(false);
break;
}
// TODO(andrew): handle AGC.
if (bytesProduced > 0)
{
const int kSamplesProduced = bytesProduced / _recAudioFrameSize;
// TODO(andrew): verify that this is always satisfied. It might
// be that ProcessOutput will try to return more than 10 ms if
// we fail to call it frequently enough.
assert(kSamplesProduced == static_cast<int>(_recBlockSize));
assert(sizeof(BYTE) == sizeof(int8_t));
_ptrAudioBuffer->SetRecordedBuffer(
reinterpret_cast<int8_t*>(data),
kSamplesProduced);
_ptrAudioBuffer->SetVQEData(0, 0, 0);
_UnLock(); // Release lock while making the callback.
_ptrAudioBuffer->DeliverRecordedData();
_Lock();
}
// Reset length to indicate buffer availability.
hr = _mediaBuffer->SetLength(0);
if (FAILED(hr))
{
_TraceCOMError(hr);
keepRecording = false;
assert(false);
break;
}
if (!(dwStatus & DMO_OUTPUT_DATA_BUFFERF_INCOMPLETE))
{
// The DMO cannot currently produce more data. This is the
// normal case; otherwise it means the DMO had more than 10 ms
// of data available and ProcessOutput should be called again.
break;
}
}
}
// ---------------------------- THREAD LOOP ---------------------------- <<
RevertCaptureThreadPriority();
if (FAILED(hr))
{
// Trigger callback from module process thread
_recError = 1;
WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
"kRecordingError message posted: capturing thread has ended "
"prematurely");
}
else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Capturing thread is now terminated properly");
}
return hr;
}
// ----------------------------------------------------------------------------
// DoCaptureThread
// ----------------------------------------------------------------------------
DWORD AudioDeviceWindowsCore::DoCaptureThread()
{
bool keepRecording = true;
HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hCaptureSamplesReadyEvent};
HRESULT hr = S_OK;
LARGE_INTEGER t1;
LARGE_INTEGER t2;
int32_t time(0);
BYTE* syncBuffer = NULL;
UINT32 syncBufIndex = 0;
_readSamples = 0;
// 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");
return 1;
}
hr = InitCaptureThreadPriority();
if (FAILED(hr))
{
return hr;
}
_Lock();
// Get size of capturing buffer (length is expressed as the number of audio frames the buffer can hold).
// This value is fixed during the capturing session.
//
UINT32 bufferLength = 0;
if (_ptrClientIn == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"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);
// Allocate memory for sync buffer.
// It is used for compensation between native 44.1 and internal 44.0 and
// for cases when the capture buffer is larger than 10ms.
//
const UINT32 syncBufferSize = 2*(bufferLength * _recAudioFrameSize);
syncBuffer = new BYTE[syncBufferSize];
if (syncBuffer == NULL)
{
return (DWORD)E_POINTER;
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] size of sync buffer : %u [bytes]", syncBufferSize);
// 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));
// Get the length of the periodic interval separating successive processing passes by
// the audio engine on the data in the endpoint buffer.
//
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));
double extraDelayMS = (double)((latency + devPeriod) / 10000.0);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] extraDelayMS : %3.2f", extraDelayMS);
double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_recBlockSize);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
// Start up the capturing stream.
//
hr = _ptrClientIn->Start();
EXIT_ON_ERROR(hr);
_UnLock();
// Set event which will ensure that the calling thread modifies the recording state to true.
//
SetEvent(_hCaptureStartedEvent);
// >> ---------------------------- THREAD LOOP ----------------------------
while (keepRecording)
{
// Wait for a capture notification event or a shutdown event
DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
switch (waitResult)
{
case WAIT_OBJECT_0 + 0: // _hShutdownCaptureEvent
keepRecording = false;
break;
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");
goto Exit;
default: // unexpected error
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on capture side");
goto Exit;
}
while (keepRecording)
{
BYTE *pData = 0;
UINT32 framesAvailable = 0;
DWORD flags = 0;
UINT64 recTime = 0;
UINT64 recPos = 0;
_Lock();
// Sanity check to ensure that essential states are not modified
// during the unlocked period.
