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gfiber / vendor / opensource / webrtc / d73f3b5ff617e5b395e2af72e29289b87a1873e9 / . / trunk / src / modules / audio_device / main / source / win / audio_device_wave_win.cc
blob: 509732c4c9915b4c4ee520d3df653feea5b347d5 [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.
*/
#include "audio_device_utility.h"
#include "audio_device_wave_win.h"
#include "audio_device_config.h"
#include "trace.h"
#include "thread_wrapper.h"
#include "event_wrapper.h"
#include <windows.h>
#include <objbase.h> // CoTaskMemAlloc, CoTaskMemFree
#include <strsafe.h> // StringCchCopy(), StringCchCat(), StringCchPrintf()
#include <cassert>
// Avoids the need of Windows 7 SDK
#ifndef WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE
#define WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE 0x0010
#endif
// Supported in Windows Vista and Windows 7.
// http://msdn.microsoft.com/en-us/library/dd370819(v=VS.85).aspx
// Taken from Mmddk.h.
#define DRV_RESERVED 0x0800
#define DRV_QUERYFUNCTIONINSTANCEID (DRV_RESERVED + 17)
#define DRV_QUERYFUNCTIONINSTANCEIDSIZE (DRV_RESERVED + 18)
#define POW2(A) (2 << ((A) - 1))
namespace webrtc {
// ============================================================================
// Construction & Destruction
// ============================================================================
// ----------------------------------------------------------------------------
// AudioDeviceWindowsWave - ctor
// ----------------------------------------------------------------------------
AudioDeviceWindowsWave::AudioDeviceWindowsWave(const WebRtc_Word32 id) :
_ptrAudioBuffer(NULL),
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_timeEvent(*EventWrapper::Create()),
_recStartEvent(*EventWrapper::Create()),
_playStartEvent(*EventWrapper::Create()),
_hGetCaptureVolumeThread(NULL),
_hShutdownGetVolumeEvent(NULL),
_hSetCaptureVolumeThread(NULL),
_hShutdownSetVolumeEvent(NULL),
_hSetCaptureVolumeEvent(NULL),
_ptrThread(NULL),
_threadID(0),
_critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
_id(id),
_mixerManager(id),
_usingInputDeviceIndex(false),
_usingOutputDeviceIndex(false),
_inputDevice(AudioDeviceModule::kDefaultDevice),
_outputDevice(AudioDeviceModule::kDefaultDevice),
_inputDeviceIndex(0),
_outputDeviceIndex(0),
_inputDeviceIsSpecified(false),
_outputDeviceIsSpecified(false),
_initialized(false),
_recIsInitialized(false),
_playIsInitialized(false),
_recording(false),
_playing(false),
_startRec(false),
_stopRec(false),
_startPlay(false),
_stopPlay(false),
_AGC(false),
_hWaveIn(NULL),
_hWaveOut(NULL),
_recChannels(N_REC_CHANNELS),
_playChannels(N_PLAY_CHANNELS),
_recBufCount(0),
_recPutBackDelay(0),
_recDelayCount(0),
_playBufCount(0),
_prevPlayTime(0),
_prevRecTime(0),
_prevTimerCheckTime(0),
_timesdwBytes(0),
_timerFaults(0),
_timerRestartAttempts(0),
_no_of_msecleft_warnings(0),
_MAX_minBuffer(65),
_useHeader(0),
_dTcheckPlayBufDelay(10),
_playBufDelay(80),
_playBufDelayFixed(80),
_minPlayBufDelay(20),
_avgCPULoad(0),
_sndCardPlayDelay(0),
_sndCardRecDelay(0),
_plSampOld(0),
_rcSampOld(0),
_playBufType(AudioDeviceModule::kAdaptiveBufferSize),
_recordedBytes(0),
_playWarning(0),
_playError(0),
_recWarning(0),
_recError(0),
_newMicLevel(0),
_minMicVolume(0),
_maxMicVolume(0)
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
// Initialize value, set to 0 if it fails
if (!QueryPerformanceFrequency(&_perfFreq))
{
_perfFreq.QuadPart = 0;
}
_hShutdownGetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hShutdownSetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
}
// ----------------------------------------------------------------------------
// AudioDeviceWindowsWave - dtor
// ----------------------------------------------------------------------------
AudioDeviceWindowsWave::~AudioDeviceWindowsWave()
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
Terminate();
delete &_recStartEvent;
delete &_playStartEvent;
delete &_timeEvent;
delete &_critSect;
delete &_critSectCb;
if (NULL != _hShutdownGetVolumeEvent)
{
CloseHandle(_hShutdownGetVolumeEvent);
_hShutdownGetVolumeEvent = NULL;
}
if (NULL != _hShutdownSetVolumeEvent)
{
CloseHandle(_hShutdownSetVolumeEvent);
_hShutdownSetVolumeEvent = NULL;
}
if (NULL != _hSetCaptureVolumeEvent)
{
CloseHandle(_hSetCaptureVolumeEvent);
_hSetCaptureVolumeEvent = NULL;
}
}
// ============================================================================
// API
// ============================================================================
// ----------------------------------------------------------------------------
// AttachAudioBuffer
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
{
CriticalSectionScoped lock(_critSect);
_ptrAudioBuffer = audioBuffer;
// inform the AudioBuffer about default settings for this implementation
_ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
_ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
_ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
_ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
}
// ----------------------------------------------------------------------------
// ActiveAudioLayer
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
{
audioLayer = AudioDeviceModule::kWindowsWaveAudio;
return 0;
}
// ----------------------------------------------------------------------------
// Init
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::Init()
{
CriticalSectionScoped lock(_critSect);
if (_initialized)
{
return 0;
}
const WebRtc_UWord32 nowTime(AudioDeviceUtility::GetTimeInMS());
_recordedBytes = 0;
_prevRecByteCheckTime = nowTime;
_prevRecTime = nowTime;
_prevPlayTime = nowTime;
_prevTimerCheckTime = nowTime;
_playWarning = 0;
_playError = 0;
_recWarning = 0;
_recError = 0;
_mixerManager.EnumerateAll();
if (_ptrThread)
{
// thread is already created and active
return 0;
}
const char* threadName = "webrtc_audio_module_thread";
_ptrThread = ThreadWrapper::CreateThread(ThreadFunc,
this,
kRealtimePriority,
threadName);
if (_ptrThread == NULL)
{
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"failed to create the audio thread");
return -1;
}
unsigned int threadID(0);
if (!_ptrThread->Start(threadID))
{
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"failed to start the audio thread");
delete _ptrThread;
_ptrThread = NULL;
return -1;
}
_threadID = threadID;
const bool periodic(true);
if (!_timeEvent.StartTimer(periodic, TIMER_PERIOD_MS))
{
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"failed to start the timer event");
if (_ptrThread->Stop())
{
delete _ptrThread;
_ptrThread = NULL;
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"unable to stop the activated thread");
}
return -1;
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"periodic timer (dT=%d) is now active", TIMER_PERIOD_MS);
_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;
}
SetThreadPriority(_hGetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
_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;
}
SetThreadPriority(_hSetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
_initialized = true;
return 0;
}
// ----------------------------------------------------------------------------
// Terminate
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::Terminate()
{
if (!_initialized)
{
return 0;
}
_critSect.Enter();
_mixerManager.Close();
if (_ptrThread)
{
ThreadWrapper* tmpThread = _ptrThread;
_ptrThread = NULL;
_critSect.Leave();
tmpThread->SetNotAlive();
_timeEvent.Set();
if (tmpThread->Stop())
{
delete tmpThread;
}
else
{
_critSect.Leave();
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"failed to close down the audio thread");
return -1;
}
}
else
{
_critSect.Leave();
}
_critSect.Enter();
SetEvent(_hShutdownGetVolumeEvent);
_critSect.Leave();
WebRtc_Word32 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");
CloseHandle(_hGetCaptureVolumeThread);
_hGetCaptureVolumeThread = NULL;
return -1;
}
_critSect.Enter();
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" volume getter thread is now closed");
SetEvent(_hShutdownSetVolumeEvent);
_critSect.Leave();
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");
CloseHandle(_hSetCaptureVolumeThread);
_hSetCaptureVolumeThread = NULL;
return -1;
}
_critSect.Enter();
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" volume setter thread is now closed");
CloseHandle(_hGetCaptureVolumeThread);
_hGetCaptureVolumeThread = NULL;
CloseHandle(_hSetCaptureVolumeThread);
_hSetCaptureVolumeThread = NULL;
_critSect.Leave();
_timeEvent.StopTimer();
_initialized = false;
_outputDeviceIsSpecified = false;
_inputDeviceIsSpecified = false;
return 0;
}
DWORD WINAPI AudioDeviceWindowsWave::GetCaptureVolumeThread(LPVOID context)
{
return(((AudioDeviceWindowsWave*)context)->DoGetCaptureVolumeThread());
}
DWORD WINAPI AudioDeviceWindowsWave::SetCaptureVolumeThread(LPVOID context)
{
return(((AudioDeviceWindowsWave*)context)->DoSetCaptureVolumeThread());
}
DWORD AudioDeviceWindowsWave::DoGetCaptureVolumeThread()
{
HANDLE waitObject = _hShutdownGetVolumeEvent;
while (1)
{
DWORD waitResult = WaitForSingleObject(waitObject,
GET_MIC_VOLUME_INTERVAL_MS);
switch (waitResult)
{
case WAIT_OBJECT_0: // _hShutdownGetVolumeEvent
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;
}
if (AGC())
{
WebRtc_UWord32 currentMicLevel = 0;
if (MicrophoneVolume(currentMicLevel) == 0)
{
// This doesn't set the system volume, just stores it.
