//-----------------------------------------------------------------------------
// Name: CSoundManager::Initialize()
// Desc: Initializes the IDirectSound object and also sets the primary buffer
// format. This function must be called before any others.
//-----------------------------------------------------------------------------
HRESULT CSoundManager::Initialize( HWND hWnd,
DWORD dwCoopLevel,
DWORD dwPrimaryChannels,
DWORD dwPrimaryFreq,
DWORD dwPrimaryBitRate )
{
HRESULT hr;
SAFE_RELEASE( m_pDS );
// Create IDirectSound using the primary sound device
if( FAILED( hr = DirectSoundCreate8( NULL, &m_pDS, NULL ) ) )
return DXTRACE_ERR( TEXT("DirectSoundCreate8"), hr );
// Set DirectSound coop level
if( FAILED( hr = m_pDS->SetCooperativeLevel( hWnd, dwCoopLevel ) ) )
return DXTRACE_ERR( TEXT("SetCooperativeLevel"), hr );
// Set primary buffer format
if( FAILED( hr = SetPrimaryBufferFormat( dwPrimaryChannels, dwPrimaryFreq, dwPrimaryBitRate ) ) )
return DXTRACE_ERR( TEXT("SetPrimaryBufferFormat"), hr );
m_dwPrimaryChannels = dwPrimaryChannels;
m_dwPrimaryFreq = dwPrimaryFreq;
m_dwPrimaryBitRate = dwPrimaryBitRate;
return S_OK;
}
// ////////////////////////////////////////////////////////////////////
//
// ////////////////////////////////////////////////////////////////////
bool DirectSound8Audio::VInitialize()
{
if(Audio::IsInitialized()) {
return (true);
}
Audio::SetInitialized(false);
Release(m_pDS);
HRESULT hr;
// Create IDirectSound using the primary sound device
if(FAILED(hr = DirectSoundCreate8(NULL, &m_pDS, NULL))) {
GF_LOG_TRACE_ERR("DirectSound8Audio::VInitialize()", "Failed to initialized the DirectSound interface");
return (false);
}
if(FAILED(hr = m_pDS->SetCooperativeLevel(m_hWnd, DSSCL_PRIORITY))) {
GF_LOG_TRACE_ERR("DirectSound8Audio::VInitialize()", "Failed to set the coop level of the DirectSound interface");
return (false);
}
if(FAILED(hr = SetPrimaryBufferFormat(8, 44100, 16))) {
GF_LOG_TRACE_ERR("DirectSound8Audio::VInitialize()", "Failed to set the primary buffer format of the DirectSound interface");
return (false);
}
Audio::SetInitialized(true);
m_AllSamples.clear();
return (true);
}
/*
===============
idAudioHardwareWIN32::Initialize
===============
*/
bool idAudioHardwareWIN32::Initialize( void ) {
int hr;
bufferSize = 0;
numSpeakers = 0;
blockAlign = 0;
SAFE_RELEASE( m_pDS );
// Create IDirectSound using the primary sound device
if( FAILED( hr = DirectSoundCreate( NULL, &m_pDS, NULL ) )) {
return false;
}
// Set primary buffer format
SetPrimaryBufferFormat( PRIMARYFREQ, 16, idSoundSystemLocal::s_numberOfSpeakers.GetInteger() );
return true;
}
bool CSoundManager::init(HWND hWnd)
{
// Set coop level to DSSCL_PRIORITY,
if( FAILED( Initialize( hWnd, DSSCL_PRIORITY) ) )
{
MessageBox( hWnd, L"Error initializing DirectSound. Sample will now exit.",
L"DirectSound Sample", MB_OK | MB_ICONERROR );
return false;
}
// set primary buffer format to stereo, 22kHz and 16-bit output.
