本文整理汇总了C++中IDirectSoundBuffer8::Unlock方法的典型用法代码示例。如果您正苦于以下问题:C++ IDirectSoundBuffer8::Unlock方法的具体用法?C++ IDirectSoundBuffer8::Unlock怎么用?C++ IDirectSoundBuffer8::Unlock使用的例子?那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在类IDirectSoundBuffer8
的用法示例。
在下文中一共展示了IDirectSoundBuffer8::Unlock方法的3个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C++代码示例。
示例1: Init
//.........这里部分代码省略.........
if( (FAILED(GUIDFromString( guid, &cGuid ))) ||
FAILED( DirectSoundCreate8(&cGuid,&dsound,NULL) ) )
throw std::runtime_error( "try again?" );
}
catch( std::runtime_error& )
{
// if the GUID failed, just open up the default dsound driver:
if( FAILED(DirectSoundCreate8(NULL,&dsound,NULL) ) )
throw std::runtime_error( "DirectSound failed to initialize!" );
}
if( FAILED(dsound->SetCooperativeLevel(GetDesktopWindow(),DSSCL_PRIORITY)) )
throw std::runtime_error( "DirectSound Error: Cooperative level could not be set." );
// Determine the user's speaker configuration, and select an expansion option as needed.
// FAIL : Directsound doesn't appear to support audio expansion >_<
DWORD speakerConfig = 2;
//dsound->GetSpeakerConfig( &speakerConfig );
IDirectSoundBuffer* buffer_;
DSBUFFERDESC desc;
// Set up WAV format structure.
memset(&wfx, 0, sizeof(WAVEFORMATEX));
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nSamplesPerSec = SampleRate;
wfx.nChannels = (WORD)speakerConfig;
wfx.wBitsPerSample = 16;
wfx.nBlockAlign = 2*(WORD)speakerConfig;
wfx.nAvgBytesPerSec = SampleRate * wfx.nBlockAlign;
wfx.cbSize = 0;
uint BufferSizeBytes = BufferSize * wfx.nBlockAlign;
// Set up DSBUFFERDESC structure.
memset(&desc, 0, sizeof(DSBUFFERDESC));
desc.dwSize = sizeof(DSBUFFERDESC);
desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_CTRLPOSITIONNOTIFY;
desc.dwBufferBytes = BufferSizeBytes * m_NumBuffers;
desc.lpwfxFormat = &wfx;
// Try a hardware buffer first, and then fall back on a software buffer if
// that one fails.
desc.dwFlags |= m_UseHardware ? DSBCAPS_LOCHARDWARE : DSBCAPS_LOCSOFTWARE;
desc.dwFlags |= m_DisableGlobalFocus ? DSBCAPS_STICKYFOCUS : DSBCAPS_GLOBALFOCUS;
if( FAILED(dsound->CreateSoundBuffer(&desc, &buffer_, 0) ) )
{
if( m_UseHardware )
{
desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_CTRLPOSITIONNOTIFY | DSBCAPS_LOCSOFTWARE;
desc.dwFlags |= m_DisableGlobalFocus ? DSBCAPS_STICKYFOCUS : DSBCAPS_GLOBALFOCUS;
if( FAILED(dsound->CreateSoundBuffer(&desc, &buffer_, 0) ) )
throw std::runtime_error( "DirectSound Error: Buffer could not be created." );
}
throw std::runtime_error( "DirectSound Error: Buffer could not be created." );
}
if( FAILED(buffer_->QueryInterface(IID_IDirectSoundBuffer8,(void**)&buffer)) || buffer == NULL )
throw std::runtime_error( "DirectSound Error: Interface could not be queried." );
buffer_->Release();
verifyc( buffer->QueryInterface(IID_IDirectSoundNotify8,(void**)&buffer_notify) );
DSBPOSITIONNOTIFY not[MAX_BUFFER_COUNT];
for(uint i=0;i<m_NumBuffers;i++)
{
buffer_events[i] = CreateEvent(NULL,FALSE,FALSE,NULL);
not[i].dwOffset = (wfx.nBlockAlign + BufferSizeBytes*(i+1)) % desc.dwBufferBytes;
not[i].hEventNotify = buffer_events[i];
}
buffer_notify->SetNotificationPositions(m_NumBuffers,not);
LPVOID p1=0,p2=0;
DWORD s1=0,s2=0;
verifyc(buffer->Lock(0,desc.dwBufferBytes,&p1,&s1,&p2,&s2,0));
assert(p2==0);
memset(p1,0,s1);
verifyc(buffer->Unlock(p1,s1,p2,s2));
//Play the buffer !