_critSect.Enter();
if (_ptrAudioBuffer)
{
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
_critSect.Leave();
}
}
}
}
DWORD AudioDeviceWindowsWave::DoSetCaptureVolumeThread()
{
HANDLE waitArray[2] = {_hShutdownSetVolumeEvent, _hSetCaptureVolumeEvent};
while (1)
{
DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
switch (waitResult)
{
case WAIT_OBJECT_0: // _hShutdownSetVolumeEvent
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;
}
_critSect.Enter();
WebRtc_UWord32 newMicLevel = _newMicLevel;
_critSect.Leave();
if (SetMicrophoneVolume(newMicLevel) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" the required modification of the microphone volume failed");
}
}
return 0;
}
// ----------------------------------------------------------------------------
// Initialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::Initialized() const
{
return (_initialized);
}
// ----------------------------------------------------------------------------
// SpeakerIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SpeakerIsAvailable(bool& available)
{
// Enumerate all avaliable speakers and make an attempt to open up the
// output mixer corresponding to the currently selected output device.
//
if (InitSpeaker() == -1)
{
available = false;
return 0;
}
// Given that InitSpeaker was successful, we know that a valid speaker exists
//
available = true;
// Close the initialized output mixer
//
_mixerManager.CloseSpeaker();
return 0;
}
// ----------------------------------------------------------------------------
// InitSpeaker
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::InitSpeaker()
{
CriticalSectionScoped lock(_critSect);
if (_playing)
{
return -1;
}
if (_mixerManager.EnumerateSpeakers() == -1)
{
// failed to locate any valid/controllable speaker
return -1;
}
if (IsUsingOutputDeviceIndex())
{
if (_mixerManager.OpenSpeaker(OutputDeviceIndex()) == -1)
{
return -1;
}
}
else
{
if (_mixerManager.OpenSpeaker(OutputDevice()) == -1)
{
return -1;
}
}
return 0;
}
// ----------------------------------------------------------------------------
// MicrophoneIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneIsAvailable(bool& available)
{
// Enumerate all avaliable microphones and make an attempt to open up the
// input mixer corresponding to the currently selected output device.
//
if (InitMicrophone() == -1)
{
available = false;
return 0;
}
// Given that InitMicrophone was successful, we know that a valid microphone exists
//
available = true;
// Close the initialized input mixer
//
_mixerManager.CloseMicrophone();
return 0;
}
// ----------------------------------------------------------------------------
// InitMicrophone
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::InitMicrophone()
{
CriticalSectionScoped lock(_critSect);
if (_recording)
{
return -1;
}
if (_mixerManager.EnumerateMicrophones() == -1)
{
// failed to locate any valid/controllable microphone
return -1;
}
if (IsUsingInputDeviceIndex())
{
if (_mixerManager.OpenMicrophone(InputDeviceIndex()) == -1)
{
return -1;
}
}
else
{
if (_mixerManager.OpenMicrophone(InputDevice()) == -1)
{
return -1;
}
}
WebRtc_UWord32 maxVol = 0;
if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" unable to retrieve max microphone volume");
}
_maxMicVolume = maxVol;
WebRtc_UWord32 minVol = 0;
if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" unable to retrieve min microphone volume");
}
_minMicVolume = minVol;
return 0;
}
// ----------------------------------------------------------------------------
// SpeakerIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::SpeakerIsInitialized() const
{
return (_mixerManager.SpeakerIsInitialized());
}
// ----------------------------------------------------------------------------
// MicrophoneIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::MicrophoneIsInitialized() const
{
return (_mixerManager.MicrophoneIsInitialized());
}
// ----------------------------------------------------------------------------
// SpeakerVolumeIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolumeIsAvailable(bool& available)
{
bool isAvailable(false);
// Enumerate all avaliable speakers and make an attempt to open up the
// output mixer corresponding to the currently selected output device.
//
if (InitSpeaker() == -1)
{
// failed to find a valid speaker
available = false;
return 0;
}
// Check if the selected speaker has a volume control
//
_mixerManager.SpeakerVolumeIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized output mixer
//
_mixerManager.CloseSpeaker();
return 0;
}
// ----------------------------------------------------------------------------
// SetSpeakerVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetSpeakerVolume(WebRtc_UWord32 volume)
{
return (_mixerManager.SetSpeakerVolume(volume));
}
// ----------------------------------------------------------------------------
// SpeakerVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolume(WebRtc_UWord32& volume) const
{
WebRtc_UWord32 level(0);
if (_mixerManager.SpeakerVolume(level) == -1)
{
return -1;
}
volume = level;
return 0;
}
// ----------------------------------------------------------------------------
// SetWaveOutVolume
//
// The low-order word contains the left-channel volume setting, and the
// high-order word contains the right-channel setting.
// A value of 0xFFFF represents full volume, and a value of 0x0000 is silence.
//
// If a device does not support both left and right volume control,
// the low-order word of dwVolume specifies the volume level,
// and the high-order word is ignored.
//
// Most devices do not support the full 16 bits of volume-level control
// and will not use the least-significant bits of the requested volume setting.
// For example, if a device supports 4 bits of volume control, the values
// 0x4000, 0x4FFF, and 0x43BE will all be truncated to 0x4000.
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
{
MMRESULT res(0);
WAVEOUTCAPS caps;
CriticalSectionScoped lock(_critSect);
if (_hWaveOut == NULL)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
}
// To determine whether the device supports volume control on both
// the left and right channels, use the WAVECAPS_LRVOLUME flag.
//
res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
TraceWaveOutError(res);
}
if (!(caps.dwSupport & WAVECAPS_VOLUME))
{
// this device does not support volume control using the waveOutSetVolume API
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
return -1;
}
if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
{
// high-order word (right channel) is ignored
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
}
DWORD dwVolume(0x00000000);
dwVolume = (DWORD)(((volumeRight & 0xFFFF) << 16) | (volumeLeft & 0xFFFF));
res = waveOutSetVolume(_hWaveOut, dwVolume);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutSetVolume() failed (err=%d)", res);
TraceWaveOutError(res);
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// WaveOutVolume
//
// The low-order word of this location contains the left-channel volume setting,
// and the high-order word contains the right-channel setting.
// A value of 0xFFFF (65535) represents full volume, and a value of 0x0000
// is silence.
//
// If a device does not support both left and right volume control,
// the low-order word of the specified location contains the mono volume level.
//
// The full 16-bit setting(s) set with the waveOutSetVolume function is returned,
// regardless of whether the device supports the full 16 bits of volume-level
// control.
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const
{
MMRESULT res(0);
WAVEOUTCAPS caps;
CriticalSectionScoped lock(_critSect);
if (_hWaveOut == NULL)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
}
// To determine whether the device supports volume control on both
// the left and right channels, use the WAVECAPS_LRVOLUME flag.
//
res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
TraceWaveOutError(res);
}
if (!(caps.dwSupport & WAVECAPS_VOLUME))
{
// this device does not support volume control using the waveOutSetVolume API
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
return -1;
}
if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
{
// high-order word (right channel) is ignored
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
}
DWORD dwVolume(0x00000000);
res = waveOutGetVolume(_hWaveOut, &dwVolume);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutGetVolume() failed (err=%d)", res);
TraceWaveOutError(res);
return -1;
}
WORD wVolumeLeft = LOWORD(dwVolume);
WORD wVolumeRight = HIWORD(dwVolume);
volumeLeft = static_cast<WebRtc_UWord16> (wVolumeLeft);
volumeRight = static_cast<WebRtc_UWord16> (wVolumeRight);
return 0;
}
// ----------------------------------------------------------------------------
// MaxSpeakerVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
{
WebRtc_UWord32 maxVol(0);
if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
{
return -1;
}
maxVolume = maxVol;
return 0;
}
// ----------------------------------------------------------------------------
// MinSpeakerVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
{
WebRtc_UWord32 minVol(0);
if (_mixerManager.MinSpeakerVolume(minVol) == -1)
{
return -1;
}
minVolume = minVol;
return 0;
}
// ----------------------------------------------------------------------------
// SpeakerVolumeStepSize
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
{
WebRtc_UWord16 delta(0);
if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
{
return -1;
}
stepSize = delta;
return 0;
}
// ----------------------------------------------------------------------------
// SpeakerMuteIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SpeakerMuteIsAvailable(bool& available)
{
bool isAvailable(false);
// Enumerate all avaliable speakers and make an attempt to open up the
// output mixer corresponding to the currently selected output device.
//
if (InitSpeaker() == -1)
{
// If we end up here it means that the selected speaker has no volume
// control, hence it is safe to state that there is no mute control
// already at this stage.
available = false;
return 0;
}
// Check if the selected speaker has a mute control
//
_mixerManager.SpeakerMuteIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized output mixer
//
_mixerManager.CloseSpeaker();
return 0;
}
// ----------------------------------------------------------------------------
// SetSpeakerMute
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetSpeakerMute(bool enable)
{
return (_mixerManager.SetSpeakerMute(enable));
}
// ----------------------------------------------------------------------------
// SpeakerMute
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SpeakerMute(bool& enabled) const
{
bool muted(0);
if (_mixerManager.SpeakerMute(muted) == -1)
{
return -1;
}
enabled = muted;
return 0;
}
// ----------------------------------------------------------------------------
// MicrophoneMuteIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneMuteIsAvailable(bool& available)
{
bool isAvailable(false);
// Enumerate all avaliable microphones and make an attempt to open up the
// input mixer corresponding to the currently selected input device.
//
if (InitMicrophone() == -1)
{
// If we end up here it means that the selected microphone has no volume
// control, hence it is safe to state that there is no boost control
// already at this stage.
available = false;
return 0;
}
// Check if the selected microphone has a mute control
//
_mixerManager.MicrophoneMuteIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized input mixer
//
_mixerManager.CloseMicrophone();
return 0;
}
// ----------------------------------------------------------------------------
// SetMicrophoneMute
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneMute(bool enable)
{
return (_mixerManager.SetMicrophoneMute(enable));
}
// ----------------------------------------------------------------------------
// MicrophoneMute
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneMute(bool& enabled) const
{
bool muted(0);
if (_mixerManager.MicrophoneMute(muted) == -1)
{
return -1;
}
enabled = muted;
return 0;
}
// ----------------------------------------------------------------------------
// MicrophoneBoostIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneBoostIsAvailable(bool& available)
{
bool isAvailable(false);
// Enumerate all avaliable microphones and make an attempt to open up the
// input mixer corresponding to the currently selected input device.
