1082 lines
43 KiB
C++
Executable File
1082 lines
43 KiB
C++
Executable File
/******************************************************************************\
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* Copyright (c) 2004-2010
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*
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* Author(s):
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* Volker Fischer
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*
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* Description:
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* Sound card interface for Windows operating systems
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*
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******************************************************************************
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*
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* This program is free software; you can redistribute it and/or modify it under
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* the terms of the GNU General Public License as published by the Free Software
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* Foundation; either version 2 of the License, or (at your option) any later
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* version.
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*
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
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* FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
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* details.
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*
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* You should have received a copy of the GNU General Public License along with
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* this program; if not, write to the Free Software Foundation, Inc.,
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* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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\******************************************************************************/
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#include "Sound.h"
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/* Implementation *************************************************************/
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// external references
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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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CSound* pSound;
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/******************************************************************************\
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* Common *
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\******************************************************************************/
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QString CSound::SetDev ( const int iNewDev )
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{
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QString strReturn = ""; // init with no error
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bool bTryLoadAnyDriver = false;
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// check if an ASIO driver was already initialized
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if ( lCurDev >= 0 )
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{
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// a device was already been initialized and is used, first clean up
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// dispose ASIO buffers
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ASIODisposeBuffers();
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// remove old driver
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ASIOExit();
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asioDrivers->removeCurrentDriver();
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const QString strErrorMessage = LoadAndInitializeDriver ( iNewDev );
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if ( !strErrorMessage.isEmpty() )
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{
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if ( iNewDev != lCurDev )
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{
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// loading and initializing the new driver failed, go back to
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// original driver and display error message
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LoadAndInitializeDriver ( lCurDev );
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}
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else
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{
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// the same driver is used but the driver properties seems to
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// have changed so that they are not compatible to our
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// software anymore
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QMessageBox::critical (
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0, APP_NAME, QString ( tr ( "The audio driver properties "
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"have changed to a state which is incompatible to this "
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"software. The selected audio device could not be used "
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"because of the following error: <b>" ) ) +
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strErrorMessage +
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QString ( tr ( "</b><br><br>Please restart the software." ) ),
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"Close", 0 );
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_exit ( 0 );
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}
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// store error return message
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strReturn = strErrorMessage;
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}
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Init ( iASIOBufferSizeStereo );
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}
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else
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{
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if ( iNewDev != INVALID_SNC_CARD_DEVICE )
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{
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// This is the first time a driver is to be initialized, we first
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// try to load the selected driver, if this fails, we try to load
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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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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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// at least one usable driver
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bTryLoadAnyDriver = true;
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}
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}
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else
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{
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// try to find one usable driver (select the first valid driver)
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bTryLoadAnyDriver = true;
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}
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}
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if ( bTryLoadAnyDriver )
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{
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// try to load and initialize any valid driver
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QVector<QString> vsErrorList =
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LoadAndInitializeFirstValidDriver();
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if ( !vsErrorList.isEmpty() )
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{
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// create error message with all details
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QString sErrorMessage = tr ( "<b>No usable ASIO audio device "
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"(driver) found.</b><br><br>"
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"In the following there is a list of all available drivers "
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"with the associated error message:<ul>" );
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for ( int i = 0; i < lNumDevs; i++ )
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{
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sErrorMessage += "<li><b>" + GetDeviceName ( i ) + "</b>: " +
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vsErrorList[i] + "</li>";
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}
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sErrorMessage += "</ul>";
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throw CGenErr ( sErrorMessage );
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}
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}
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return strReturn;
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}
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QVector<QString> CSound::LoadAndInitializeFirstValidDriver()
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{
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QVector<QString> vsErrorList;
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// load and initialize first valid ASIO driver
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bool bValidDriverDetected = false;
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int iCurDriverIdx = 0;
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// try all available drivers in the system ("lNumDevs" devices)
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while ( !bValidDriverDetected && ( iCurDriverIdx < lNumDevs ) )
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{
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// try to load and initialize current driver, store error message
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const QString strCurError =
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LoadAndInitializeDriver ( iCurDriverIdx );
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vsErrorList.append ( strCurError );
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if ( strCurError.isEmpty() )
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{
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// initialization was successful
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bValidDriverDetected = true;
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// store ID of selected driver
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lCurDev = iCurDriverIdx;
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// empty error list shows that init was successful
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vsErrorList.clear();
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}
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// try next driver
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iCurDriverIdx++;
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}
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return vsErrorList;
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}
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QString CSound::LoadAndInitializeDriver ( int iDriverIdx )
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{
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// first check and correct input parameter
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if ( iDriverIdx >= lNumDevs )
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{
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// we assume here that at least one driver is in the system
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iDriverIdx = 0;
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}
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// load driver
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loadAsioDriver ( cDriverNames[iDriverIdx] );
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if ( ASIOInit ( &driverInfo ) != ASE_OK )
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{
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// clean up and return error string
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asioDrivers->removeCurrentDriver();
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return tr ( "The audio driver could not be initialized." );
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}
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// check device capabilities if it fullfills our requirements
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const QString strStat = CheckDeviceCapabilities();
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// store ID of selected driver if initialization was successful
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if ( strStat.isEmpty() )
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{
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lCurDev = iDriverIdx;
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}
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else
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{
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// driver cannot be used, clean up
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asioDrivers->removeCurrentDriver();
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}
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return strStat;
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}
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QString CSound::CheckDeviceCapabilities()
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{
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// This function checks if our required input/output channel
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// properties are supported by the selected device. If the return
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// string is empty, the device can be used, otherwise the error
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// message is returned.