if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
{
_UnLock();
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"input state has been modified during unlocked period");
goto Exit;
}
// Find out how much capture data is available
//
hr = _ptrCaptureClient->GetBuffer(&pData, // packet which is ready to be read by used
&framesAvailable, // #frames in the captured packet (can be zero)
&flags, // support flags (check)
&recPos, // device position of first audio frame in data packet
&recTime); // value of performance counter at the time of recording the first audio frame
if (SUCCEEDED(hr))
{
if (AUDCLNT_S_BUFFER_EMPTY == hr)
{
// Buffer was empty => start waiting for a new capture notification event
_UnLock();
break;
}
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");
pData = NULL;
}
assert(framesAvailable != 0);
if (pData)
{
CopyMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], pData, framesAvailable*_recAudioFrameSize);
}
else
{
ZeroMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], framesAvailable*_recAudioFrameSize);
}
assert(syncBufferSize >= (syncBufIndex*_recAudioFrameSize)+framesAvailable*_recAudioFrameSize);
// Release the capture buffer
//
hr = _ptrCaptureClient->ReleaseBuffer(framesAvailable);
EXIT_ON_ERROR(hr);
_readSamples += framesAvailable;
syncBufIndex += framesAvailable;
QueryPerformanceCounter(&t1);
// Get the current recording and playout delay.
uint32_t sndCardRecDelay = (uint32_t)
(((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime)
/ 10000) + (10*syncBufIndex) / _recBlockSize - 10);
uint32_t sndCardPlayDelay =
static_cast<uint32_t>(_sndCardPlayDelay);
_sndCardRecDelay = sndCardRecDelay;
while (syncBufIndex >= _recBlockSize)
{
if (_ptrAudioBuffer)
{
_ptrAudioBuffer->SetRecordedBuffer((const int8_t*)syncBuffer, _recBlockSize);
_ptrAudioBuffer->SetVQEData(sndCardPlayDelay,
sndCardRecDelay,
0);
_ptrAudioBuffer->SetTypingStatus(KeyPressed());
QueryPerformanceCounter(&t1); // measure time: START
_UnLock(); // release lock while making the callback
_ptrAudioBuffer->DeliverRecordedData();
_Lock(); // restore the lock
QueryPerformanceCounter(&t2); // measure time: STOP
// Measure "average CPU load".
// Basically what we do here is to measure how many percent of our 10ms period
// is used for encoding and decoding. This value shuld be used as a warning indicator
// only and not seen as an absolute value. Running at ~100% will lead to bad QoS.
time = (int)(t2.QuadPart - t1.QuadPart);
_avgCPULoad = (float)(_avgCPULoad*.99 + (time + _playAcc) / (double)(_perfCounterFreq.QuadPart));
_playAcc = 0;
// Sanity check to ensure that essential states are not modified during the unlocked period
if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
{
_UnLock();
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
goto Exit;
}
}
// store remaining data which was not able to deliver as 10ms segment
MoveMemory(&syncBuffer[0], &syncBuffer[_recBlockSize*_recAudioFrameSize], (syncBufIndex-_recBlockSize)*_recAudioFrameSize);
syncBufIndex -= _recBlockSize;
sndCardRecDelay -= 10;
}
if (_AGC)
{
uint32_t newMicLevel = _ptrAudioBuffer->NewMicLevel();
if (newMicLevel != 0)
{
// 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);
// We store this outside of the audio buffer to avoid
// having it overwritten by the getter thread.