//
if (InitMicrophone() == -1)
{
// If we end up here it means that the selected microphone has no volume
// control, hence it is safe to state that there is no boost control
// already at this stage.
available = false;
return 0;
}
// Check if the selected microphone has a boost control
//
_mixerManager.MicrophoneBoostIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized input mixer
//
_mixerManager.CloseMicrophone();
return 0;
}
// ----------------------------------------------------------------------------
// SetMicrophoneBoost
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneBoost(bool enable)
{
return (_mixerManager.SetMicrophoneBoost(enable));
}
// ----------------------------------------------------------------------------
// MicrophoneBoost
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneBoost(bool& enabled) const
{
bool onOff(0);
if (_mixerManager.MicrophoneBoost(onOff) == -1)
{
return -1;
}
enabled = onOff;
return 0;
}
// ----------------------------------------------------------------------------
// StereoRecordingIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StereoRecordingIsAvailable(bool& available)
{
available = true;
return 0;
}
// ----------------------------------------------------------------------------
// SetStereoRecording
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetStereoRecording(bool enable)
{
if (enable)
_recChannels = 2;
else
_recChannels = 1;
return 0;
}
// ----------------------------------------------------------------------------
// StereoRecording
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StereoRecording(bool& enabled) const
{
if (_recChannels == 2)
enabled = true;
else
enabled = false;
return 0;
}
// ----------------------------------------------------------------------------
// StereoPlayoutIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StereoPlayoutIsAvailable(bool& available)
{
available = true;
return 0;
}
// ----------------------------------------------------------------------------
// SetStereoPlayout
//
// Specifies the number of output channels.
//
// NOTE - the setting will only have an effect after InitPlayout has
// been called.
//
// 16-bit mono:
//
// Each sample is 2 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
// For each sample, the first byte is the low-order byte of channel 0 and the
// second byte is the high-order byte of channel 0.
//
// 16-bit stereo:
//
// Each sample is 4 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
// For each sample, the first byte is the low-order byte of channel 0 (left channel);
// the second byte is the high-order byte of channel 0; the third byte is the
// low-order byte of channel 1 (right channel); and the fourth byte is the
// high-order byte of channel 1.
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetStereoPlayout(bool enable)
{
if (enable)
_playChannels = 2;
else
_playChannels = 1;
return 0;
}
// ----------------------------------------------------------------------------
// StereoPlayout
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StereoPlayout(bool& enabled) const
{
if (_playChannels == 2)
enabled = true;
else
enabled = false;
return 0;
}
// ----------------------------------------------------------------------------
// SetAGC
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetAGC(bool enable)
{
_AGC = enable;
return 0;
}
// ----------------------------------------------------------------------------
// AGC
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::AGC() const
{
return _AGC;
}
// ----------------------------------------------------------------------------
// MicrophoneVolumeIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolumeIsAvailable(bool& available)
{
bool isAvailable(false);
// Enumerate all avaliable microphones and make an attempt to open up the
// input mixer corresponding to the currently selected output device.
//
if (InitMicrophone() == -1)
{
// Failed to find valid microphone
available = false;
return 0;
}
// Check if the selected microphone has a volume control
//
_mixerManager.MicrophoneVolumeIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized input mixer
//
_mixerManager.CloseMicrophone();
return 0;
}
// ----------------------------------------------------------------------------
// SetMicrophoneVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneVolume(WebRtc_UWord32 volume)
{
return (_mixerManager.SetMicrophoneVolume(volume));
}
// ----------------------------------------------------------------------------
// MicrophoneVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolume(WebRtc_UWord32& volume) const
{
WebRtc_UWord32 level(0);
if (_mixerManager.MicrophoneVolume(level) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to retrive current microphone level");
return -1;
}
volume = level;
return 0;
}
// ----------------------------------------------------------------------------
// MaxMicrophoneVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
{
// _maxMicVolume can be zero in AudioMixerManager::MaxMicrophoneVolume():
// (1) API GetLineControl() returns failure at querying the max Mic level.
// (2) API GetLineControl() returns maxVolume as zero in rare cases.
// Both cases show we don't have access to the mixer controls.
// We return -1 here to indicate that.
if (_maxMicVolume == 0)
{
return -1;
}
maxVolume = _maxMicVolume;;
return 0;
}
// ----------------------------------------------------------------------------
// MinMicrophoneVolume
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
{
minVolume = _minMicVolume;
return 0;
}
// ----------------------------------------------------------------------------
// MicrophoneVolumeStepSize
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
{
WebRtc_UWord16 delta(0);
if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
{
return -1;
}
stepSize = delta;
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDevices
// ----------------------------------------------------------------------------
WebRtc_Word16 AudioDeviceWindowsWave::PlayoutDevices()
{
return (waveOutGetNumDevs());
}
// ----------------------------------------------------------------------------
// SetPlayoutDevice I (II)
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutDevice(WebRtc_UWord16 index)
{
if (_playIsInitialized)
{
return -1;
}
UINT nDevices = waveOutGetNumDevs();
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio output devices is %u", nDevices);
if (index < 0 || index > (nDevices-1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
return -1;
}
_usingOutputDeviceIndex = true;
_outputDeviceIndex = index;
_outputDeviceIsSpecified = true;
return 0;
}
// ----------------------------------------------------------------------------
// SetPlayoutDevice II (II)
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
{
if (_playIsInitialized)
{
return -1;
}
if (device == AudioDeviceModule::kDefaultDevice)
{
}
else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
{
}
_usingOutputDeviceIndex = false;
_outputDevice = device;
_outputDeviceIsSpecified = true;
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDeviceName
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
{
WebRtc_UWord16 nDevices(PlayoutDevices());
// Special fix for the case when the user asks for the name of the default device.
//
if (index == (WebRtc_UWord16)(-1))
{
index = 0;
}
if ((index > (nDevices-1)) || (name == NULL))
{
return -1;
}
memset(name, 0, kAdmMaxDeviceNameSize);
if (guid != NULL)
{
memset(guid, 0, kAdmMaxGuidSize);
}
WAVEOUTCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
MMRESULT res;
res = waveOutGetDevCapsW(index, &caps, sizeof(WAVEOUTCAPSW));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCapsW() failed (err=%d)", res);
return -1;
}
if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
}
if (guid == NULL)
{
return 0;
}
// It is possible to get the unique endpoint ID string using the Wave API.
// However, it is only supported on Windows Vista and Windows 7.
size_t cbEndpointId(0);
// Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
// Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
res = waveOutMessage((HWAVEOUT)IntToPtr(index),
DRV_QUERYFUNCTIONINSTANCEIDSIZE,
(DWORD_PTR)&cbEndpointId, NULL);
if (res != MMSYSERR_NOERROR)
{
// DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
TraceWaveOutError(res);
// Best we can do is to copy the friendly name and use it as guid
if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
}
return 0;
}
// waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
WCHAR *pstrEndpointId = NULL;
pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
// Get the endpoint ID string for this waveOut device.
res = waveOutMessage((HWAVEOUT)IntToPtr(index),
DRV_QUERYFUNCTIONINSTANCEID,
(DWORD_PTR)pstrEndpointId,
cbEndpointId);
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
TraceWaveOutError(res);
// Best we can do is to copy the friendly name and use it as guid
if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
}
CoTaskMemFree(pstrEndpointId);
return 0;
}
if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
}
CoTaskMemFree(pstrEndpointId);
return 0;
}
// ----------------------------------------------------------------------------
// RecordingDeviceName
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
{
WebRtc_UWord16 nDevices(RecordingDevices());
// Special fix for the case when the user asks for the name of the default device.
//
if (index == (WebRtc_UWord16)(-1))
{
index = 0;
}
if ((index > (nDevices-1)) || (name == NULL))
{
return -1;
}
memset(name, 0, kAdmMaxDeviceNameSize);
if (guid != NULL)
{
memset(guid, 0, kAdmMaxGuidSize);
}
WAVEINCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
MMRESULT res;
res = waveInGetDevCapsW(index, &caps, sizeof(WAVEINCAPSW));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCapsW() failed (err=%d)", res);
return -1;
}
if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
}
if (guid == NULL)
{
return 0;
}
// It is possible to get the unique endpoint ID string using the Wave API.
// However, it is only supported on Windows Vista and Windows 7.
size_t cbEndpointId(0);
// Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
// Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
res = waveInMessage((HWAVEIN)IntToPtr(index),
DRV_QUERYFUNCTIONINSTANCEIDSIZE,
(DWORD_PTR)&cbEndpointId, NULL);
if (res != MMSYSERR_NOERROR)
{
// DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
TraceWaveInError(res);
// Best we can do is to copy the friendly name and use it as guid
if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
}
return 0;
}
// waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
WCHAR *pstrEndpointId = NULL;
pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
// Get the endpoint ID string for this waveOut device.
