put ASIO member variables in class, prepared for ASIO input/output channel selection
This commit is contained in:
Volker Fischer
parent
29eb59d464
commit
9828ceec81
2 changed files with 557 additions and 511 deletions
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@ -33,24 +33,7 @@
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extern AsioDrivers* asioDrivers;
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bool loadAsioDriver ( char *name );
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// TODO the following variables should be in the class definition but we cannot
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// do it here since we have static callback functions which cannot access the
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// class members :-(((
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// ASIO stuff
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ASIODriverInfo driverInfo;
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ASIOBufferInfo bufferInfos[2 * NUM_IN_OUT_CHANNELS]; // for input and output buffers -> "2 *"
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ASIOChannelInfo channelInfos[2 * NUM_IN_OUT_CHANNELS];
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bool bASIOPostOutput;
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ASIOCallbacks asioCallbacks;
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int iASIOBufferSizeMono;
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int iASIOBufferSizeStereo;
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CVector<int16_t> vecsTmpAudioSndCrdStereo;
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QMutex ASIOMutex;
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// TEST
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// pointer to our sound object
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CSound* pSound;
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@ -115,7 +98,7 @@ QString CSound::SetDev ( const int iNewDev )
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// the first available driver in the system. If this fails, too, we
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// throw an error that no driver is available -> it does not make
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// sense to start the llcon software if no audio hardware is
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// available
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// available.
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if ( !LoadAndInitializeDriver ( iNewDev ).isEmpty() )
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{
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// loading and initializing the new driver failed, try to find
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@ -257,43 +240,58 @@ QString CSound::CheckDeviceCapabilities()
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// return error string
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return tr ( "The audio device does not support the "
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"required number of channels. The required number of channels "
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"is: " ) + QString().setNum ( NUM_IN_OUT_CHANNELS );
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"for input and output is: " ) +
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QString().setNum ( NUM_IN_OUT_CHANNELS );
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}
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// check sample format
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for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
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// clip number of input/output channels to our maximum
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if ( lNumInChan > MAX_NUM_IN_OUT_CHANNELS )
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{
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// check all used input and output channels
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channelInfos[i].channel = i % NUM_IN_OUT_CHANNELS;
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if ( i < NUM_IN_OUT_CHANNELS )
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{
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channelInfos[i].isInput = ASIOTrue;
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lNumInChan = MAX_NUM_IN_OUT_CHANNELS;
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}
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else
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if ( lNumOutChan > MAX_NUM_IN_OUT_CHANNELS )
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{
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channelInfos[i].isInput = ASIOFalse;
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lNumOutChan = MAX_NUM_IN_OUT_CHANNELS;
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}
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ASIOGetChannelInfo ( &channelInfos[i] );
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// check supported sample formats
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if ( ( channelInfos[i].type != ASIOSTInt16LSB ) &&
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( channelInfos[i].type != ASIOSTInt24LSB ) &&
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( channelInfos[i].type != ASIOSTInt32LSB ) &&
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( channelInfos[i].type != ASIOSTFloat32LSB ) &&
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( channelInfos[i].type != ASIOSTFloat64LSB ) &&
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( channelInfos[i].type != ASIOSTInt32LSB16 ) &&
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( channelInfos[i].type != ASIOSTInt32LSB18 ) &&
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( channelInfos[i].type != ASIOSTInt32LSB20 ) &&
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( channelInfos[i].type != ASIOSTInt32LSB24 ) &&
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( channelInfos[i].type != ASIOSTInt16MSB ) &&
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( channelInfos[i].type != ASIOSTInt24MSB ) &&
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( channelInfos[i].type != ASIOSTInt32MSB ) &&
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( channelInfos[i].type != ASIOSTFloat32MSB ) &&
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( channelInfos[i].type != ASIOSTFloat64MSB ) &&
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( channelInfos[i].type != ASIOSTInt32MSB16 ) &&
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( channelInfos[i].type != ASIOSTInt32MSB18 ) &&
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( channelInfos[i].type != ASIOSTInt32MSB20 ) &&
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( channelInfos[i].type != ASIOSTInt32MSB24 ) )
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// query channel infos for all available input channels
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for ( int i = 0; i < lNumInChan; i++ )
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{
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// setup for input channels
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channelInfosInput[i].isInput = ASIOTrue;
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channelInfosInput[i].channel = i;
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ASIOGetChannelInfo ( &channelInfosInput[i] );
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// Check supported sample formats.