_newMicLevel = newMicLevel;
SetEvent(_hSetCaptureVolumeEvent);
}
}
}
else
{
// If GetBuffer returns AUDCLNT_E_BUFFER_ERROR, the thread consuming the audio samples
// must wait for the next processing pass. The client might benefit from keeping a count
// 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);
goto Exit;
}
_UnLock();
}
}
// ---------------------------- THREAD LOOP ---------------------------- <<
if (_ptrClientIn)
{
hr = _ptrClientIn->Stop();
}
Exit:
if (FAILED(hr))
{
_ptrClientIn->Stop();
_UnLock();
_TraceCOMError(hr);
}
RevertCaptureThreadPriority();
_Lock();
if (keepRecording)
{
if (_ptrClientIn != NULL)
{
hr = _ptrClientIn->Stop();
if (FAILED(hr))
{
_TraceCOMError(hr);
}
hr = _ptrClientIn->Reset();
if (FAILED(hr))
{
_TraceCOMError(hr);
}
}
// Trigger callback from module process thread
_recError = 1;
WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: capturing thread has ended pre-maturely");
}
else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Capturing thread is now terminated properly");
}
SAFE_RELEASE(_ptrClientIn);
SAFE_RELEASE(_ptrCaptureClient);
_UnLock();
if (syncBuffer)
{
delete [] syncBuffer;
}
return (DWORD)hr;
}
int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable)
{
if (_recIsInitialized)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"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");
return -1;
}
_builtInAecEnabled = enable;
return 0;
}
int AudioDeviceWindowsCore::SetDMOProperties()
{
HRESULT hr = S_OK;
assert(_dmo != NULL);
rtc::scoped_refptr<IPropertyStore> ps;
{
IPropertyStore* ptrPS = NULL;
hr = _dmo->QueryInterface(IID_IPropertyStore,
reinterpret_cast<void**>(&ptrPS));
if (FAILED(hr) || ptrPS == NULL)
{
_TraceCOMError(hr);
return -1;
}
ps = ptrPS;
SAFE_RELEASE(ptrPS);
}
// Set the AEC system mode.
// SINGLE_CHANNEL_AEC - AEC processing only.
if (SetVtI4Property(ps,
MFPKEY_WMAAECMA_SYSTEM_MODE,
SINGLE_CHANNEL_AEC))
{
return -1;
}
// Set the AEC source mode.
// VARIANT_TRUE - Source mode (we poll the AEC for captured data).
if (SetBoolProperty(ps,
MFPKEY_WMAAECMA_DMO_SOURCE_MODE,
VARIANT_TRUE) == -1)
{
return -1;
}
// Enable the feature mode.
// This lets us override all the default processing settings below.
if (SetBoolProperty(ps,
MFPKEY_WMAAECMA_FEATURE_MODE,
VARIANT_TRUE) == -1)
{
return -1;
}
// Disable analog AGC (default enabled).
if (SetBoolProperty(ps,
MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER,
VARIANT_FALSE) == -1)
{
return -1;
}
// Disable noise suppression (default enabled).
// 0 - Disabled, 1 - Enabled
if (SetVtI4Property(ps,
MFPKEY_WMAAECMA_FEATR_NS,
0) == -1)
{
return -1;
}
// Relevant parameters to leave at default settings:
// MFPKEY_WMAAECMA_FEATR_AGC - Digital AGC (disabled).
// MFPKEY_WMAAECMA_FEATR_CENTER_CLIP - AEC center clipping (enabled).
// MFPKEY_WMAAECMA_FEATR_ECHO_LENGTH - Filter length (256 ms).
// TODO(andrew): investigate decresing the length to 128 ms.
// MFPKEY_WMAAECMA_FEATR_FRAME_SIZE - Frame size (0).
// 0 is automatic; defaults to 160 samples (or 10 ms frames at the
// selected 16 kHz) as long as mic array processing is disabled.
// MFPKEY_WMAAECMA_FEATR_NOISE_FILL - Comfort noise (enabled).
// MFPKEY_WMAAECMA_FEATR_VAD - VAD (disabled).