res = waveInMessage((HWAVEIN)IntToPtr(index),
DRV_QUERYFUNCTIONINSTANCEID,
(DWORD_PTR)pstrEndpointId,
cbEndpointId);
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
TraceWaveInError(res);
// Best we can do is to copy the friendly name and use it as guid
if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
}
CoTaskMemFree(pstrEndpointId);
return 0;
}
if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
}
CoTaskMemFree(pstrEndpointId);
return 0;
}
// ----------------------------------------------------------------------------
// RecordingDevices
// ----------------------------------------------------------------------------
WebRtc_Word16 AudioDeviceWindowsWave::RecordingDevices()
{
return (waveInGetNumDevs());
}
// ----------------------------------------------------------------------------
// SetRecordingDevice I (II)
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetRecordingDevice(WebRtc_UWord16 index)
{
if (_recIsInitialized)
{
return -1;
}
UINT nDevices = waveInGetNumDevs();
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio input devices is %u", nDevices);
if (index < 0 || index > (nDevices-1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
return -1;
}
_usingInputDeviceIndex = true;
_inputDeviceIndex = index;
_inputDeviceIsSpecified = true;
return 0;
}
// ----------------------------------------------------------------------------
// SetRecordingDevice II (II)
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
{
if (device == AudioDeviceModule::kDefaultDevice)
{
}
else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
{
}
if (_recIsInitialized)
{
return -1;
}
_usingInputDeviceIndex = false;
_inputDevice = device;
_inputDeviceIsSpecified = true;
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::PlayoutIsAvailable(bool& available)
{
available = false;
// Try to initialize the playout side
WebRtc_Word32 res = InitPlayout();
// Cancel effect of initialization
StopPlayout();
if (res != -1)
{
available = true;
}
return 0;
}
// ----------------------------------------------------------------------------
// RecordingIsAvailable
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::RecordingIsAvailable(bool& available)
{
available = false;
// Try to initialize the recording side
WebRtc_Word32 res = InitRecording();
// Cancel effect of initialization
StopRecording();
if (res != -1)
{
available = true;
}
return 0;
}
// ----------------------------------------------------------------------------
// InitPlayout
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::InitPlayout()
{
CriticalSectionScoped lock(_critSect);
if (_playing)
{
return -1;
}
if (!_outputDeviceIsSpecified)
{
return -1;
}
if (_playIsInitialized)
{
return 0;
}
// Initialize the speaker (devices might have been added or removed)
if (InitSpeaker() == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
}
// Enumerate all availiable output devices
EnumeratePlayoutDevices();
// Start by closing any existing wave-output devices
//
MMRESULT res(MMSYSERR_ERROR);
if (_hWaveOut != NULL)
{
res = waveOutClose(_hWaveOut);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
TraceWaveOutError(res);
}
}
// Set the output wave format
//
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nChannels = _playChannels; // mono <=> 1, stereo <=> 2
waveFormat.nSamplesPerSec = N_PLAY_SAMPLES_PER_SEC;
waveFormat.wBitsPerSample = 16;
waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;
// Open the given waveform-audio output device for playout
//
HWAVEOUT hWaveOut(NULL);
if (IsUsingOutputDeviceIndex())
{
// verify settings first
res = waveOutOpen(NULL, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
if (MMSYSERR_NOERROR == res)
{
// open the given waveform-audio output device for recording
res = waveOutOpen(&hWaveOut, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening output device corresponding to device ID %u", _outputDeviceIndex);
}
}
else
{
if (_outputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
{
// check if it is possible to open the default communication device (supported on Windows 7)
res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
if (MMSYSERR_NOERROR == res)
{
// if so, open the default communication device for real
res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
}
else
{
// use default device since default communication device was not avaliable
res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
}
}
else if (_outputDevice == AudioDeviceModule::kDefaultDevice)
{
// open default device since it has been requested
res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
if (MMSYSERR_NOERROR == res)
{
res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default output device");
}
}
}
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutOpen() failed (err=%d)", res);
TraceWaveOutError(res);
return -1;
}
// Log information about the aquired output device
//
WAVEOUTCAPS caps;
res = waveOutGetDevCaps((UINT_PTR)hWaveOut, &caps, sizeof(WAVEOUTCAPS));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
TraceWaveOutError(res);
}
UINT deviceID(0);
res = waveOutGetID(hWaveOut, &deviceID);
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetID() failed (err=%d)", res);
TraceWaveOutError(res);
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
// Store valid handle for the open waveform-audio output device
_hWaveOut = hWaveOut;
// Store the input wave header as well
_waveFormatOut = waveFormat;
// Prepare wave-out headers
//
const WebRtc_UWord8 bytesPerSample = 2*_playChannels;
for (int n = 0; n < N_BUFFERS_OUT; n++)
{
// set up the output wave header
_waveHeaderOut[n].lpData = _playBuffer[n];
_waveHeaderOut[n].dwBufferLength = bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES;
_waveHeaderOut[n].dwFlags = 0;
_waveHeaderOut[n].dwLoops = 0;
memset(_playBuffer[n], 0, bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES);
// The waveOutPrepareHeader function prepares a waveform-audio data block for playback.
// The lpData, dwBufferLength, and dwFlags members of the WAVEHDR structure must be set
// before calling this function.
//
res = waveOutPrepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (err=%d)", n, res);
TraceWaveOutError(res);
}
// perform extra check to ensure that the header is prepared
if (_waveHeaderOut[n].dwFlags != WHDR_PREPARED)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (dwFlags != WHDR_PREPARED)", n);
}
}
// Mark playout side as initialized
_playIsInitialized = true;
_dTcheckPlayBufDelay = 10; // check playback buffer delay every 10 ms
_playBufCount = 0; // index of active output wave header (<=> output buffer index)
_playBufDelay = 80; // buffer delay/size is initialized to 80 ms and slowly decreased until er < 25
_minPlayBufDelay = 25; // minimum playout buffer delay
_MAX_minBuffer = 65; // adaptive minimum playout buffer delay cannot be larger than this value
_intro = 1; // Used to make sure that adaption starts after (2000-1700)/100 seconds
_waitCounter = 1700; // Counter for start of adaption of playback buffer
_erZeroCounter = 0; // Log how many times er = 0 in consequtive calls to RecTimeProc
_useHeader = 0; // Counts number of "useHeader" detections. Stops at 2.
_writtenSamples = 0;
_writtenSamplesOld = 0;
_playedSamplesOld = 0;
_sndCardPlayDelay = 0;
_sndCardRecDelay = 0;
WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id,"initial playout status: _playBufDelay=%d, _minPlayBufDelay=%d",
_playBufDelay, _minPlayBufDelay);
return 0;
}
// ----------------------------------------------------------------------------
// InitRecording
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::InitRecording()
{
CriticalSectionScoped lock(_critSect);
if (_recording)
{
return -1;
}
if (!_inputDeviceIsSpecified)
{
return -1;
}
if (_recIsInitialized)
{
return 0;
}
_avgCPULoad = 0;
_playAcc = 0;
// Initialize the microphone (devices might have been added or removed)
if (InitMicrophone() == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
}
// Enumerate all availiable input devices
EnumerateRecordingDevices();
// Start by closing any existing wave-input devices
//
MMRESULT res(MMSYSERR_ERROR);
if (_hWaveIn != NULL)
{
res = waveInClose(_hWaveIn);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
TraceWaveInError(res);
}
}
// Set the input wave format
//
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nChannels = _recChannels; // mono <=> 1, stereo <=> 2
waveFormat.nSamplesPerSec = N_REC_SAMPLES_PER_SEC;
waveFormat.wBitsPerSample = 16;
waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;
// Open the given waveform-audio input device for recording
//
HWAVEIN hWaveIn(NULL);
if (IsUsingInputDeviceIndex())
{
// verify settings first
res = waveInOpen(NULL, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
if (MMSYSERR_NOERROR == res)
{
// open the given waveform-audio input device for recording
res = waveInOpen(&hWaveIn, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening input device corresponding to device ID %u", _inputDeviceIndex);
}
}
else
{
if (_inputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
{
// check if it is possible to open the default communication device (supported on Windows 7)
res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
if (MMSYSERR_NOERROR == res)
{
// if so, open the default communication device for real
res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
}
else
{
// use default device since default communication device was not avaliable
res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
}
}
else if (_inputDevice == AudioDeviceModule::kDefaultDevice)
{
// open default device since it has been requested
res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
if (MMSYSERR_NOERROR == res)
{
res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default input device");
}
}
}
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInOpen() failed (err=%d)", res);
TraceWaveInError(res);
return -1;
}
// Log information about the aquired input device
//
WAVEINCAPS caps;
res = waveInGetDevCaps((UINT_PTR)hWaveIn, &caps, sizeof(WAVEINCAPS));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
TraceWaveInError(res);
}
UINT deviceID(0);
res = waveInGetID(hWaveIn, &deviceID);
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetID() failed (err=%d)", res);
TraceWaveInError(res);
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
// Store valid handle for the open waveform-audio input device
_hWaveIn = hWaveIn;
// Store the input wave header as well
_waveFormatIn = waveFormat;
// Mark recording side as initialized
_recIsInitialized = true;
_recBufCount = 0; // index of active input wave header (<=> input buffer index)
_recDelayCount = 0; // ensures that input buffers are returned with certain delay
return 0;
}
// ----------------------------------------------------------------------------
// StartRecording
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StartRecording()
{
if (!_recIsInitialized)
{
return -1;
}
if (_recording)
{
return 0;
}
// set state to ensure that the recording starts from the audio thread
_startRec = true;
// the audio thread will signal when recording has stopped
if (kEventTimeout == _recStartEvent.Wait(10000))
{
_startRec = false;
StopRecording();
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
return -1;
}
if (_recording)
{
// the recording state is set by the audio thread after recording has started
}
else
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// StopRecording
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StopRecording()
{
CriticalSectionScoped lock(_critSect);
if (!_recIsInitialized)
{
return 0;
}
if (_hWaveIn == NULL)
{
return -1;
}
bool wasRecording = _recording;
_recIsInitialized = false;
_recording = false;
MMRESULT res;
// Stop waveform-adio input. If there are any buffers in the queue, the
// current buffer will be marked as done (the dwBytesRecorded member in
// the header will contain the length of data), but any empty buffers in
// the queue will remain there.
//
res = waveInStop(_hWaveIn);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStop() failed (err=%d)", res);
TraceWaveInError(res);
}
// Stop input on the given waveform-audio input device and resets the current
// position to zero. All pending buffers are marked as done and returned to
// the application.
//
res = waveInReset(_hWaveIn);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInReset() failed (err=%d)", res);
TraceWaveInError(res);
}
// Clean up the preparation performed by the waveInPrepareHeader function.
// Only unprepare header if recording was ever started (and headers are prepared).
//
if (wasRecording)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInUnprepareHeader() will be performed");
for (int n = 0; n < N_BUFFERS_IN; n++)
{
res = waveInUnprepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader() failed (err=%d)", res);
TraceWaveInError(res);
}
}
}
// Close the given waveform-audio input device.
