jamulus/windows/sound.cpp
2010-02-04 20:25:57 +00:00

1057 lines
42 KiB
C++
Executable file

/******************************************************************************\
* Copyright (c) 2004-2010
*
* Author(s):
* Volker Fischer
*
* Description:
* Sound card interface for Windows operating systems
*
******************************************************************************
*
* This program is free software; you can redistribute it and/or modify it under
* the terms of the GNU General Public License as published by the Free Software
* Foundation; either version 2 of the License, or (at your option) any later
* version.
*
* This program is distributed in the hope that it will be useful, but WITHOUT
* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
* FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
* details.
*
* You should have received a copy of the GNU General Public License along with
* this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
\******************************************************************************/
#include "Sound.h"
/* Implementation *************************************************************/
// external references
extern AsioDrivers* asioDrivers;
bool loadAsioDriver ( char *name );
// TODO the following variables should be in the class definition but we cannot
// do it here since we have static callback functions which cannot access the
// class members :-(((
// ASIO stuff
ASIODriverInfo driverInfo;
ASIOBufferInfo bufferInfos[2 * NUM_IN_OUT_CHANNELS]; // for input and output buffers -> "2 *"
ASIOChannelInfo channelInfos[2 * NUM_IN_OUT_CHANNELS];
bool bASIOPostOutput;
ASIOCallbacks asioCallbacks;
int iASIOBufferSizeMono;
int iASIOBufferSizeStereo;
CVector<int16_t> vecsTmpAudioSndCrdStereo;
QMutex ASIOMutex;
// TEST
CSound* pSound;
/******************************************************************************\
* Common *
\******************************************************************************/
QString CSound::SetDev ( const int iNewDev )
{
QString strReturn = ""; // init with no error
bool bTryLoadAnyDriver = false;
// check if an ASIO driver was already initialized
if ( lCurDev >= 0 )
{
// a device was already been initialized and is used, first clean up
// dispose ASIO buffers
ASIODisposeBuffers();
// remove old driver
ASIOExit();
asioDrivers->removeCurrentDriver();
const QString strErrorMessage = LoadAndInitializeDriver ( iNewDev );
if ( !strErrorMessage.isEmpty() )
{
if ( iNewDev != lCurDev )
{
// loading and initializing the new driver failed, go back to
// original driver and display error message
LoadAndInitializeDriver ( lCurDev );
}
else
{
// the same driver is used but the driver properties seems to
// have changed so that they are not compatible to our
// software anymore
QMessageBox::critical (
0, APP_NAME, QString ( tr ( "The audio driver properties "
"have changed to a state which is incompatible to this "
"software. The selected audio device could not be used "
"because of the following error: <b>" ) ) +
strErrorMessage +
QString ( tr ( "</b><br><br>Please restart the software." ) ),
"Close", 0 );
_exit ( 0 );
}
// store error return message
strReturn = strErrorMessage;
}
Init ( iASIOBufferSizeStereo );
}
else
{
if ( iNewDev != INVALID_SNC_CARD_DEVICE )
{
// This is the first time a driver is to be initialized, we first
// try to load the selected driver, if this fails, we try to load
// the first available driver in the system. If this fails, too, we
// throw an error that no driver is available -> it does not make
// sense to start the llcon software if no audio hardware is
// available
if ( !LoadAndInitializeDriver ( iNewDev ).isEmpty() )
{
// loading and initializing the new driver failed, try to find
// at least one usable driver
bTryLoadAnyDriver = true;
}
}
else
{
// try to find one usable driver (select the first valid driver)
bTryLoadAnyDriver = true;
}
}
if ( bTryLoadAnyDriver )
{
// try to load and initialize any valid driver
QVector<QString> vsErrorList =
LoadAndInitializeFirstValidDriver();
if ( !vsErrorList.isEmpty() )
{
// create error message with all details
QString sErrorMessage = tr ( "<b>No usable ASIO audio device "