// Set the devices selected by VoE. If using a default device, we need to
// search for the device index.
int inDevIndex = _inputDeviceIndex;
int outDevIndex = _outputDeviceIndex;
if (!_usingInputDeviceIndex)
{
ERole role = eCommunications;
if (_inputDevice == AudioDeviceModule::kDefaultDevice)
{
role = eConsole;
}
if (_GetDefaultDeviceIndex(eCapture, role, &inDevIndex) == -1)
{
return -1;
}
}
if (!_usingOutputDeviceIndex)
{
ERole role = eCommunications;
if (_outputDevice == AudioDeviceModule::kDefaultDevice)
{
role = eConsole;
}
if (_GetDefaultDeviceIndex(eRender, role, &outDevIndex) == -1)
{
return -1;
}
}
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);
if (SetVtI4Property(ps,
MFPKEY_WMAAECMA_DEVICE_INDEXES,
devIndex) == -1)
{
return -1;
}
return 0;
}
int AudioDeviceWindowsCore::SetBoolProperty(IPropertyStore* ptrPS,
REFPROPERTYKEY key,
VARIANT_BOOL value)
{
PROPVARIANT pv;
PropVariantInit(&pv);
pv.vt = VT_BOOL;
pv.boolVal = value;
HRESULT hr = ptrPS->SetValue(key, pv);
PropVariantClear(&pv);
if (FAILED(hr))
{
_TraceCOMError(hr);
return -1;
}
return 0;
}
int AudioDeviceWindowsCore::SetVtI4Property(IPropertyStore* ptrPS,
REFPROPERTYKEY key,
LONG value)
{
PROPVARIANT pv;
PropVariantInit(&pv);
pv.vt = VT_I4;
pv.lVal = value;
HRESULT hr = ptrPS->SetValue(key, pv);
PropVariantClear(&pv);
if (FAILED(hr))
{
_TraceCOMError(hr);
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// _RefreshDeviceList
//
// Creates a new list of endpoint rendering or capture devices after
// deleting any previously created (and possibly out-of-date) list of
// such devices.
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr = S_OK;
IMMDeviceCollection *pCollection = NULL;
assert(dir == eRender || dir == eCapture);
assert(_ptrEnumerator != NULL);
// Create a fresh list of devices using the specified direction
hr = _ptrEnumerator->EnumAudioEndpoints(
dir,
DEVICE_STATE_ACTIVE,
&pCollection);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(pCollection);
return -1;
}
if (dir == eRender)
{
SAFE_RELEASE(_ptrRenderCollection);
_ptrRenderCollection = pCollection;
}
else
{
SAFE_RELEASE(_ptrCaptureCollection);
_ptrCaptureCollection = pCollection;
}
return 0;
}
// ----------------------------------------------------------------------------
// _DeviceListCount
//
// Gets a count of the endpoint rendering or capture devices in the
// current list of such devices.
// ----------------------------------------------------------------------------
int16_t AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr = S_OK;
UINT count = 0;
assert(eRender == dir || eCapture == dir);
if (eRender == dir && NULL != _ptrRenderCollection)
{
hr = _ptrRenderCollection->GetCount(&count);
}
else if (NULL != _ptrCaptureCollection)
{
hr = _ptrCaptureCollection->GetCount(&count);
}
if (FAILED(hr))
{
_TraceCOMError(hr);
return -1;
}
return static_cast<int16_t> (count);
}
// ----------------------------------------------------------------------------
// _GetListDeviceName
//
// Gets the friendly name of an endpoint rendering or capture device
// from the current list of such devices. The caller uses an index
// into the list to identify the device.
//
// Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
// in _RefreshDeviceList().
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
assert(dir == eRender || dir == eCapture);
if (eRender == dir && NULL != _ptrRenderCollection)
{
hr = _ptrRenderCollection->Item(index, &pDevice);
}
else if (NULL != _ptrCaptureCollection)
{
hr = _ptrCaptureCollection->Item(index, &pDevice);
}
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(pDevice);
return -1;
}
int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
SAFE_RELEASE(pDevice);
return res;
}
// ----------------------------------------------------------------------------
// _GetDefaultDeviceName
//
// Gets the friendly name of an endpoint rendering or capture device
// given a specified device role.
//
// Uses: _ptrEnumerator
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
assert(dir == eRender || dir == eCapture);
assert(role == eConsole || role == eCommunications);
assert(_ptrEnumerator != NULL);
hr = _ptrEnumerator->GetDefaultAudioEndpoint(
dir,
role,
&pDevice);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(pDevice);
return -1;
}
int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
SAFE_RELEASE(pDevice);
return res;
}
// ----------------------------------------------------------------------------
// _GetListDeviceID
//
// Gets the unique ID string of an endpoint rendering or capture device
// from the current list of such devices. The caller uses an index
// into the list to identify the device.