//
res = waveInClose(_hWaveIn);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
TraceWaveInError(res);
}
// Set the wave input handle to NULL
//
_hWaveIn = NULL;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveIn is now set to NULL");
return 0;
}
// ----------------------------------------------------------------------------
// RecordingIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::RecordingIsInitialized() const
{
return (_recIsInitialized);
}
// ----------------------------------------------------------------------------
// Recording
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::Recording() const
{
return (_recording);
}
// ----------------------------------------------------------------------------
// PlayoutIsInitialized
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::PlayoutIsInitialized() const
{
return (_playIsInitialized);
}
// ----------------------------------------------------------------------------
// StartPlayout
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StartPlayout()
{
if (!_playIsInitialized)
{
return -1;
}
if (_playing)
{
return 0;
}
// set state to ensure that playout starts from the audio thread
_startPlay = true;
// the audio thread will signal when recording has started
if (kEventTimeout == _playStartEvent.Wait(10000))
{
_startPlay = false;
StopPlayout();
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playout");
return -1;
}
if (_playing)
{
// the playing state is set by the audio thread after playout has started
}
else
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playing");
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// StopPlayout
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::StopPlayout()
{
CriticalSectionScoped lock(_critSect);
if (!_playIsInitialized)
{
return 0;
}
if (_hWaveOut == NULL)
{
return -1;
}
_playIsInitialized = false;
_playing = false;
_sndCardPlayDelay = 0;
_sndCardRecDelay = 0;
MMRESULT res;
// The waveOutReset function stops playback on the given waveform-audio
// output device and resets the current position to zero. All pending
// playback buffers are marked as done (WHDR_DONE) and returned to the application.
// After this function returns, the application can send new playback buffers
// to the device by calling waveOutWrite, or close the device by calling waveOutClose.
//
res = waveOutReset(_hWaveOut);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutReset() failed (err=%d)", res);
TraceWaveOutError(res);
}
// The waveOutUnprepareHeader function cleans up the preparation performed
// by the waveOutPrepareHeader function. This function must be called after
// the device driver is finished with a data block.
// You must call this function before freeing the buffer.
//
for (int n = 0; n < N_BUFFERS_OUT; n++)
{
res = waveOutUnprepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutUnprepareHeader() failed (err=%d)", res);
TraceWaveOutError(res);
}
}
// The waveOutClose function closes the given waveform-audio output device.
// The close operation fails if the device is still playing a waveform-audio
// buffer that was previously sent by calling waveOutWrite. Before calling
// waveOutClose, the application must wait for all buffers to finish playing
// or call the waveOutReset function to terminate playback.
//
res = waveOutClose(_hWaveOut);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
TraceWaveOutError(res);
}
_hWaveOut = NULL;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveOut is now set to NULL");
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutDelay
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::PlayoutDelay(WebRtc_UWord16& delayMS) const
{
CriticalSectionScoped lock(_critSect);
delayMS = (WebRtc_UWord16)_sndCardPlayDelay;
return 0;
}
// ----------------------------------------------------------------------------
// RecordingDelay
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::RecordingDelay(WebRtc_UWord16& delayMS) const
{
CriticalSectionScoped lock(_critSect);
delayMS = (WebRtc_UWord16)_sndCardRecDelay;
return 0;
}
// ----------------------------------------------------------------------------
// Playing
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::Playing() const
{
return (_playing);
}
// ----------------------------------------------------------------------------
// SetPlayoutBuffer
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS)
{
CriticalSectionScoped lock(_critSect);
_playBufType = type;
if (type == AudioDeviceModule::kFixedBufferSize)
{
_playBufDelayFixed = sizeMS;
}
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutBuffer
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
{
CriticalSectionScoped lock(_critSect);
type = _playBufType;
if (type == AudioDeviceModule::kFixedBufferSize)
{
sizeMS = _playBufDelayFixed;
}
else
{
sizeMS = _playBufDelay;
}
return 0;
}
// ----------------------------------------------------------------------------
// CPULoad
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::CPULoad(WebRtc_UWord16& load) const
{
load = static_cast<WebRtc_UWord16>(100*_avgCPULoad);
return 0;
}
// ----------------------------------------------------------------------------
// PlayoutWarning
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::PlayoutWarning() const
{
return ( _playWarning > 0);
}
// ----------------------------------------------------------------------------
// PlayoutError
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::PlayoutError() const
{
return ( _playError > 0);
}
// ----------------------------------------------------------------------------
// RecordingWarning
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::RecordingWarning() const
{
return ( _recWarning > 0);
}
// ----------------------------------------------------------------------------
// RecordingError
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::RecordingError() const
{
return ( _recError > 0);
}
// ----------------------------------------------------------------------------
// ClearPlayoutWarning
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::ClearPlayoutWarning()
{
_playWarning = 0;
}
// ----------------------------------------------------------------------------
// ClearPlayoutError
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::ClearPlayoutError()
{
_playError = 0;
}
// ----------------------------------------------------------------------------
// ClearRecordingWarning
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::ClearRecordingWarning()
{
_recWarning = 0;
}
// ----------------------------------------------------------------------------
// ClearRecordingError
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::ClearRecordingError()
{
_recError = 0;
}
// ============================================================================
// Private Methods
// ============================================================================
// ----------------------------------------------------------------------------
// InputSanityCheckAfterUnlockedPeriod
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::InputSanityCheckAfterUnlockedPeriod() const
{
if (_hWaveIn == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// OutputSanityCheckAfterUnlockedPeriod
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::OutputSanityCheckAfterUnlockedPeriod() const
{
if (_hWaveOut == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
return -1;
}
return 0;
}
// ----------------------------------------------------------------------------
// EnumeratePlayoutDevices
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::EnumeratePlayoutDevices()
{
WebRtc_UWord16 nDevices(PlayoutDevices());
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#output devices: %u", nDevices);
WAVEOUTCAPS caps;
MMRESULT res;
for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
{
res = waveOutGetDevCaps(deviceID, &caps, sizeof(WAVEOUTCAPS));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
if (caps.dwFormats & WAVE_FORMAT_48S16)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
}
if (caps.dwFormats & WAVE_FORMAT_48M16)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
TraceSupportFlags(caps.dwSupport);
}
return 0;
}
// ----------------------------------------------------------------------------
// EnumerateRecordingDevices
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::EnumerateRecordingDevices()
{
WebRtc_UWord16 nDevices(RecordingDevices());
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#input devices: %u", nDevices);
WAVEINCAPS caps;
MMRESULT res;
for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
{
res = waveInGetDevCaps(deviceID, &caps, sizeof(WAVEINCAPS));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
if (caps.dwFormats & WAVE_FORMAT_48S16)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
}
if (caps.dwFormats & WAVE_FORMAT_48M16)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
}
else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
}
return 0;
}
// ----------------------------------------------------------------------------
// TraceSupportFlags
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::TraceSupportFlags(DWORD dwSupport) const
{
TCHAR buf[256];
StringCchPrintf(buf, 128, TEXT("support flags : 0x%x "), dwSupport);
if (dwSupport & WAVECAPS_PITCH)
{
// supports pitch control
StringCchCat(buf, 256, TEXT("(PITCH)"));
}
if (dwSupport & WAVECAPS_PLAYBACKRATE)
{
// supports playback rate control
StringCchCat(buf, 256, TEXT("(PLAYBACKRATE)"));
}
if (dwSupport & WAVECAPS_VOLUME)
{
// supports volume control
StringCchCat(buf, 256, TEXT("(VOLUME)"));
}
if (dwSupport & WAVECAPS_LRVOLUME)
{
// supports separate left and right volume control
StringCchCat(buf, 256, TEXT("(LRVOLUME)"));
}
if (dwSupport & WAVECAPS_SYNC)
{
// the driver is synchronous and will block while playing a buffer
StringCchCat(buf, 256, TEXT("(SYNC)"));
}
if (dwSupport & WAVECAPS_SAMPLEACCURATE)
{
// returns sample-accurate position information
StringCchCat(buf, 256, TEXT("(SAMPLEACCURATE)"));
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
}
// ----------------------------------------------------------------------------
// TraceWaveInError
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::TraceWaveInError(MMRESULT error) const
{
TCHAR buf[MAXERRORLENGTH];
TCHAR msg[MAXERRORLENGTH];
StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
waveInGetErrorText(error, msg, MAXERRORLENGTH);
StringCchCat(buf, MAXERRORLENGTH, msg);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
}
// ----------------------------------------------------------------------------
// TraceWaveOutError
// ----------------------------------------------------------------------------
void AudioDeviceWindowsWave::TraceWaveOutError(MMRESULT error) const
{
TCHAR buf[MAXERRORLENGTH];
TCHAR msg[MAXERRORLENGTH];
StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
waveOutGetErrorText(error, msg, MAXERRORLENGTH);
StringCchCat(buf, MAXERRORLENGTH, msg);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
}
// ----------------------------------------------------------------------------
// PrepareStartPlayout
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::PrepareStartPlayout()
{
CriticalSectionScoped lock(_critSect);
if (_hWaveOut == NULL)
{
return -1;
}
// A total of 30ms of data is immediately placed in the SC buffer
//
WebRtc_Word8 zeroVec[4*PLAY_BUF_SIZE_IN_SAMPLES]; // max allocation
memset(zeroVec, 0, 4*PLAY_BUF_SIZE_IN_SAMPLES);
{
Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
}
_playAcc = 0;
_playWarning = 0;
_playError = 0;
_dc_diff_mean = 0;
_dc_y_prev = 0;
_dc_penalty_counter = 20;
_dc_prevtime = 0;
_dc_prevplay = 0;
return 0;
}
// ----------------------------------------------------------------------------
// PrepareStartRecording
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::PrepareStartRecording()
{
CriticalSectionScoped lock(_critSect);
if (_hWaveIn == NULL)
{
return -1;
}
_playAcc = 0;
_recordedBytes = 0;
_recPutBackDelay = REC_PUT_BACK_DELAY;
MMRESULT res;
MMTIME mmtime;
mmtime.wType = TIME_SAMPLES;
res = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition(TIME_SAMPLES) failed (err=%d)", res);
TraceWaveInError(res);
}
_read_samples = mmtime.u.sample;
_read_samples_old = _read_samples;
_rec_samples_old = mmtime.u.sample;
_wrapCounter = 0;
for (int n = 0; n < N_BUFFERS_IN; n++)
{
const WebRtc_UWord8 nBytesPerSample = 2*_recChannels;
// set up the input wave header
_waveHeaderIn[n].lpData = _recBuffer[n];
_waveHeaderIn[n].dwBufferLength = nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
_waveHeaderIn[n].dwFlags = 0;
_waveHeaderIn[n].dwBytesRecorded = 0;
_waveHeaderIn[n].dwUser = 0;
memset(_recBuffer[n], 0, nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES);
// prepare a buffer for waveform-audio input
res = waveInPrepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", n, res);
TraceWaveInError(res);
}
// send an input buffer to the given waveform-audio input device
res = waveInAddBuffer(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", n, res);
TraceWaveInError(res);
}
}
// start input on the given waveform-audio input device
res = waveInStart(_hWaveIn);
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStart() failed (err=%d)", res);
TraceWaveInError(res);
}
return 0;
}
// ----------------------------------------------------------------------------
// GetPlayoutBufferDelay
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::GetPlayoutBufferDelay(WebRtc_UWord32& writtenSamples, WebRtc_UWord32& playedSamples)
{
int i;
int ms_Header;
long playedDifference;
int msecInPlayoutBuffer(0); // #milliseconds of audio in the playout buffer
const WebRtc_UWord16 nSamplesPerMs = (WebRtc_UWord16)(N_PLAY_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
MMRESULT res;
MMTIME mmtime;
if (!_playing)
{
playedSamples = 0;
return (0);
}
// Retrieve the current playback position.