"(driver) found.</b><br><br>"
"In the following there is a list of all available drivers "
"with the associated error message:<ul>" );
for ( int i = 0; i < lNumDevs; i++ )
{
sErrorMessage += "<li><b>" + GetDeviceName ( i ) + "</b>: " +
vsErrorList[i] + "</li>";
}
sErrorMessage += "</ul>";
throw CGenErr ( sErrorMessage );
}
}
return strReturn;
}
QVector<QString> CSound::LoadAndInitializeFirstValidDriver()
{
QVector<QString> vsErrorList;
// load and initialize first valid ASIO driver
bool bValidDriverDetected = false;
int iCurDriverIdx = 0;
// try all available drivers in the system ("lNumDevs" devices)
while ( !bValidDriverDetected && ( iCurDriverIdx < lNumDevs ) )
{
// try to load and initialize current driver, store error message
const QString strCurError =
LoadAndInitializeDriver ( iCurDriverIdx );
vsErrorList.append ( strCurError );
if ( strCurError.isEmpty() )
{
// initialization was successful
bValidDriverDetected = true;
// store ID of selected driver
lCurDev = iCurDriverIdx;
// empty error list shows that init was successful
vsErrorList.clear();
}
// try next driver
iCurDriverIdx++;
}
return vsErrorList;
}
QString CSound::LoadAndInitializeDriver ( int iDriverIdx )
{
// first check and correct input parameter
if ( iDriverIdx >= lNumDevs )
{
// we assume here that at least one driver is in the system
iDriverIdx = 0;
}
// load driver
loadAsioDriver ( cDriverNames[iDriverIdx] );
if ( ASIOInit ( &driverInfo ) != ASE_OK )
{
// clean up and return error string
asioDrivers->removeCurrentDriver();
return tr ( "The audio driver could not be initialized." );
}
// check device capabilities if it fullfills our requirements
const QString strStat = CheckDeviceCapabilities();
// store ID of selected driver if initialization was successful
if ( strStat.isEmpty() )
{
lCurDev = iDriverIdx;
}
else
{
// driver cannot be used, clean up
asioDrivers->removeCurrentDriver();
}
return strStat;
}
QString CSound::CheckDeviceCapabilities()
{
// This function checks if our required input/output channel
// properties are supported by the selected device. If the return
// string is empty, the device can be used, otherwise the error
// message is returned.
// check the sample rate
const ASIOError CanSaRateReturn = ASIOCanSampleRate ( SYSTEM_SAMPLE_RATE );
if ( ( CanSaRateReturn == ASE_NoClock ) ||
( CanSaRateReturn == ASE_NotPresent ) )
{
// return error string
return tr ( "The audio device does not support the "
"required sample rate. The required sample rate is: " ) +
QString().setNum ( SYSTEM_SAMPLE_RATE ) + " Hz";
}
// check the number of available channels
long lNumInChan;
long lNumOutChan;
ASIOGetChannels ( &lNumInChan, &lNumOutChan );
if ( ( lNumInChan < NUM_IN_OUT_CHANNELS ) ||
( lNumOutChan < NUM_IN_OUT_CHANNELS ) )
{
// return error string
return tr ( "The audio device does not support the "
"required number of channels. The required number of channels "
"is: " ) + QString().setNum ( NUM_IN_OUT_CHANNELS );
}
// check sample format
for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
{
// check all used input and output channels
channelInfos[i].channel = i % NUM_IN_OUT_CHANNELS;
if ( i < NUM_IN_OUT_CHANNELS )
{
channelInfos[i].isInput = ASIOTrue;
}
else
{
channelInfos[i].isInput = ASIOFalse;
}
ASIOGetChannelInfo ( &channelInfos[i] );
// check supported sample formats
if ( ( channelInfos[i].type != ASIOSTInt16LSB ) &&
( channelInfos[i].type != ASIOSTInt24LSB ) &&
( channelInfos[i].type != ASIOSTInt32LSB ) &&
( channelInfos[i].type != ASIOSTFloat32LSB ) &&
( channelInfos[i].type != ASIOSTFloat64LSB ) &&
( channelInfos[i].type != ASIOSTInt32LSB16 ) &&
( channelInfos[i].type != ASIOSTInt32LSB18 ) &&
( channelInfos[i].type != ASIOSTInt32LSB20 ) &&
( channelInfos[i].type != ASIOSTInt32LSB24 ) &&
( channelInfos[i].type != ASIOSTInt16MSB ) &&
( channelInfos[i].type != ASIOSTInt24MSB ) &&
( channelInfos[i].type != ASIOSTInt32MSB ) &&
( channelInfos[i].type != ASIOSTFloat32MSB ) &&
( channelInfos[i].type != ASIOSTFloat64MSB ) &&
( channelInfos[i].type != ASIOSTInt32MSB16 ) &&
( channelInfos[i].type != ASIOSTInt32MSB18 ) &&
( channelInfos[i].type != ASIOSTInt32MSB20 ) &&