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// Actually, it would be enough to have at least two channels which
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// support the required sample format. But since we have support for
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// all known sample types, the following check should always pass and
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// therefore we throw the error message on any channel which does not
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// fullfill the sample format requirement (quick hack solution).
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if ( !CheckSampleTypeSupported ( channelInfosInput[i].type ) )
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{
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// return error string
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return tr ( "Required audio sample format not available." );
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}
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}
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// query channel infos for all available output channels
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for ( int i = 0; i < lNumOutChan; i++ )
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{
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// setup for output channels
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channelInfosOutput[i].isInput = ASIOFalse;
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channelInfosOutput[i].channel = i;
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ASIOGetChannelInfo ( &channelInfosOutput[i] );
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// Check supported sample formats.
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// Actually, it would be enough to have at least two channels which
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// support the required sample format. But since we have support for
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// all known sample types, the following check should always pass and
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// therefore we throw the error message on any channel which does not
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// fullfill the sample format requirement (quick hack solution).
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if ( !CheckSampleTypeSupported ( channelInfosOutput[i].type ) )
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{
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// return error string
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return tr ( "Required audio sample format not available." );
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@ -435,6 +433,24 @@ int CSound::Init ( const int iNewPrefMonoBufferSize )
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// create and activate ASIO buffers (buffer size in samples),
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// dispose old buffers (if any)
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ASIODisposeBuffers();
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// init buffer infos, we always want to have two input and
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// two output channels
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for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
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{
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// prepare input channels
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bufferInfos[i].isInput = ASIOTrue;
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bufferInfos[i].channelNum = vSelectedInputChannels[i];
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bufferInfos[i].buffers[0] = 0;
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bufferInfos[i].buffers[1] = 0;
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// prepare output channels
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bufferInfos[NUM_IN_OUT_CHANNELS + i].isInput = ASIOFalse;
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bufferInfos[NUM_IN_OUT_CHANNELS + i].channelNum = vSelectedOutputChannels[i];
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bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[0] = 0;
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bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[1] = 0;
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}
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ASIOCreateBuffers ( bufferInfos,
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2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
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iASIOBufferSizeMono, &asioCallbacks );
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@ -467,14 +483,15 @@ void CSound::Stop()
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}
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CSound::CSound ( void (*fpNewCallback) ( CVector<int16_t>& psData, void* arg ), void* arg ) :
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CSoundBase ( true, fpNewCallback, arg )
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CSoundBase ( true, fpNewCallback, arg ),
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vSelectedInputChannels ( NUM_IN_OUT_CHANNELS ),
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vSelectedOutputChannels ( NUM_IN_OUT_CHANNELS )
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{
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int i;
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// TEST
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// init pointer to our sound object
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pSound = this;
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// get available ASIO driver names in system
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for ( i = 0; i < MAX_NUMBER_SOUND_CARDS; i++ )
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{
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@ -500,22 +517,12 @@ pSound = this;
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// init device index with illegal value to show that driver is not initialized
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lCurDev = -1;
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// init buffer infos, we always want to have two input and
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// two output channels
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for ( i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
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{