//
mmtime.wType = TIME_SAMPLES; // number of waveform-audio samples
res = waveOutGetPosition(_hWaveOut, &mmtime, sizeof(mmtime));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetPosition() failed (err=%d)", res);
TraceWaveOutError(res);
}
writtenSamples = _writtenSamples; // #samples written to the playout buffer
playedSamples = mmtime.u.sample; // current playout position in the playout buffer
// derive remaining amount (in ms) of data in the playout buffer
msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
// DEBUG_PRINTP("msecInPlayoutBuffer=%u\n", msecInPlayoutBuffer);
playedDifference = (long) (_playedSamplesOld - playedSamples);
if (playedDifference > 64000)
{
// If the sound cards number-of-played-out-samples variable wraps around before
// written_sampels wraps around this needs to be adjusted. This can happen on
// sound cards that uses less than 32 bits to keep track of number of played out
// sampels. To avoid being fooled by sound cards that sometimes produces false
// output we compare old value minus the new value with a large value. This is
// neccessary because some SC:s produce an output like 153, 198, 175, 230 which
// would trigger the wrap-around function if we didn't compare with a large value.
// The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
i = 31;
while((_playedSamplesOld <= (unsigned long)POW2(i)) && (i > 14)) {
i--;
}
if((i < 31) && (i > 14)) {
// Avoid adjusting when there is 32-bit wrap-around since that is
// something neccessary.
//
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "msecleft() => wrap around occured: %d bits used by sound card)", (i+1));
_writtenSamples = _writtenSamples - POW2(i + 1);
writtenSamples = _writtenSamples;
msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
}
}
else if ((_writtenSamplesOld > POW2(31)) && (writtenSamples < 96000))
{
// Wrap around as expected after having used all 32 bits. (But we still
// test if the wrap around happened earlier which it should not)
i = 31;
while (_writtenSamplesOld <= (unsigned long)POW2(i)) {
i--;
}
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured after having used all 32 bits)");
_writtenSamplesOld = writtenSamples;
_playedSamplesOld = playedSamples;
msecInPlayoutBuffer = (int)((writtenSamples + POW2(i + 1) - playedSamples)/nSamplesPerMs);
}
else if ((writtenSamples < 96000) && (playedSamples > POW2(31)))
{
// Wrap around has, as expected, happened for written_sampels before
// playedSampels so we have to adjust for this until also playedSampels
// has had wrap around.
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured: correction of output is done)");
_writtenSamplesOld = writtenSamples;
_playedSamplesOld = playedSamples;
msecInPlayoutBuffer = (int)((writtenSamples + POW2(32) - playedSamples)/nSamplesPerMs);
}
_writtenSamplesOld = writtenSamples;
_playedSamplesOld = playedSamples;
// We use the following formaula to track that playout works as it should
// y=playedSamples/48 - timeGetTime();
// y represent the clock drift between system clock and sound card clock - should be fairly stable
// When the exponential mean value of diff(y) goes away from zero something is wrong
// The exponential formula will accept 1% clock drift but not more
// The driver error means that we will play to little audio and have a high negative clock drift
// We kick in our alternative method when the clock drift reaches 20%
int diff,y;
int unsigned time =0;
// If we have other problems that causes playout glitches
// we don't want to switch playout method.
// Check if playout buffer is extremely low, or if we haven't been able to
// exectue our code in more than 40 ms
time = timeGetTime();
if ((msecInPlayoutBuffer < 20) || (time - _dc_prevtime > 40))
{
_dc_penalty_counter = 100;
}
if ((playedSamples != 0))
{
y = playedSamples/48 - time;
if ((_dc_y_prev != 0) && (_dc_penalty_counter == 0))
{
diff = y - _dc_y_prev;
_dc_diff_mean = (990*_dc_diff_mean)/1000 + 10*diff;
}
_dc_y_prev = y;
}
if (_dc_penalty_counter)
{
_dc_penalty_counter--;
}
if (_dc_diff_mean < -200)
{
// Always reset the filter
_dc_diff_mean = 0;
// Problem is detected. Switch delay method and set min buffer to 80.
// Reset the filter and keep monitoring the filter output.
// If issue is detected a second time, increase min buffer to 100.
// If that does not help, we must modify this scheme further.
_useHeader++;
if (_useHeader == 1)
{
_minPlayBufDelay = 80;
_playWarning = 1; // only warn first time
WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #1: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
}
else if (_useHeader == 2)
{
_minPlayBufDelay = 100; // add some more safety
WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #2: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
}
else
{
// This state should not be entered... (HA)
WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "further actions are required!");
}
if (_playWarning == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout warning exists");
}
_playWarning = 1; // triggers callback from module process thread
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kPlayoutWarning message posted: switching to alternative playout delay method");
}
_dc_prevtime = time;
_dc_prevplay = playedSamples;
// Try a very rough method of looking at how many buffers are still playing
ms_Header = 0;
for (i = 0; i < N_BUFFERS_OUT; i++) {
if ((_waveHeaderOut[i].dwFlags & WHDR_INQUEUE)!=0) {
ms_Header += 10;
}
}
if ((ms_Header-50) > msecInPlayoutBuffer) {
// Test for cases when GetPosition appears to be screwed up (currently just log....)
TCHAR infoStr[300];
if (_no_of_msecleft_warnings%20==0)
{
StringCchPrintf(infoStr, 300, TEXT("writtenSamples=%i, playedSamples=%i, msecInPlayoutBuffer=%i, ms_Header=%i"), writtenSamples, playedSamples, msecInPlayoutBuffer, ms_Header);
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, (const char*)infoStr);
}
_no_of_msecleft_warnings++;
}
// If this is true we have had a problem with the playout
if (_useHeader > 0)
{
return (ms_Header);
}
if (ms_Header < msecInPlayoutBuffer)
{
if (_no_of_msecleft_warnings % 100 == 0)
{
TCHAR str[300];
StringCchPrintf(str, 300, TEXT("_no_of_msecleft_warnings=%i, msecInPlayoutBuffer=%i ms_Header=%i (minBuffer=%i buffersize=%i writtenSamples=%i playedSamples=%i)"),
_no_of_msecleft_warnings, msecInPlayoutBuffer, ms_Header, _minPlayBufDelay, _playBufDelay, writtenSamples, playedSamples);
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, (const char*)str);
}
_no_of_msecleft_warnings++;
ms_Header -= 6; // Round off as we only have 10ms resolution + Header info is usually slightly delayed compared to GetPosition
if (ms_Header < 0)
ms_Header = 0;
return (ms_Header);
}
else
{
return (msecInPlayoutBuffer);
}
}
// ----------------------------------------------------------------------------
// GetRecordingBufferDelay
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::GetRecordingBufferDelay(WebRtc_UWord32& readSamples, WebRtc_UWord32& recSamples)
{
long recDifference;
MMTIME mmtime;
MMRESULT mmr;
const WebRtc_UWord16 nSamplesPerMs = (WebRtc_UWord16)(N_REC_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
// Retrieve the current input position of the given waveform-audio input device
//
mmtime.wType = TIME_SAMPLES;
mmr = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
if (MMSYSERR_NOERROR != mmr)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition() failed (err=%d)", mmr);
TraceWaveInError(mmr);
}
readSamples = _read_samples; // updated for each full fram in RecProc()
recSamples = mmtime.u.sample; // remaining time in input queue (recorded but not read yet)
recDifference = (long) (_rec_samples_old - recSamples);
if( recDifference > 64000) {
WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 1 (recDifference =%d)", recDifference);
// If the sound cards number-of-recorded-samples variable wraps around before
// read_sampels wraps around this needs to be adjusted. This can happen on
// sound cards that uses less than 32 bits to keep track of number of played out
// sampels. To avoid being fooled by sound cards that sometimes produces false
// output we compare old value minus the new value with a large value. This is
// neccessary because some SC:s produce an output like 153, 198, 175, 230 which
// would trigger the wrap-around function if we didn't compare with a large value.
// The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
//
int i = 31;
while((_rec_samples_old <= (unsigned long)POW2(i)) && (i > 14))
i--;
if((i < 31) && (i > 14)) {
// Avoid adjusting when there is 32-bit wrap-around since that is
// somethying neccessary.
//
_read_samples = _read_samples - POW2(i + 1);
readSamples = _read_samples;
_wrapCounter++;
} else {
WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"AEC (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
}
}
if((_wrapCounter>200)){
// Do nothing, handled later
}
else if((_rec_samples_old > POW2(31)) && (recSamples < 96000)) {
WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 2 (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
// Wrap around as expected after having used all 32 bits.
_read_samples_old = readSamples;
_rec_samples_old = recSamples;
_wrapCounter++;
return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
} else if((recSamples < 96000) && (readSamples > POW2(31))) {
WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 3 (readSamples %d recSamples %d)",readSamples, recSamples);
// Wrap around has, as expected, happened for rec_sampels before
// readSampels so we have to adjust for this until also readSampels
// has had wrap around.