( channelInfos[i].type != ASIOSTInt32MSB24 ) )
{
// return error string
return tr ( "Required audio sample format not available." );
}
}
// everything is ok, return empty string for "no error" case
return "";
}
int CSound::GetActualBufferSize ( const int iDesiredBufferSizeMono )
{
int iActualBufferSizeMono;
// query the usable buffer sizes
ASIOGetBufferSize ( &HWBufferInfo.lMinSize,
&HWBufferInfo.lMaxSize,
&HWBufferInfo.lPreferredSize,
&HWBufferInfo.lGranularity );
// calculate "nearest" buffer size and set internal parameter accordingly
// first check minimum and maximum values
if ( iDesiredBufferSizeMono <= HWBufferInfo.lMinSize )
{
iActualBufferSizeMono = HWBufferInfo.lMinSize;
}
else
{
if ( iDesiredBufferSizeMono >= HWBufferInfo.lMaxSize )
{
iActualBufferSizeMono = HWBufferInfo.lMaxSize;
}
else
{
// ASIO SDK 2.2: "Notes: When minimum and maximum buffer size are
// equal, the preferred buffer size has to be the same value as
// well; granularity should be 0 in this case."
if ( HWBufferInfo.lMinSize == HWBufferInfo.lMaxSize )
{
iActualBufferSizeMono = HWBufferInfo.lMinSize;
}
else
{
// General case ------------------------------------------------
// initialization
int iTrialBufSize = HWBufferInfo.lMinSize;
int iLastTrialBufSize = HWBufferInfo.lMinSize;
bool bSizeFound = false;
// test loop
while ( ( iTrialBufSize <= HWBufferInfo.lMaxSize ) && ( !bSizeFound ) )
{
if ( iTrialBufSize >= iDesiredBufferSizeMono )
{
// test which buffer size fits better: the old one or the
// current one
if ( ( iTrialBufSize - iDesiredBufferSizeMono ) >
( iDesiredBufferSizeMono - iLastTrialBufSize ) )
{
iTrialBufSize = iLastTrialBufSize;
}
// exit while loop
bSizeFound = true;
}
if ( !bSizeFound )
{
// store old trial buffer size
iLastTrialBufSize = iTrialBufSize;
// increment trial buffer size (check for special case first)
if ( HWBufferInfo.lGranularity == -1 )
{
// special case: buffer sizes are a power of 2
iTrialBufSize *= 2;
}
else
{
iTrialBufSize += HWBufferInfo.lGranularity;
}
}
}
// set ASIO buffer size
iActualBufferSizeMono = iTrialBufSize;
}
}
}
return iActualBufferSizeMono;
}
int CSound::Init ( const int iNewPrefMonoBufferSize )
{
ASIOMutex.lock(); // get mutex lock
{
// get the actual sound card buffer size which is supported
// by the audio hardware
iASIOBufferSizeMono =
GetActualBufferSize ( iNewPrefMonoBufferSize );
// init base clasee
CSoundBase::Init ( iASIOBufferSizeMono );
// set internal buffer size value and calculate stereo buffer size
iASIOBufferSizeStereo = 2 * iASIOBufferSizeMono;
// set the sample rate
ASIOSetSampleRate ( SYSTEM_SAMPLE_RATE );
// create memory for intermediate audio buffer
vecsTmpAudioSndCrdStereo.Init ( iASIOBufferSizeStereo );
// create and activate ASIO buffers (buffer size in samples),
// dispose old buffers (if any)
ASIODisposeBuffers();
ASIOCreateBuffers ( bufferInfos,
2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
iASIOBufferSizeMono, &asioCallbacks );
// check wether the driver requires the ASIOOutputReady() optimization
// (can be used by the driver to reduce output latency by one block)
bASIOPostOutput = ( ASIOOutputReady() == ASE_OK );
}
ASIOMutex.unlock();
return iASIOBufferSizeMono;
}
void CSound::Start()
{
// start audio
ASIOStart();
// call base class
CSoundBase::Start();
}
void CSound::Stop()
{
// stop audio
ASIOStop();
// call base class
CSoundBase::Stop();
}
CSound::CSound ( void (*fpNewCallback) ( CVector<int16_t>& psData, void* arg ), void* arg ) :
CSoundBase ( true, fpNewCallback, arg )
{
int i;
// TEST
pSound = this;
// get available ASIO driver names in system
for ( i = 0; i < MAX_NUMBER_SOUND_CARDS; i++ )
{
cDriverNames[i] = new char[32];
}
loadAsioDriver ( "dummy" ); // to initialize external object
lNumDevs = asioDrivers->getDriverNames ( cDriverNames, MAX_NUMBER_SOUND_CARDS );
// in case we do not have a driver available, throw error
if ( lNumDevs == 0 )
{
throw CGenErr ( tr ( "<b>No ASIO audio device (driver) found.</b><br><br>"
"The " ) + APP_NAME + tr ( " software requires the low latency audio "
"interface <b>ASIO</b> to work properly. This is no standard "
"Windows audio interface and therefore a special audio driver is "