//
// Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
// in _RefreshDeviceList().
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
assert(dir == eRender || dir == eCapture);
if (eRender == dir && NULL != _ptrRenderCollection)
{
hr = _ptrRenderCollection->Item(index, &pDevice);
}
else if (NULL != _ptrCaptureCollection)
{
hr = _ptrCaptureCollection->Item(index, &pDevice);
}
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(pDevice);
return -1;
}
int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
SAFE_RELEASE(pDevice);
return res;
}
// ----------------------------------------------------------------------------
// _GetDefaultDeviceID
//
// Gets the uniqe device ID of an endpoint rendering or capture device
// given a specified device role.
//
// Uses: _ptrEnumerator
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr = S_OK;
IMMDevice *pDevice = NULL;
assert(dir == eRender || dir == eCapture);
assert(role == eConsole || role == eCommunications);
assert(_ptrEnumerator != NULL);
hr = _ptrEnumerator->GetDefaultAudioEndpoint(
dir,
role,
&pDevice);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(pDevice);
return -1;
}
int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
SAFE_RELEASE(pDevice);
return res;
}
int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
ERole role,
int* index)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr = S_OK;
WCHAR szDefaultDeviceID[MAX_PATH] = {0};
WCHAR szDeviceID[MAX_PATH] = {0};
const size_t kDeviceIDLength = sizeof(szDeviceID)/sizeof(szDeviceID[0]);
assert(kDeviceIDLength ==
sizeof(szDefaultDeviceID) / sizeof(szDefaultDeviceID[0]));
if (_GetDefaultDeviceID(dir,
role,
szDefaultDeviceID,
kDeviceIDLength) == -1)
{
return -1;
}
IMMDeviceCollection* collection = _ptrCaptureCollection;
if (dir == eRender)
{
collection = _ptrRenderCollection;
}
if (!collection)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Device collection not valid");
return -1;
}
UINT count = 0;
hr = collection->GetCount(&count);
if (FAILED(hr))
{
_TraceCOMError(hr);
return -1;
}
*index = -1;
for (UINT i = 0; i < count; i++)
{
memset(szDeviceID, 0, sizeof(szDeviceID));
rtc::scoped_refptr<IMMDevice> device;
{
IMMDevice* ptrDevice = NULL;
hr = collection->Item(i, &ptrDevice);
if (FAILED(hr) || ptrDevice == NULL)
{
_TraceCOMError(hr);
return -1;
}
device = ptrDevice;
SAFE_RELEASE(ptrDevice);
}
if (_GetDeviceID(device, szDeviceID, kDeviceIDLength) == -1)
{
return -1;
}
if (wcsncmp(szDefaultDeviceID, szDeviceID, kDeviceIDLength) == 0)
{
// Found a match.
*index = i;
break;
}
}
if (*index == -1)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Unable to find collection index for default device");
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// _GetDeviceName
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
LPWSTR pszBuffer,
int bufferLen)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
static const WCHAR szDefault[] = L"<Device not available>";
HRESULT hr = E_FAIL;
IPropertyStore *pProps = NULL;
PROPVARIANT varName;
assert(pszBuffer != NULL);
assert(bufferLen > 0);
if (pDevice != NULL)
{
hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
if (FAILED(hr))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"IMMDevice::OpenPropertyStore failed, hr = 0x%08X", hr);
}
}
// Initialize container for property value.