_read_samples_old = readSamples;
_rec_samples_old = recSamples;
_wrapCounter++;
return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
}
_read_samples_old = _read_samples;
_rec_samples_old = recSamples;
int res=(((int)_rec_samples_old - (int)_read_samples_old)/nSamplesPerMs);
if((res > 2000)||(res < 0)||(_wrapCounter>200)){
// Reset everything
WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"msec_read error (res %d wrapCounter %d)",res, _wrapCounter);
MMTIME mmtime;
mmtime.wType = TIME_SAMPLES;
mmr=waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
if (mmr != MMSYSERR_NOERROR) {
WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "waveInGetPosition failed (mmr=%d)", mmr);
}
_read_samples=mmtime.u.sample;
_read_samples_old=_read_samples;
_rec_samples_old=mmtime.u.sample;
// Guess a decent value
res = 20;
}
_wrapCounter = 0;
return res;
}
// ============================================================================
// Thread Methods
// ============================================================================
// ----------------------------------------------------------------------------
// ThreadFunc
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::ThreadFunc(void* pThis)
{
return (static_cast<AudioDeviceWindowsWave*>(pThis)->ThreadProcess());
}
// ----------------------------------------------------------------------------
// ThreadProcess
// ----------------------------------------------------------------------------
bool AudioDeviceWindowsWave::ThreadProcess()
{
WebRtc_UWord32 time(0);
WebRtc_UWord32 playDiff(0);
WebRtc_UWord32 recDiff(0);
LONGLONG playTime(0);
LONGLONG recTime(0);
switch (_timeEvent.Wait(1000))
{
case kEventSignaled:
break;
case kEventError:
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "EventWrapper::Wait() failed => restarting timer");
_timeEvent.StopTimer();
_timeEvent.StartTimer(true, TIMER_PERIOD_MS);
return true;
case kEventTimeout:
return true;
}
time = AudioDeviceUtility::GetTimeInMS();
if (_startPlay)
{
if (PrepareStartPlayout() == 0)
{
_prevTimerCheckTime = time;
_prevPlayTime = time;
_startPlay = false;
_playing = true;
_playStartEvent.Set();
}
}
if (_startRec)
{
if (PrepareStartRecording() == 0)
{
_prevTimerCheckTime = time;
_prevRecTime = time;
_prevRecByteCheckTime = time;
_startRec = false;
_recording = true;
_recStartEvent.Set();
}
}
if (_playing)
{
playDiff = time - _prevPlayTime;
}
if (_recording)
{
recDiff = time - _prevRecTime;
}
if (_playing || _recording)
{
RestartTimerIfNeeded(time);
}
if (_playing &&
(playDiff > (WebRtc_UWord32)(_dTcheckPlayBufDelay - 1)) ||
(playDiff < 0))
{
Lock();
if (_playing)
{
if (PlayProc(playTime) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
}
_prevPlayTime = time;
if (playTime != 0)
_playAcc += playTime;
}
UnLock();
}
if (_playing && (playDiff > 12))
{
// It has been a long time since we were able to play out, try to
// compensate by calling PlayProc again.
//
Lock();
if (_playing)
{
if (PlayProc(playTime))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
}
_prevPlayTime = time;
if (playTime != 0)
_playAcc += playTime;
}
UnLock();
}
if (_recording &&
(recDiff > REC_CHECK_TIME_PERIOD_MS) ||
(recDiff < 0))
{
Lock();
if (_recording)
{
WebRtc_Word32 nRecordedBytes(0);
WebRtc_UWord16 maxIter(10);
// Deliver all availiable recorded buffers and update the CPU load measurement.
// We use a while loop here to compensate for the fact that the multi-media timer
// can sometimed enter a "bad state" after hibernation where the resolution is
// reduced from ~1ms to ~10-15 ms.
//
while ((nRecordedBytes = RecProc(recTime)) > 0)
{
maxIter--;
_recordedBytes += nRecordedBytes;
if (recTime && _perfFreq.QuadPart)
{
// Measure the average CPU load:
// This is a simplified expression where an exponential filter is used:
// _avgCPULoad = 0.99 * _avgCPULoad + 0.01 * newCPU,
// newCPU = (recTime+playAcc)/f is time in seconds
// newCPU / 0.01 is the fraction of a 10 ms period
// The two 0.01 cancels each other.
// NOTE - assumes 10ms audio buffers.
//
_avgCPULoad = (float)(_avgCPULoad*.99 + (recTime+_playAcc)/(double)(_perfFreq.QuadPart));
_playAcc = 0;
}
if (maxIter == 0)
{
// If we get this message ofte, our compensation scheme is not sufficient.
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "failed to compensate for reduced MM-timer resolution");
}
}
if (nRecordedBytes == -1)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "RecProc() failed");
}
_prevRecTime = time;
// Monitor the recording process and generate error/warning callbacks if needed
MonitorRecording(time);
}
UnLock();
}
if (!_recording)
{
_prevRecByteCheckTime = time;
_avgCPULoad = 0;
}
return true;
}
// ----------------------------------------------------------------------------
// RecProc
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
{
MMRESULT res;
WebRtc_UWord32 bufCount(0);
WebRtc_UWord32 nBytesRecorded(0);
consumedTime = 0;
// count modulo N_BUFFERS_IN (0,1,2,...,(N_BUFFERS_IN-1),0,1,2,..)
if (_recBufCount == N_BUFFERS_IN)
{
_recBufCount = 0;
}
bufCount = _recBufCount;
// take mono/stereo mode into account when deriving size of a full buffer
const WebRtc_UWord16 bytesPerSample = 2*_recChannels;
const WebRtc_UWord32 fullBufferSizeInBytes = bytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
// read number of recorded bytes for the given input-buffer
nBytesRecorded = _waveHeaderIn[bufCount].dwBytesRecorded;
if (nBytesRecorded == fullBufferSizeInBytes ||
(nBytesRecorded > 0))
{
WebRtc_Word32 msecOnPlaySide;
WebRtc_Word32 msecOnRecordSide;
WebRtc_UWord32 writtenSamples;
WebRtc_UWord32 playedSamples;
WebRtc_UWord32 readSamples, recSamples;
bool send = true;
WebRtc_UWord32 nSamplesRecorded = (nBytesRecorded/bytesPerSample); // divide by 2 or 4 depending on mono or stereo
if (nBytesRecorded == fullBufferSizeInBytes)
{
_timesdwBytes = 0;
}
else
{
// Test if it is stuck on this buffer
_timesdwBytes++;
if (_timesdwBytes < 5)
{
// keep trying
return (0);
}
else
{
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id,"nBytesRecorded=%d => don't use", nBytesRecorded);
_timesdwBytes = 0;
send = false;
}
}
// store the recorded buffer (no action will be taken if the #recorded samples is not a full buffer)
_ptrAudioBuffer->SetRecordedBuffer(_waveHeaderIn[bufCount].lpData, nSamplesRecorded);
// update #samples read
_read_samples += nSamplesRecorded;
// Check how large the playout and recording buffers are on the sound card.
// This info is needed by the AEC.
//
msecOnPlaySide = GetPlayoutBufferDelay(writtenSamples, playedSamples);
msecOnRecordSide = GetRecordingBufferDelay(readSamples, recSamples);
// If we use the alternative playout delay method, skip the clock drift compensation
// since it will be an unreliable estimate and might degrade AEC performance.
WebRtc_Word32 drift = (_useHeader > 0) ? 0 : GetClockDrift(playedSamples, recSamples);
_ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, drift);
// Store the play and rec delay values for video synchronization
_sndCardPlayDelay = msecOnPlaySide;
_sndCardRecDelay = msecOnRecordSide;
LARGE_INTEGER t1,t2;
if (send)
{
QueryPerformanceCounter(&t1);
// deliver recorded samples at specified sample rate, mic level etc. to the observer using callback
UnLock();
_ptrAudioBuffer->DeliverRecordedData();
Lock();
QueryPerformanceCounter(&t2);
if (InputSanityCheckAfterUnlockedPeriod() == -1)
{
// assert(false);
return -1;
}
}
if (_AGC)
{
WebRtc_UWord32 newMicLevel = _ptrAudioBuffer->NewMicLevel();
if (newMicLevel != 0)
{
// The VQE will only deliver non-zero microphone levels when a change is needed.
WEBRTC_TRACE(kTraceStream, kTraceUtility, _id,"AGC change of volume: => new=%u", newMicLevel);
// We store this outside of the audio buffer to avoid
// having it overwritten by the getter thread.
_newMicLevel = newMicLevel;
SetEvent(_hSetCaptureVolumeEvent);
}
}
// return utilized buffer to queue after specified delay (default is 4)
if (_recDelayCount > (_recPutBackDelay-1))
{
// deley buffer counter to compensate for "put-back-delay"
bufCount = (bufCount + N_BUFFERS_IN - _recPutBackDelay) % N_BUFFERS_IN;
// reset counter so we can make new detection
_waveHeaderIn[bufCount].dwBytesRecorded = 0;
// return the utilized wave-header after certain delay (given by _recPutBackDelay)
res = waveInUnprepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader(%d) failed (err=%d)", bufCount, res);
TraceWaveInError(res);
}
// ensure that the utilized header can be used again
res = waveInPrepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", bufCount, res);
TraceWaveInError(res);
return -1;
}
// add the utilized buffer to the queue again
res = waveInAddBuffer(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
if (res != MMSYSERR_NOERROR)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", bufCount, res);
TraceWaveInError(res);
if (_recPutBackDelay < 50)
{
_recPutBackDelay++;
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "_recPutBackDelay increased to %d", _recPutBackDelay);
}
else
{
if (_recError == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
}
_recError = 1; // triggers callback from module process thread
WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: _recPutBackDelay=%u", _recPutBackDelay);
}
}
} // if (_recDelayCount > (_recPutBackDelay-1))
if (_recDelayCount < (_recPutBackDelay+1))
{
_recDelayCount++;
}
// increase main buffer count since one complete buffer has now been delivered
_recBufCount++;
if (send) {
// Calculate processing time
consumedTime = (int)(t2.QuadPart-t1.QuadPart);
// handle wraps, time should not be higher than a second
if ((consumedTime > _perfFreq.QuadPart) || (consumedTime < 0))
consumedTime = 0;
}
} // if ((nBytesRecorded == fullBufferSizeInBytes))
return nBytesRecorded;
}
// ----------------------------------------------------------------------------
// PlayProc
// ----------------------------------------------------------------------------
int AudioDeviceWindowsWave::PlayProc(LONGLONG& consumedTime)
{
WebRtc_Word32 remTimeMS(0);
WebRtc_Word8 playBuffer[4*PLAY_BUF_SIZE_IN_SAMPLES];
WebRtc_UWord32 writtenSamples(0);
WebRtc_UWord32 playedSamples(0);
LARGE_INTEGER t1;
LARGE_INTEGER t2;
consumedTime = 0;
_waitCounter++;
// Get number of ms of sound that remains in the sound card buffer for playback.