"required. Either your sound card has a native ASIO driver (which "
"is recommended) or you might want to use alternative drivers like "
"the ASIO4All or kX driver." ) );
}
asioDrivers->removeCurrentDriver();
// init device index with illegal value to show that driver is not initialized
lCurDev = -1;
// init buffer infos, we always want to have two input and
// two output channels
for ( i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
{
// prepare input channels
bufferInfos[i].isInput = ASIOTrue;
bufferInfos[i].channelNum = i;
bufferInfos[i].buffers[0] = 0;
bufferInfos[i].buffers[1] = 0;
// prepare output channels
bufferInfos[NUM_IN_OUT_CHANNELS + i].isInput = ASIOFalse;
bufferInfos[NUM_IN_OUT_CHANNELS + i].channelNum = i;
bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[0] = 0;
bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[1] = 0;
}
// set up the asioCallback structure
asioCallbacks.bufferSwitch = &bufferSwitch;
asioCallbacks.sampleRateDidChange = &sampleRateChanged;
asioCallbacks.asioMessage = &asioMessages;
asioCallbacks.bufferSwitchTimeInfo = &bufferSwitchTimeInfo;
}
CSound::~CSound()
{
// cleanup ASIO stuff
ASIOStop();
ASIODisposeBuffers();
ASIOExit();
asioDrivers->removeCurrentDriver();
}
// ASIO callbacks -------------------------------------------------------------
ASIOTime* CSound::bufferSwitchTimeInfo ( ASIOTime* timeInfo,
long index,
ASIOBool processNow )
{
bufferSwitch ( index, processNow );
return 0L;
}
void CSound::bufferSwitch ( long index, ASIOBool processNow )
{
int iCurSample;
ASIOMutex.lock(); // get mutex lock
{
// perform the processing for input and output
for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ ) // stereo
{
if ( bufferInfos[i].isInput == ASIOTrue )
{
// CAPTURE -----------------------------------------------------
// copy new captured block in thread transfer buffer (copy
// mono data interleaved in stereo buffer)
switch ( channelInfos[i].type )
{
case ASIOSTInt16LSB:
// no type conversion required, just copy operation
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t*> ( bufferInfos[i].buffers[index] )[iCurSample];
}
break;
case ASIOSTInt24LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
int iCurSam = 0;
memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
iCurSam >>= 8;
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( iCurSam );
}
break;
case ASIOSTInt32LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] >> 16 );
}
break;
case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<float*> (
bufferInfos[i].buffers[index] )[iCurSample] * _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++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<double*> (
bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
}
break;
case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFF );
}
break;
case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
}
break;
case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
}
break;
case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
}
break;
case ASIOSTInt16MSB:
// NOT YET TESTED
// flip bits
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
Flip16Bits ( ( static_cast<int16_t*> (
bufferInfos[i].buffers[index] ) )[iCurSample] );
}
break;
case ASIOSTInt24MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// because the bits are flipped, we do not have to perform the
// shift by 8 bits
int iCurSam = 0;
memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
}
break;
case ASIOSTInt32MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// flip bits and convert to 16 bit
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
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++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<float> (
Flip32Bits ( static_cast<int32_t*> (
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++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<double> (
Flip64Bits ( static_cast<int64_t*> (
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++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
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++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