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// prepare input channels
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bufferInfos[i].isInput = ASIOTrue;
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bufferInfos[i].channelNum = i;
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bufferInfos[i].buffers[0] = 0;
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bufferInfos[i].buffers[1] = 0;
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// prepare output channels
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bufferInfos[NUM_IN_OUT_CHANNELS + i].isInput = ASIOFalse;
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bufferInfos[NUM_IN_OUT_CHANNELS + i].channelNum = i;
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bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[0] = 0;
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bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[1] = 0;
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}
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// init selected channel numbers with defaults: use first available
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// channels for input and output
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vSelectedInputChannels[0] = 0;
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vSelectedInputChannels[1] = 1;
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vSelectedOutputChannels[0] = 0;
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vSelectedOutputChannels[1] = 1;
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// set up the asioCallback structure
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asioCallbacks.bufferSwitch = &bufferSwitch;
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@ -542,336 +549,357 @@ ASIOTime* CSound::bufferSwitchTimeInfo ( ASIOTime* timeInfo,
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return 0L;
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}
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bool CSound::CheckSampleTypeSupported ( const ASIOSampleType SamType )
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{
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// check for supported sample types
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return ( ( SamType == ASIOSTInt16LSB ) ||
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( SamType == ASIOSTInt24LSB ) ||
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( SamType == ASIOSTInt32LSB ) ||
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( SamType == ASIOSTFloat32LSB ) ||
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( SamType == ASIOSTFloat64LSB ) ||
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( SamType == ASIOSTInt32LSB16 ) ||
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( SamType == ASIOSTInt32LSB18 ) ||
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( SamType == ASIOSTInt32LSB20 ) ||
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( SamType == ASIOSTInt32LSB24 ) ||
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( SamType == ASIOSTInt16MSB ) ||
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( SamType == ASIOSTInt24MSB ) ||
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( SamType == ASIOSTInt32MSB ) ||
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( SamType == ASIOSTFloat32MSB ) ||
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( SamType == ASIOSTFloat64MSB ) ||
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( SamType == ASIOSTInt32MSB16 ) ||
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( SamType == ASIOSTInt32MSB18 ) ||
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( SamType == ASIOSTInt32MSB20 ) ||
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( SamType == ASIOSTInt32MSB24 ) );
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}
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void CSound::bufferSwitch ( long index, ASIOBool processNow )
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{
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int iCurSample;
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ASIOMutex.lock(); // get mutex lock
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{
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// perform the processing for input and output
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for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ ) // stereo
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pSound->ASIOMutex.lock(); // get mutex lock
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{
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if ( bufferInfos[i].isInput == ASIOTrue )
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// CAPTURE -------------------------------------------------------------
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for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
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{
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// CAPTURE -----------------------------------------------------
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const int iInChNum = i;
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// copy new captured block in thread transfer buffer (copy
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// mono data interleaved in stereo buffer)
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switch ( channelInfos[i].type )
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switch ( pSound->channelInfosInput[pSound->vSelectedInputChannels[i]].type )
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{
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case ASIOSTInt16LSB:
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// no type conversion required, just copy operation
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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static_cast<int16_t*> ( bufferInfos[i].buffers[index] )[iCurSample];
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample];
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}
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break;
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case ASIOSTInt24LSB:
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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int iCurSam = 0;
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memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
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memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
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iCurSam >>= 8;