//
remTimeMS = GetPlayoutBufferDelay(writtenSamples, playedSamples);
// The threshold can be adaptive or fixed. The adaptive scheme is updated
// also for fixed mode but the updated threshold is not utilized.
//
const WebRtc_UWord16 thresholdMS =
(_playBufType == AudioDeviceModule::kAdaptiveBufferSize) ? _playBufDelay : _playBufDelayFixed;
if (remTimeMS < thresholdMS + 9)
{
_dTcheckPlayBufDelay = 5;
if (remTimeMS == 0)
{
WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id, "playout buffer is empty => we must adapt...");
if (_waitCounter > 30)
{
_erZeroCounter++;
if (_erZeroCounter == 2)
{
_playBufDelay += 15;
_minPlayBufDelay += 20;
_waitCounter = 50;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0,erZero=2): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
}
else if (_erZeroCounter == 3)
{
_erZeroCounter = 0;
_playBufDelay += 30;
_minPlayBufDelay += 25;
_waitCounter = 0;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=3): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
}
else
{
_minPlayBufDelay += 10;
_playBufDelay += 15;
_waitCounter = 50;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=1): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
}
}
}
else if (remTimeMS < _minPlayBufDelay)
{
// If there is less than 25 ms of audio in the play out buffer
// increase the buffersize limit value. _waitCounter prevents
// _playBufDelay to be increased every time this function is called.
if (_waitCounter > 30)
{
_playBufDelay += 10;
if (_intro == 0)
_waitCounter = 0;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is increased: playBufDelay=%u", _playBufDelay);
}
}
else if (remTimeMS < thresholdMS - 9)
{
_erZeroCounter = 0;
}
else
{
_erZeroCounter = 0;
_dTcheckPlayBufDelay = 10;
}
QueryPerformanceCounter(&t1); // measure time: START
// Ask for new PCM data to be played out using the AudioDeviceBuffer.
// Ensure that this callback is executed without taking the audio-thread lock.
//
UnLock();
WebRtc_UWord32 nSamples = _ptrAudioBuffer->RequestPlayoutData(PLAY_BUF_SIZE_IN_SAMPLES);
Lock();
if (OutputSanityCheckAfterUnlockedPeriod() == -1)
{
// assert(false);
return -1;
}
nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
if (nSamples != PLAY_BUF_SIZE_IN_SAMPLES)
{
WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "invalid number of output samples(%d)", nSamples);
}
QueryPerformanceCounter(&t2); // measure time: STOP
consumedTime = (int)(t2.QuadPart - t1.QuadPart);
Write(playBuffer, PLAY_BUF_SIZE_IN_SAMPLES);
} // if (er < thresholdMS + 9)
else if (thresholdMS + 9 < remTimeMS )
{
_erZeroCounter = 0;
_dTcheckPlayBufDelay = 2; // check buffer more often
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Need to check playout buffer more often (dT=%u, remTimeMS=%u)", _dTcheckPlayBufDelay, remTimeMS);
}
// If the buffersize has been stable for 20 seconds try to decrease the buffer size
if (_waitCounter > 2000)
{
_intro = 0;
_playBufDelay--;
_waitCounter = 1990;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is decreased: playBufDelay=%u", _playBufDelay);
}
// Limit the minimum sound card (playback) delay to adaptive minimum delay
if (_playBufDelay < _minPlayBufDelay)
{
_playBufDelay = _minPlayBufDelay;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %u", _minPlayBufDelay);
}
// Limit the maximum sound card (playback) delay to 150 ms
if (_playBufDelay > 150)
{
_playBufDelay = 150;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %d", _playBufDelay);
}
// Upper limit of the minimum sound card (playback) delay to 65 ms.
// Deactivated during "useHeader mode" (_useHeader > 0).
if (_minPlayBufDelay > _MAX_minBuffer &&
(_useHeader == 0))
{
_minPlayBufDelay = _MAX_minBuffer;
WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Minimum playout threshold is limited to %d", _MAX_minBuffer);
}
return (0);
}
// ----------------------------------------------------------------------------
// Write
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::Write(WebRtc_Word8* data, WebRtc_UWord16 nSamples)
{
if (_hWaveOut == NULL)
{
return -1;
}
if (_playIsInitialized)
{
MMRESULT res;
const WebRtc_UWord16 bufCount(_playBufCount);
// Place data in the memory associated with _waveHeaderOut[bufCount]
//
const WebRtc_Word16 nBytes = (2*_playChannels)*nSamples;
memcpy(&_playBuffer[bufCount][0], &data[0], nBytes);
// Send a data block to the given waveform-audio output device.
//
// When the buffer is finished, the WHDR_DONE bit is set in the dwFlags
// member of the WAVEHDR structure. The buffer must be prepared with the
// waveOutPrepareHeader function before it is passed to waveOutWrite.
// Unless the device is paused by calling the waveOutPause function,
// playback begins when the first data block is sent to the device.
//
res = waveOutWrite(_hWaveOut, &_waveHeaderOut[bufCount], sizeof(_waveHeaderOut[bufCount]));
if (MMSYSERR_NOERROR != res)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutWrite(%d) failed (err=%d)", bufCount, res);
TraceWaveOutError(res);
_writeErrors++;
if (_writeErrors > 10)
{
if (_playError == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout error exists");
}
_playError = 1; // triggers callback from module process thread
WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: _writeErrors=%u", _writeErrors);
}
return -1;
}
_playBufCount = (_playBufCount+1) % N_BUFFERS_OUT; // increase buffer counter modulo size of total buffer
_writtenSamples += nSamples; // each sample is 2 or 4 bytes
_writeErrors = 0;
}
return 0;
}
// ----------------------------------------------------------------------------
// GetClockDrift
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::GetClockDrift(const WebRtc_UWord32 plSamp, const WebRtc_UWord32 rcSamp)
{
int drift = 0;
unsigned int plSampDiff = 0, rcSampDiff = 0;
if (plSamp >= _plSampOld)
{
plSampDiff = plSamp - _plSampOld;
}
else
{
// Wrap
int i = 31;
while(_plSampOld <= (unsigned int)POW2(i))
{
i--;
}
// Add the amount remaining prior to wrapping
plSampDiff = plSamp + POW2(i + 1) - _plSampOld;
}
if (rcSamp >= _rcSampOld)
{
rcSampDiff = rcSamp - _rcSampOld;
}
else
{ // Wrap
int i = 31;
while(_rcSampOld <= (unsigned int)POW2(i))
{
i--;
}
rcSampDiff = rcSamp + POW2(i + 1) - _rcSampOld;
}
drift = plSampDiff - rcSampDiff;
_plSampOld = plSamp;
_rcSampOld = rcSamp;
return drift;
}
// ----------------------------------------------------------------------------
// MonitorRecording
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::MonitorRecording(const WebRtc_UWord32 time)
{
const WebRtc_UWord16 bytesPerSample = 2*_recChannels;
const WebRtc_UWord32 nRecordedSamples = _recordedBytes/bytesPerSample;
if (nRecordedSamples > 5*N_REC_SAMPLES_PER_SEC)
{
// 5 seconds of audio has been recorded...
if ((time - _prevRecByteCheckTime) > 5700)
{
// ...and it was more than 5.7 seconds since we last did this check <=>
// we have not been able to record 5 seconds of audio in 5.7 seconds,
// hence a problem should be reported.
// This problem can be related to USB overload.
//
if (_recWarning == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording warning exists");
}
_recWarning = 1; // triggers callback from module process thread
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kRecordingWarning message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
}
_recordedBytes = 0; // restart "check again when 5 seconds are recorded"
_prevRecByteCheckTime = time; // reset timer to measure time for recording of 5 seconds
}
if ((time - _prevRecByteCheckTime) > 8000)
{
// It has been more than 8 seconds since we able to confirm that 5 seconds of
// audio was recorded, hence we have not been able to record 5 seconds in
// 8 seconds => the complete recording process is most likely dead.
//
if (_recError == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
}
_recError = 1; // triggers callback from module process thread
WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
_prevRecByteCheckTime = time;
}
return 0;
}
// ----------------------------------------------------------------------------
// MonitorRecording
//
// Restart timer if needed (they seem to be messed up after a hibernate).
// ----------------------------------------------------------------------------
WebRtc_Word32 AudioDeviceWindowsWave::RestartTimerIfNeeded(const WebRtc_UWord32 time)
{
const WebRtc_UWord32 diffMS = time - _prevTimerCheckTime;
_prevTimerCheckTime = time;
if (diffMS > 7)
{
// one timer-issue detected...
_timerFaults++;
if (_timerFaults > 5 && _timerRestartAttempts < 2)
{
// Reinitialize timer event if event fails to execute at least every 5ms.
// On some machines it helps and the timer starts working as it should again;
// however, not all machines (we have seen issues on e.g. IBM T60).
// Therefore, the scheme below ensures that we do max 2 attempts to restart the timer.
// For the cases where restart does not do the trick, we compensate for the reduced
// resolution on both the recording and playout sides.
WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, " timer issue detected => timer is restarted");
_timeEvent.StopTimer();
_timeEvent.StartTimer(true, TIMER_PERIOD_MS);
// make sure timer gets time to start up and we don't kill/start timer serveral times over and over again
_timerFaults = -20;
_timerRestartAttempts++;
}
}
else
{
// restart timer-check scheme since we are OK
_timerFaults = 0;
_timerRestartAttempts = 0;
}
return 0;
}
} // namespace webrtc