PropVariantInit(&varName);
if (SUCCEEDED(hr))
{
// Get the endpoint device's friendly-name property.
hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
if (FAILED(hr))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"IPropertyStore::GetValue failed, hr = 0x%08X", hr);
}
}
if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt))
{
hr = E_FAIL;
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"IPropertyStore::GetValue returned no value, hr = 0x%08X", hr);
}
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);
}
if (SUCCEEDED(hr) && (varName.pwszVal != NULL))
{
// Copy the valid device name to the provided ouput buffer.
wcsncpy_s(pszBuffer, bufferLen, varName.pwszVal, _TRUNCATE);
}
else
{
// Failed to find the device name.
wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
}
PropVariantClear(&varName);
SAFE_RELEASE(pProps);
return 0;
}
// ----------------------------------------------------------------------------
// _GetDeviceID
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
static const WCHAR szDefault[] = L"<Device not available>";
HRESULT hr = E_FAIL;
LPWSTR pwszID = NULL;
assert(pszBuffer != NULL);
assert(bufferLen > 0);
if (pDevice != NULL)
{
hr = pDevice->GetId(&pwszID);
}
if (hr == S_OK)
{
// Found the device ID.
wcsncpy_s(pszBuffer, bufferLen, pwszID, _TRUNCATE);
}
else
{
// Failed to find the device ID.
wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
}
CoTaskMemFree(pwszID);
return 0;
}
// ----------------------------------------------------------------------------
// _GetDefaultDevice
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
HRESULT hr(S_OK);
assert(_ptrEnumerator != NULL);
hr = _ptrEnumerator->GetDefaultAudioEndpoint(
dir,
role,
ppDevice);
if (FAILED(hr))
{
_TraceCOMError(hr);
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// _GetListDevice
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice)
{
HRESULT hr(S_OK);
assert(_ptrEnumerator != NULL);
IMMDeviceCollection *pCollection = NULL;
hr = _ptrEnumerator->EnumAudioEndpoints(
dir,
DEVICE_STATE_ACTIVE, // only active endpoints are OK
&pCollection);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(pCollection);
return -1;
}
hr = pCollection->Item(
index,
ppDevice);
if (FAILED(hr))
{
_TraceCOMError(hr);
SAFE_RELEASE(pCollection);
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// _EnumerateEndpointDevicesAll
// ----------------------------------------------------------------------------
int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow) const
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
assert(_ptrEnumerator != NULL);
HRESULT hr = S_OK;
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.
// Get states for *all* endpoint devices.
// Output: IMMDeviceCollection interface.
hr = _ptrEnumerator->EnumAudioEndpoints(
dataFlow, // data-flow direction (input parameter)
DEVICE_STATE_ACTIVE | DEVICE_STATE_DISABLED | DEVICE_STATE_UNPLUGGED,
&pCollection); // release interface when done
EXIT_ON_ERROR(hr);
// use the IMMDeviceCollection interface...
UINT count = 0;
// Retrieve a count of the devices in the device collection.
hr = pCollection->GetCount(&count);
EXIT_ON_ERROR(hr);
if (dataFlow == eRender)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#rendering endpoint devices (counting all): %u", count);
else if (dataFlow == eCapture)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#capturing endpoint devices (counting all): %u", count);
if (count == 0)
{
return 0;
}
// Each loop prints the name of an endpoint device.
for (ULONG i = 0; i < count; i++)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Endpoint %d:", i);
// Get pointer to endpoint number i.
// Output: IMMDevice interface.
hr = pCollection->Item(
i,
&pEndpoint);
CONTINUE_ON_ERROR(hr);
// use the IMMDevice interface of the specified endpoint device...
// 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);
// Retrieve an interface to the device's property store.
// Output: IPropertyStore interface.
hr = pEndpoint->OpenPropertyStore(
STGM_READ,
&pProps);
CONTINUE_ON_ERROR(hr);
// use the IPropertyStore interface...
PROPVARIANT varName;
// Initialize container for property value.
PropVariantInit(&varName);
// Get the endpoint's friendly-name property.
// Example: "Speakers (Realtek High Definition Audio)"
hr = pProps->GetValue(
PKEY_Device_FriendlyName,
&varName);
CONTINUE_ON_ERROR(hr);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", 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);
if (dwState & DEVICE_STATE_DISABLED)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : DISABLED", dwState);
if (dwState & DEVICE_STATE_NOTPRESENT)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : NOTPRESENT", dwState);
if (dwState & DEVICE_STATE_UNPLUGGED)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : UNPLUGGED", dwState);
// Check the hardware volume capabilities.