verifyc(buffer->Play(0,0,DSBPLAY_LOOPING));
// Start Thread
myLastWrite = 0;
dsound_running = true;
thread = CreateThread(NULL,0,(LPTHREAD_START_ROUTINE)RThread<StereoOut16>,this,0,&tid);
SetThreadPriority(thread,THREAD_PRIORITY_ABOVE_NORMAL);
return 0;
}
示例2: Init
s32 Init()
{
numBuffers = Config_DSoundOut.NumBuffers;
//
// Initialize DSound
//
GUID cGuid;
try
{
if( Config_DSoundOut.Device.empty() )
throw std::runtime_error( "screw it" );
// Convert from unicode to ANSI:
char guid[256];
sprintf_s( guid, "%S", Config_DSoundOut.Device.c_str() );
if( (FAILED(GUIDFromString( guid, &cGuid ))) ||
FAILED( DirectSoundCreate8(&cGuid,&dsound,NULL) ) )
throw std::runtime_error( "try again?" );
}
catch( std::runtime_error& )
{
// if the GUID failed, just open up the default dsound driver:
if( FAILED(DirectSoundCreate8(NULL,&dsound,NULL) ) )
throw std::runtime_error( "DirectSound failed to initialize!" );
}
if( FAILED(dsound->SetCooperativeLevel(GetDesktopWindow(),DSSCL_PRIORITY)) )
throw std::runtime_error( "DirectSound Error: Cooperative level could not be set." );
// Determine the user's speaker configuration, and select an expansion option as needed.
// FAIL : Directsound doesn't appear to support audio expansion >_<
DWORD speakerConfig = 2;
//dsound->GetSpeakerConfig( &speakerConfig );
IDirectSoundBuffer* buffer_;
DSBUFFERDESC desc;
// Set up WAV format structure.
memset(&wfx, 0, sizeof(WAVEFORMATEX));
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nSamplesPerSec = SampleRate;
wfx.nChannels = (WORD)speakerConfig;
wfx.wBitsPerSample = 16;
wfx.nBlockAlign = 2*(WORD)speakerConfig;
wfx.nAvgBytesPerSec = SampleRate * wfx.nBlockAlign;
wfx.cbSize = 0;
uint BufferSizeBytes = BufferSize * wfx.nBlockAlign;
// Set up DSBUFFERDESC structure.
memset(&desc, 0, sizeof(DSBUFFERDESC));
desc.dwSize = sizeof(DSBUFFERDESC);
desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_CTRLPOSITIONNOTIFY;// _CTRLPAN | DSBCAPS_CTRLVOLUME | DSBCAPS_CTRLFREQUENCY;
desc.dwBufferBytes = BufferSizeBytes * numBuffers;
desc.lpwfxFormat = &wfx;
desc.dwFlags |= DSBCAPS_LOCSOFTWARE;
desc.dwFlags |= DSBCAPS_GLOBALFOCUS;
if( FAILED(dsound->CreateSoundBuffer(&desc,&buffer_,0) ) )
throw std::runtime_error( "DirectSound Error: Interface could not be queried." );
if( FAILED(buffer_->QueryInterface(IID_IDirectSoundBuffer8,(void**)&buffer)) )
throw std::runtime_error( "DirectSound Error: Interface could not be queried." );
buffer_->Release();
verifyc( buffer->QueryInterface(IID_IDirectSoundNotify8,(void**)&buffer_notify) );
DSBPOSITIONNOTIFY not[MAX_BUFFER_COUNT];
for(u32 i=0;i<numBuffers;i++)
{
// [Air] note: wfx.nBlockAlign modifier was *10 -- seems excessive to me but maybe
// it was needed for some quirky driver? Theoretically we want the notification as soon
// as possible after the buffer has finished playing.
buffer_events[i] = CreateEvent(NULL,FALSE,FALSE,NULL);
not[i].dwOffset = (wfx.nBlockAlign + BufferSizeBytes*(i+1)) % desc.dwBufferBytes;
not[i].hEventNotify = buffer_events[i];
}
buffer_notify->SetNotificationPositions(numBuffers,not);
LPVOID p1=0,p2=0;
DWORD s1=0,s2=0;
verifyc(buffer->Lock(0,desc.dwBufferBytes,&p1,&s1,&p2,&s2,0));
assert(p2==0);
memset(p1,0,s1);
verifyc(buffer->Unlock(p1,s1,p2,s2));
//Play the buffer !
verifyc(buffer->Play(0,0,DSBPLAY_LOOPING));
//.........这里部分代码省略.........