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++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
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++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
}
break;
}
}
}
// call processing callback function
pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
// perform the processing for input and output
for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ ) // stereo
{
if ( bufferInfos[i].isInput != ASIOTrue )
{
// PLAYBACK ----------------------------------------------------
// copy data from sound card in output buffer (copy
// interleaved stereo data in mono sound card buffer)
switch ( channelInfos[i].type )
{
case ASIOSTInt16LSB:
// no type conversion required, just copy operation
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
static_cast<int16_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum];
}
break;
case ASIOSTInt24LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert current sample in 24 bit format
int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
iCurSam <<= 8;
memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
}
break;
case ASIOSTInt32LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
const float fCurSam = static_cast<float> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<float*> ( 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++ )
{
const double fCurSam = static_cast<double> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<double*> ( 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++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
( iCurSam << 8 );
}
break;
case ASIOSTInt16MSB:
// NOT YET TESTED
// flip bits
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
( (int16_t*) bufferInfos[i].buffers[index] )[iCurSample] =
Flip16Bits ( vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
}
break;
case ASIOSTInt24MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < 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] ) );
memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
}
break;
case ASIOSTInt32MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit and flip bits
int iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
const float fCurSam = static_cast<float> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<float> ( Flip32Bits ( static_cast<int32_t> (
fCurSam / _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++ )
{
const double fCurSam = static_cast<double> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<double> ( Flip64Bits ( static_cast<int64_t> (
fCurSam / _MAXSHORT ) ) );
}
break;
case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( 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 )
{
ASIOOutputReady();
}
}
ASIOMutex.unlock();
}
long CSound::asioMessages ( long selector,
long value,
void* message,
double* opt )
{
long ret = 0;
switch ( selector )
{
case kAsioEngineVersion:
// return the supported ASIO version of the host application
ret = 2L; // Host ASIO implementation version, 2 or higher
break;
// both messages might be send if the buffer size changes
case kAsioBufferSizeChange:
case kAsioResetRequest:
pSound->EmitReinitRequestSignal();
ret = 1L; // 1L if request is accepted or 0 otherwise
break;
}
return ret;
}
int16_t CSound::Flip16Bits ( const int16_t iIn )
{
uint16_t iMask = ( 1 << 15 );
int16_t iOut = 0;
for ( unsigned int i = 0; i < 16; i++ )
{
// copy current bit to correct position
iOut |= ( iIn & iMask ) ? 1 : 0;
// shift out value and mask by one bit
iOut <<= 1;
iMask >>= 1;
}
return iOut;
}
int32_t CSound::Flip32Bits ( const int32_t iIn )
{
uint32_t iMask = ( static_cast<uint32_t> ( 1 ) << 31 );
int32_t iOut = 0;
for ( unsigned int i = 0; i < 32; i++ )
{
// copy current bit to correct position
iOut |= ( iIn & iMask ) ? 1 : 0;
// shift out value and mask by one bit
iOut <<= 1;
iMask >>= 1;
}
return iOut;
}
int64_t CSound::Flip64Bits ( const int64_t iIn )
{
uint64_t iMask = ( static_cast<uint64_t> ( 1 ) << 63 );
int64_t iOut = 0;
for ( unsigned int i = 0; i < 64; i++ )
{
// copy current bit to correct position
iOut |= ( iIn & iMask ) ? 1 : 0;
// shift out value and mask by one bit
iOut <<= 1;
iMask >>= 1;
}
return iOut;
}