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// check the sample rate
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const ASIOError CanSaRateReturn = ASIOCanSampleRate ( SYSTEM_SAMPLE_RATE );
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if ( ( CanSaRateReturn == ASE_NoClock ) ||
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( CanSaRateReturn == ASE_NotPresent ) )
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{
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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 sample rate. The required sample rate is: " ) +
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QString().setNum ( SYSTEM_SAMPLE_RATE ) + " Hz";
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}
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// check the number of available channels
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long lNumInChan;
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long lNumOutChan;
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ASIOGetChannels ( &lNumInChan, &lNumOutChan );
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if ( ( lNumInChan < NUM_IN_OUT_CHANNELS ) ||
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( lNumOutChan < NUM_IN_OUT_CHANNELS ) )
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{
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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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}
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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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{
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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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}
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else
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{
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channelInfos[i].isInput = ASIOFalse;
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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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{
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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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// everything is ok, return empty string for "no error" case
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return "";
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}
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int CSound::GetActualBufferSize ( const int iDesiredBufferSizeMono )
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{
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int iActualBufferSizeMono;
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// query the usable buffer sizes
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ASIOGetBufferSize ( &HWBufferInfo.lMinSize,
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&HWBufferInfo.lMaxSize,
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&HWBufferInfo.lPreferredSize,
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&HWBufferInfo.lGranularity );
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// calculate "nearest" buffer size and set internal parameter accordingly
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// first check minimum and maximum values
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if ( iDesiredBufferSizeMono <= HWBufferInfo.lMinSize )
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{
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iActualBufferSizeMono = HWBufferInfo.lMinSize;
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}
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else
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{
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if ( iDesiredBufferSizeMono >= HWBufferInfo.lMaxSize )
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{
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iActualBufferSizeMono = HWBufferInfo.lMaxSize;
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}
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else
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{
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// ASIO SDK 2.2: "Notes: When minimum and maximum buffer size are
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// equal, the preferred buffer size has to be the same value as
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// well; granularity should be 0 in this case."
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if ( HWBufferInfo.lMinSize == HWBufferInfo.lMaxSize )
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{
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iActualBufferSizeMono = HWBufferInfo.lMinSize;
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}
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else
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{
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if ( ( HWBufferInfo.lGranularity < -1 ) ||
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( HWBufferInfo.lGranularity == 0 ) )
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{
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// Special case (seen for EMU audio cards): granularity is
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// zero or less than zero (make sure to exclude the special
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// case of -1).
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// There is no definition of this case in the ASIO SDK
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// document. We assume here that all buffer sizes in between
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// minimum and maximum buffer sizes are allowed.