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( iCurSam );
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}
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break;
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case ASIOSTInt32LSB:
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( static_cast<int32_t*> (
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bufferInfos[i].buffers[index] )[iCurSample] >> 16 );
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pSound->bufferInfos[i].buffers[index] )[iCurSample] >> 16 );
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}
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break;
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case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( static_cast<float*> (
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bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
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pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
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}
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break;
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case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( static_cast<double*> (
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bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
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pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
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}
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break;
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case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( static_cast<int32_t*> (
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bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFF );
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFF );
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}
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break;
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case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( ( static_cast<int32_t*> (
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bufferInfos[i].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
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}
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break;
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case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( ( static_cast<int32_t*> (
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bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
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}
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break;
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case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( ( static_cast<int32_t*> (
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bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
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}
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break;
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case ASIOSTInt16MSB:
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// NOT YET TESTED
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// flip bits
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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Flip16Bits ( ( static_cast<int16_t*> (
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bufferInfos[i].buffers[index] ) )[iCurSample] );
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pSound->bufferInfos[i].buffers[index] ) )[iCurSample] );
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}
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break;
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case ASIOSTInt24MSB:
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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{
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// because the bits are flipped, we do not have to perform the
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// shift by 8 bits
|
||||
int iCurSam = 0;
|
||||
memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
|
||||
memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
|
||||
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB:
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// flip bits and convert to 16 bit
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
|
||||
bufferInfos[i].buffers[index] )[iCurSample] ) >> 16 );
|
||||
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) >> 16 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
static_cast<int16_t> ( static_cast<float> (
|
||||
Flip32Bits ( static_cast<int32_t*> (
|
||||
bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
|
||||
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
static_cast<int16_t> ( static_cast<double> (
|
||||
Flip64Bits ( static_cast<int64_t*> (
|
||||
bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
|
||||
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
|
||||
bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFF );
|
||||
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFF );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
|
||||
bufferInfos[i].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
|
||||
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
|
||||
bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
|
||||
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
|
||||
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
|
||||
bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
|
||||
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
|
||||
}
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// call processing callback function
|
||||
pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
|
||||
pSound->ProcessCallback ( pSound->vecsTmpAudioSndCrdStereo );
|
||||
|
||||
// perform the processing for input and output
|
||||
for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ ) // stereo
|
||||
// PLAYBACK ------------------------------------------------------------
|
||||
for ( int i = NUM_IN_OUT_CHANNELS; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
|
||||
{
|
||||
if ( bufferInfos[i].isInput != ASIOTrue )
|
||||
{
|
||||
// PLAYBACK ----------------------------------------------------
|
||||
const int iOutChNum = i - NUM_IN_OUT_CHANNELS;
|
||||
|
||||
// copy data from sound card in output buffer (copy
|
||||
// interleaved stereo data in mono sound card buffer)
|
||||
switch ( channelInfos[i].type )
|
||||
switch ( pSound->channelInfosOutput[pSound->vSelectedOutputChannels[iOutChNum]].type )
|
||||
{
|
||||
case ASIOSTInt16LSB:
|
||||
// no type conversion required, just copy operation
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
static_cast<int16_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum];
|
||||
static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum];
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt24LSB:
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert current sample in 24 bit format
|
||||
int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
iCurSam <<= 8;
|
||||
|
||||
memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
|
||||
memcpy ( ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32LSB:
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
( iCurSam << 16 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
const float fCurSam = static_cast<float> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
fCurSam / _MAXSHORT;
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
const double fCurSam = static_cast<double> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<double*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<double*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
fCurSam / _MAXSHORT;
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
iCurSam;
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
( iCurSam << 2 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
( iCurSam << 4 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
( iCurSam << 8 );
|
||||
}
|
||||
break;
|
||||
|
|
@ -879,47 +907,47 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
|
|||
case ASIOSTInt16MSB:
|
||||
// NOT YET TESTED
|
||||
// flip bits
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
( (int16_t*) bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
Flip16Bits ( vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
( (int16_t*) pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
Flip16Bits ( pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt24MSB:
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// because the bits are flipped, we do not have to perform the
|
||||
// shift by 8 bits
|
||||
int32_t iCurSam = static_cast<int32_t> ( Flip16Bits (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] ) );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ) );
|
||||
|
||||
memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
|
||||
memcpy ( ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB:
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit and flip bits
|
||||
int iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
Flip32Bits ( iCurSam << 16 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
const float fCurSam = static_cast<float> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<float> ( Flip32Bits ( static_cast<int32_t> (
|
||||
fCurSam / _MAXSHORT ) ) );
|
||||
}
|
||||
|
|
@ -927,12 +955,12 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
|
|||
|
||||
case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
const double fCurSam = static_cast<double> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<double> ( Flip64Bits ( static_cast<int64_t> (
|
||||
fCurSam / _MAXSHORT ) ) );
|
||||
}
|
||||
|
|
@ -940,67 +968,66 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
|
|||
|
||||
case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
Flip32Bits ( iCurSam );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
Flip32Bits ( iCurSam << 2 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
Flip32Bits ( iCurSam << 4 );
|
||||
}
|
||||
break;
|
||||
|
||||
case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
|
||||
// NOT YET TESTED
|
||||
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
||||
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
|
||||
{
|
||||
// convert to 32 bit
|
||||
const int32_t iCurSam = static_cast<int32_t> (
|
||||
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
|
||||
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
|
||||
|
||||
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
|
||||
Flip32Bits ( iCurSam << 8 );
|
||||
}
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// finally if the driver supports the ASIOOutputReady() optimization,
|
||||
// do it here, all data are in place -----------------------------------
|
||||
if ( bASIOPostOutput )
|
||||
if ( pSound->bASIOPostOutput )
|
||||
{
|
||||
ASIOOutputReady();
|
||||
}
|
||||
}
|
||||
ASIOMutex.unlock();
|
||||
pSound->ASIOMutex.unlock();
|
||||
}
|
||||
|
||||
long CSound::asioMessages ( long selector,
|
||||
|
|
|
|||
|
|
@ -40,14 +40,14 @@
|
|||
|
||||
|
||||
/* Definitions ****************************************************************/
|
||||
#define NUM_IN_OUT_CHANNELS 2 /* Stereo recording (but we only
|
||||
use one channel for recording) */
|
||||
// stereo for input and output
|
||||
#define NUM_IN_OUT_CHANNELS 2
|
||||
|
||||
#define MAX_SND_BUF_IN 100
|
||||
#define MAX_SND_BUF_OUT 100
|
||||
|
||||
#define NUM_SOUND_BUFFERS_IN 4
|
||||
#define NUM_SOUND_BUFFERS_OUT 4
|
||||
// define the maximum number of audio channel for input/output we can store
|
||||
// channel infos for (and therefore this is the maximum number of entries in
|
||||
// the channel selection combo box regardless of the actual available number
|
||||
// of channels by the audio device)
|
||||
#define MAX_NUM_IN_OUT_CHANNELS 32
|
||||
|
||||
|
||||
/* Classes ********************************************************************/
|
||||
|
|
@ -74,6 +74,17 @@ protected:
|
|||
QString LoadAndInitializeDriver ( int iIdx );
|
||||
int GetActualBufferSize ( const int iDesiredBufferSizeMono );
|
||||
QString CheckDeviceCapabilities();
|
||||
bool CheckSampleTypeSupported ( const ASIOSampleType SamType );
|
||||
|
||||
int iASIOBufferSizeMono;
|
||||
int iASIOBufferSizeStereo;
|
||||
|
||||
CVector<int> vSelectedInputChannels;
|
||||
CVector<int> vSelectedOutputChannels;
|
||||
|
||||
CVector<int16_t> vecsTmpAudioSndCrdStereo;
|
||||
|
||||
QMutex ASIOMutex;
|
||||
|
||||
// utility functions
|
||||
static int16_t Flip16Bits ( const int16_t iIn );
|
||||
|
|
@ -89,6 +100,14 @@ protected:
|
|||
long lGranularity;
|
||||
} HWBufferInfo;
|
||||
|
||||
// ASIO stuff
|
||||
ASIODriverInfo driverInfo;
|
||||
ASIOBufferInfo bufferInfos[2 * NUM_IN_OUT_CHANNELS]; // for input and output buffers -> "2 *"
|
||||
ASIOChannelInfo channelInfosInput[MAX_NUM_IN_OUT_CHANNELS];
|
||||
ASIOChannelInfo channelInfosOutput[MAX_NUM_IN_OUT_CHANNELS];
|
||||
bool bASIOPostOutput;
|
||||
ASIOCallbacks asioCallbacks;
|
||||
|
||||
// callbacks
|
||||
static void bufferSwitch ( long index, ASIOBool processNow );
|
||||
static ASIOTime* bufferSwitchTimeInfo ( ASIOTime* timeInfo, long index, ASIOBool processNow );
|
||||
|
|
|
|||
Loading…
Reference in a new issue