if( FAILED( SetPrimaryBufferFormat( 2, 22050, 16 ) ) )
{
MessageBox( hWnd, L"Error initializing DirectSound. Sample will now exit.",
L"DirectSound Sample", MB_OK | MB_ICONERROR );
return false;
}
return true;
}
HX_RESULT CHXAudioDeviceDS::_DoOpen(const HXAudioFormat* pFormat)
{
HX_RESULT theErr = HXR_FAIL;
if(!m_hwnd || !m_hSoundDll)
return theErr;
// close open resources
InternalClose() ;
/* Get the IDirectSound interface */
HXBOOL bUsingDS8 = TRUE;
FPDIRECTSOUNDCREATE fpCreateDS = (FPDIRECTSOUNDCREATE) ::GetProcAddress(m_hSoundDll, TEXT("DirectSoundCreate8") );
if(!fpCreateDS)
{
bUsingDS8 = FALSE;
fpCreateDS = (FPDIRECTSOUNDCREATE) ::GetProcAddress(m_hSoundDll, TEXT("DirectSoundCreate"));
if(!fpCreateDS)
return theErr;
}
theErr = fpCreateDS(NULL, &m_pDSDev, NULL);
if (FAILED(theErr))
return theErr;
/* set the cooperative level. Because we want control over the format of the
primary buffer (16 bit, multichannel!), we need DSSCL_PRIORITY. */
m_pDSDev->SetCooperativeLevel(m_hwnd, DSSCL_PRIORITY );
/* set the format of the primary buffer. This will fail on WDM systems (because
the kernel mixer termines the primary buffer format), but is important on
non-WDM systems.
This might change the m_WaveFormat structure from a WAVE_FORMAT_EXTENSIBLE
to a WAVEFORMATEX.
Ignore the result.
*/
SetPrimaryBufferFormat() ;
/* Now open a secondary buffer. */
DSBUFFERDESC bufferDesc;
::memset(&bufferDesc, 0, sizeof(DSBUFFERDESC));
bufferDesc.dwSize = sizeof(DSBUFFERDESC);
bufferDesc.lpwfxFormat = m_pWaveFormat;
// Manipulate the buffer size so that is is an exact multiple of the block size.
// This will ensure that our write positions on the buffer are the same in every loop.
// We need to do this so that we have fixed playback notification positions marking the end each write block.
m_nBlocksPerBuffer = (m_pWaveFormat->nAvgBytesPerSec*BUFFER_TIME)/pFormat->uMaxBlockSize;
m_ulTotalBuffer = pFormat->uMaxBlockSize*m_nBlocksPerBuffer;
m_ulLoopTime = (double)m_ulTotalBuffer / (double)m_pWaveFormat->nAvgBytesPerSec;
bufferDesc.dwBufferBytes = m_ulTotalBuffer ;
bufferDesc.dwFlags =
DSBCAPS_CTRLVOLUME | // so we can control the volume
DSBCAPS_GETCURRENTPOSITION2 | // finer position reports
DSBCAPS_CTRLPOSITIONNOTIFY | // have them reported here
DSBCAPS_GLOBALFOCUS | // take control!
DSBCAPS_STICKYFOCUS |
#ifdef HELIX_FEATURE_AUDIO_DEVICE_HOOKS
(bUsingDS8 && m_pWaveFormat->nChannels <= 2 && !m_bOpaqueFormat ? DSBCAPS_CTRLFX : 0);
#else
0;
#endif
/* Again, try with WAVE_FORMAT_EXTENSIBLE first, but only if multichannel. */
theErr = !DS_OK ;
if (m_pWaveFormat->nChannels > 2 || m_bOpaqueFormat)
{
if(!m_bOpaqueFormat)
m_pWaveFormat->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
theErr = m_pDSDev->CreateSoundBuffer(&bufferDesc, &m_pSecondaryBuffer, NULL);
}
if (theErr != DS_OK && !m_bOpaqueFormat)
{
/* and if that fails, try WAVEFORMATEX */
m_pWaveFormat->wFormatTag = WAVE_FORMAT_PCM;
theErr = m_pDSDev->CreateSoundBuffer(&bufferDesc, &m_pSecondaryBuffer, NULL);
}
/* call it a day and count our blessings. */
switch (theErr)
{
case DS_OK:
theErr = HXR_OK;
break;
case DSERR_OUTOFMEMORY:
theErr = HXR_OUTOFMEMORY;
break;
default:
theErr = HXR_FAIL;
break;
}
if (SUCCEEDED(theErr) && m_pSecondaryBuffer)
{
m_eState = E_DEV_OPENED;
KillThreadAndEvent();
SetWindowLong(m_hwnd, GWL_USERDATA, (LONG)this);
// Create the event to be signalled on playback position notifications and the thread to wait for those events to be signalled.
m_hDSNotifyEvent = CreateEvent(NULL, TRUE, FALSE, kDSWaitEvent);
// now set the notification positions for direct sound playback.