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iActualBufferSizeMono = iDesiredBufferSizeMono;
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}
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else
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{
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// General case --------------------------------------------
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// initialization
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int iTrialBufSize = HWBufferInfo.lMinSize;
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int iLastTrialBufSize = HWBufferInfo.lMinSize;
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bool bSizeFound = false;
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// test loop
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while ( ( iTrialBufSize <= HWBufferInfo.lMaxSize ) && ( !bSizeFound ) )
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{
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if ( iTrialBufSize >= iDesiredBufferSizeMono )
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{
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// test which buffer size fits better: the old one or the
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// current one
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if ( ( iTrialBufSize - iDesiredBufferSizeMono ) >
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( iDesiredBufferSizeMono - iLastTrialBufSize ) )
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{
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iTrialBufSize = iLastTrialBufSize;
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}
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// exit while loop
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bSizeFound = true;
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}
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if ( !bSizeFound )
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{
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// store old trial buffer size
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iLastTrialBufSize = iTrialBufSize;
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// increment trial buffer size (check for special
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// case first)
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if ( HWBufferInfo.lGranularity == -1 )
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{
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// special case: buffer sizes are a power of 2
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iTrialBufSize *= 2;
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}
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else
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{
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iTrialBufSize += HWBufferInfo.lGranularity;
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}
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}
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}
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// clip trial buffer size (it may happen in the while
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// routine that "iTrialBufSize" is larger than "lMaxSize" in
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// case "lMaxSize - lMinSize" is not divisible by the
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// granularity)
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if ( iTrialBufSize > HWBufferInfo.lMaxSize )
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{
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iTrialBufSize = HWBufferInfo.lMaxSize;
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}
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// set ASIO buffer size
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iActualBufferSizeMono = iTrialBufSize;
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}
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}
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}
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}
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return iActualBufferSizeMono;
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}
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int CSound::Init ( const int iNewPrefMonoBufferSize )
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{
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ASIOMutex.lock(); // get mutex lock
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{
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// get the actual sound card buffer size which is supported
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// by the audio hardware
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iASIOBufferSizeMono =
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GetActualBufferSize ( iNewPrefMonoBufferSize );
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// init base clasee
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CSoundBase::Init ( iASIOBufferSizeMono );
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// set internal buffer size value and calculate stereo buffer size
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iASIOBufferSizeStereo = 2 * iASIOBufferSizeMono;
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// set the sample rate
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ASIOSetSampleRate ( SYSTEM_SAMPLE_RATE );
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// create memory for intermediate audio buffer
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vecsTmpAudioSndCrdStereo.Init ( iASIOBufferSizeStereo );
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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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ASIOCreateBuffers ( bufferInfos,
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2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
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iASIOBufferSizeMono, &asioCallbacks );
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// check wether the driver requires the ASIOOutputReady() optimization
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// (can be used by the driver to reduce output latency by one block)
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bASIOPostOutput = ( ASIOOutputReady() == ASE_OK );
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}
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ASIOMutex.unlock();
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return iASIOBufferSizeMono;
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}
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void CSound::Start()
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{
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// start audio
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ASIOStart();
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// call base class
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CSoundBase::Start();
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}
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void CSound::Stop()
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{
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// stop audio
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ASIOStop();
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// call base class
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CSoundBase::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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{
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int i;
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// TEST
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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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cDriverNames[i] = new char[32];
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}
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loadAsioDriver ( "dummy" ); // to initialize external object
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lNumDevs = asioDrivers->getDriverNames ( cDriverNames, MAX_NUMBER_SOUND_CARDS );
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// in case we do not have a driver available, throw error
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if ( lNumDevs == 0 )
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{
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throw CGenErr ( tr ( "<b>No ASIO audio device (driver) found.</b><br><br>"
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"The " ) + APP_NAME + tr ( " software requires the low latency audio "
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"interface <b>ASIO</b> to work properly. This is no standard "
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"Windows audio interface and therefore a special audio driver is "
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"required. Either your sound card has a native ASIO driver (which "
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"is recommended) or you might want to use alternative drivers like "
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"the ASIO4All or kX driver." ) );
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}
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asioDrivers->removeCurrentDriver();
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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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// set up the asioCallback structure
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asioCallbacks.bufferSwitch = &bufferSwitch;
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asioCallbacks.sampleRateDidChange = &sampleRateChanged;
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asioCallbacks.asioMessage = &asioMessages;
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asioCallbacks.bufferSwitchTimeInfo = &bufferSwitchTimeInfo;
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}
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CSound::~CSound()
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{
|
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// cleanup ASIO stuff
|
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ASIOStop();
|
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ASIODisposeBuffers();
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ASIOExit();
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asioDrivers->removeCurrentDriver();
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}
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|
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// ASIO callbacks -------------------------------------------------------------
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ASIOTime* CSound::bufferSwitchTimeInfo ( ASIOTime* timeInfo,
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long index,
|
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ASIOBool processNow )
|
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{
|
|
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;
|
|
}
|