DWORD dwHwSupportMask = 0;
hr = pEndpoint->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL,
NULL, (void**)&pEndpointVolume);
CONTINUE_ON_ERROR(hr);
hr = pEndpointVolume->QueryHardwareSupport(&dwHwSupportMask);
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);
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);
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);
// 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);
if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
{
// Get the volume range.
float fLevelMinDB(0.0);
float fLevelMaxDB(0.0);
float fVolumeIncrementDB(0.0);
hr = pEndpointVolume->GetVolumeRange(
&fLevelMinDB,
&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);
// The volume range from vmin = fLevelMinDB to vmax = fLevelMaxDB is divided
// into n uniform intervals of size vinc = fVolumeIncrementDB, where
// n = (vmax ?vmin) / vinc.
// 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);
// Get information about the current step in the volume range.
// This method represents the volume level of the audio stream that enters or leaves
// the audio endpoint device as an index or "step" in a range of discrete volume levels.
// Output value nStepCount is the number of steps in the range. Output value nStep
// is the step index of the current volume level. If the number of steps is n = nStepCount,
// then step index nStep can assume values from 0 (minimum volume) to n ?1 (maximum volume).
UINT nStep(0);
UINT nStepCount(0);
hr = pEndpointVolume->GetVolumeStepInfo(
&nStep,
&nStepCount);
CONTINUE_ON_ERROR(hr);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "volume steps : %d (nStep), %d (nStepCount)", nStep, nStepCount);
}
Next:
if (FAILED(hr)) {
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Error when logging device information");
}
CoTaskMemFree(pwszID);
pwszID = NULL;
PropVariantClear(&varName);
SAFE_RELEASE(pProps);
SAFE_RELEASE(pEndpoint);
SAFE_RELEASE(pEndpointVolume);
}
SAFE_RELEASE(pCollection);
return 0;
Exit:
_TraceCOMError(hr);
CoTaskMemFree(pwszID);
pwszID = NULL;
SAFE_RELEASE(pCollection);
SAFE_RELEASE(pEndpoint);
SAFE_RELEASE(pEndpointVolume);
SAFE_RELEASE(pProps);
return -1;
}
// ----------------------------------------------------------------------------
// _TraceCOMError
// ----------------------------------------------------------------------------
void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const
{
TCHAR buf[MAXERRORLENGTH];
TCHAR errorText[MAXERRORLENGTH];
const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
FORMAT_MESSAGE_IGNORE_INSERTS;
const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
// Gets the system's human readable message string for this HRESULT.
// All error message in English by default.
DWORD messageLength = ::FormatMessageW(dwFlags,
0,
hr,
dwLangID,
errorText,
MAXERRORLENGTH,
NULL);
assert(messageLength <= MAXERRORLENGTH);
// Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
for (; messageLength && ::isspace(errorText[messageLength - 1]);
--messageLength)
{
errorText[messageLength - 1] = '\0';
}
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Core Audio method failed (hr=0x%x)", hr);
StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
StringCchCat(buf, MAXERRORLENGTH, errorText);
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
}
// ----------------------------------------------------------------------------
// WideToUTF8
// ----------------------------------------------------------------------------
char* AudioDeviceWindowsCore::WideToUTF8(const TCHAR* src) const {
#ifdef UNICODE
const size_t kStrLen = sizeof(_str);
memset(_str, 0, kStrLen);
// Get required size (in bytes) to be able to complete the conversion.
unsigned int required_size = (unsigned int)WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0);
if (required_size <= kStrLen)
{
// Process the entire input string, including the terminating null char.
if (WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, kStrLen, 0, 0) == 0)
memset(_str, 0, kStrLen);
}
return _str;
#else
return const_cast<char*>(src);
#endif
}
bool AudioDeviceWindowsCore::KeyPressed() const{
int key_down = 0;
for (int key = VK_SPACE; key < VK_NUMLOCK; key++) {
short res = GetAsyncKeyState(key);
key_down |= res & 0x1; // Get the LSB
}
return (key_down > 0);
}
} // namespace webrtc
#endif // WEBRTC_WINDOWS_CORE_AUDIO_BUILD