IDirectSoundNotify* pNotify = NULL;
m_pSecondaryBuffer->QueryInterface(IID_IDirectSoundNotify, (void**)&pNotify);
if(pNotify && m_hDSNotifyEvent)
{
DSBPOSITIONNOTIFY* aPositionNotify = new DSBPOSITIONNOTIFY[m_nBlocksPerBuffer];
if(aPositionNotify)
{
for(int i = 0; i < m_nBlocksPerBuffer; i++)
{
aPositionNotify[i].dwOffset = i * pFormat->uMaxBlockSize;
aPositionNotify[i].hEventNotify = m_hDSNotifyEvent;
}
}
pNotify->SetNotificationPositions(m_nBlocksPerBuffer, aPositionNotify);
delete[] aPositionNotify;
DWORD dwWaitThreadID(0);
m_hWaitThread = CreateThread(NULL, 0, EventThreadProc, (LPVOID)this, 0, &dwWaitThreadID);
SetDebuggerThreadName(dwWaitThreadID, "DirectSound Audio Device Thread");
SetThreadPriority( m_hWaitThread, THREAD_PRIORITY_HIGHEST );
}
HX_RELEASE(pNotify);
m_pSecondaryBuffer->SetVolume(DSBVOLUME_MAX);
m_pSecondaryBuffer->SetCurrentPosition(0);
// detect whether we are playing remotely(via Remote Desktop)
// disable audio device hook if it is to avoid poor audio quality
int bRemoteSession = GetSystemMetrics(SM_REMOTESESSION);
if(!bRemoteSession && bUsingDS8 && m_pWaveFormat->nChannels <= 2 && !m_bOpaqueFormat)
{
LoadDirectSoundFilter();
}
}
m_ulCurrPlayTime = 0;
m_ulCurrLoopTime = 0;
m_ulLoops = 0;
// Setup converter to convert from samples/sec to milliseconds
m_TSConverter.SetBase(m_pWaveFormat->nSamplesPerSec, 1000);
return theErr;
}
// Sets a general quality of playback (CPU time vs. fidelity)
HRESULT DS3DSoundEngine::SetQuality(Quality quality)
{
HRESULT hr;
// stop all playing sound buffers
VirtualSoundList::iterator iterSound;
for (iterSound = m_listActiveSounds.begin();
iterSound != m_listActiveSounds.end(); ++iterSound)
{
if ((*iterSound)->HasPlayingBuffer())
{
hr = (*iterSound)->StopBuffer();
if (ZFailed(hr)) return hr;
}
}
switch (quality)
{
case minQuality:
hr = SetPrimaryBufferFormat(22050, 8, 1);
if ((hr != DSERR_PRIOLEVELNEEDED) && ZFailed(hr)) return hr;
m_nNumBuffersDesired = 8;
m_nNumBuffersMax = 8;
break;
case midQuality:
default:
hr = SetPrimaryBufferFormat(22050, 16, 2);
if ((hr != DSERR_PRIOLEVELNEEDED) && ZFailed(hr)) return hr;
if (m_bAllowHardware)
{
m_nNumBuffersDesired = max(16, (int)m_dscaps.dwMaxHwMixingStreamingBuffers * 2 / 3);
m_nNumBuffersMax = max(24, (int)m_dscaps.dwMaxHwMixingStreamingBuffers);
}
else
{
m_nNumBuffersDesired = 16;
m_nNumBuffersMax = 24;
}
break;
case maxQuality:
hr = SetPrimaryBufferFormat(44100, 16, 2);
if ((hr != DSERR_PRIOLEVELNEEDED) && ZFailed(hr)) return hr;
if (m_bAllowHardware)
{
m_nNumBuffersDesired = max(24, (int)m_dscaps.dwMaxHwMixingStreamingBuffers - 8);
m_nNumBuffersMax = max(32, (int)m_dscaps.dwMaxHwMixingStreamingBuffers);
}
else
{
m_nNumBuffersDesired = 24;
m_nNumBuffersMax = 32;
}
break;
}
m_quality = quality;
return S_OK;
};