示例3: CreateSound
//.........这里部分代码省略.........
// Read in the wave file header.
count = fread(&waveFileHeader, sizeof(waveFileHeader), 1, filePtr);
assert(count == 1);
// Check that the chunk ID is the RIFF format.
assert( (waveFileHeader.chunkId[0] == 'R') &&
(waveFileHeader.chunkId[1] == 'I') &&
(waveFileHeader.chunkId[2] == 'F') &&
(waveFileHeader.chunkId[3] == 'F'));
// Check that the file format is the WAVE format.
assert( (waveFileHeader.format[0] == 'W') &&
(waveFileHeader.format[1] == 'A') &&
(waveFileHeader.format[2] == 'V') &&
(waveFileHeader.format[3] == 'E'));
// Check that the sub chunk ID is the fmt format.
assert( (waveFileHeader.subChunkId[0] == 'f') &&
(waveFileHeader.subChunkId[1] == 'm') &&
(waveFileHeader.subChunkId[2] == 't') &&
(waveFileHeader.subChunkId[3] == ' '));
// Check that the audio format is WAVE_FORMAT_PCM.
assert(waveFileHeader.audioFormat == WAVE_FORMAT_PCM);
// Check that the wave file was recorded in stereo format.
assert(waveFileHeader.numChannels == 2);
// Check that the wave file was recorded at a sample rate of 44.1 KHz.
assert(waveFileHeader.sampleRate == 44100);
// Ensure that the wave file was recorded in 16 bit format.
assert(waveFileHeader.bitsPerSample == 16);
// Check for the data chunk header.
assert( (waveFileHeader.dataChunkId[0] == 'd') &&
(waveFileHeader.dataChunkId[1] == 'a') &&
(waveFileHeader.dataChunkId[2] == 't') &&
(waveFileHeader.dataChunkId[3] == 'a'));
// Set the wave format of secondary buffer that this wave file will be loaded onto.
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = 44100;
waveFormat.wBitsPerSample = 16;
waveFormat.nChannels = 2;
waveFormat.nBlockAlign = (waveFormat.wBitsPerSample / 8) * waveFormat.nChannels;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;
// Set the buffer description of the secondary sound buffer that the wave file will be loaded onto.
bufferDesc.dwSize = sizeof(DSBUFFERDESC);
bufferDesc.dwFlags = DSBCAPS_CTRLVOLUME;
bufferDesc.dwBufferBytes = waveFileHeader.dataSize;
bufferDesc.dwReserved = 0;
bufferDesc.lpwfxFormat = &waveFormat;
bufferDesc.guid3DAlgorithm = GUID_NULL;
// Create a temporary sound buffer with the specific buffer settings.
result = m_pDirectSound->CreateSoundBuffer(&bufferDesc, &tempBuffer, NULL);
assert(!FAILED(result));
// Test the buffer format against the direct sound 8 interface and create the secondary buffer.
result = tempBuffer->QueryInterface(IID_IDirectSoundBuffer8, (void**)&pSecondaryBuffer);
assert(!FAILED(result));
// Release the temporary buffer.
tempBuffer->Release();
tempBuffer = 0;
// Move to the beginning of the wave data which starts at the end of the data chunk header.
fseek(filePtr, sizeof(SWaveHeaderType), SEEK_SET);
// Create a temporary buffer to hold the wave file data.
waveData = new unsigned char[waveFileHeader.dataSize];
assert(waveData);
// Read in the wave file data into the newly created buffer.
count = fread(waveData, 1, waveFileHeader.dataSize, filePtr);
assert(count == waveFileHeader.dataSize);
// Close the file once done reading.
error = fclose(filePtr);
assert(error == 0);
// Lock the secondary buffer to write wave data into it.
result = pSecondaryBuffer->Lock(0, waveFileHeader.dataSize, (void**)&bufferPtr, (DWORD*)&bufferSize, NULL, 0, 0);
assert(!FAILED(result));
// Copy the wave data into the buffer.
memcpy(bufferPtr, waveData, waveFileHeader.dataSize);
// Unlock the secondary buffer after the data has been written to it.
result = pSecondaryBuffer->Unlock((void*)bufferPtr, bufferSize, NULL, 0);
assert(!FAILED(result));
// Release the wave data since it was copied into the secondary buffer.
delete [] waveData;
return CSound(pSecondaryBuffer);
}