926 lines
31 KiB
C++
Executable File
926 lines
31 KiB
C++
Executable File
/******************************************************************************\
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* Copyright (c) 2004-2009
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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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#include <qmutex.h>
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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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// mutex
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QMutex ASIOMutex;
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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 iBufferSizeMono;
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int iBufferSizeStereo;
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int iASIOBufferSizeMono;
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// event
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HANDLE m_ASIOEvent;
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// wave in
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short* psCaptureBuffer;
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int iBufferPosCapture;
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bool bCaptureBufferOverrun;
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// wave out
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short* psPlayBuffer;
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int iBufferPosPlay;
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bool bPlayBufferUnderrun;
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int iMinNumSndBuf;
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int iCurNumSndBufIn;
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int iCurNumSndBufOut;
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int iNewNumSndBufIn;
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int iNewNumSndBufOut;
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bool bSetNumSndBufToMinimumValue;
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// we must implement these functions here to get access to global variables
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int CSound::GetOutNumBuf() { return iNewNumSndBufOut; }
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int CSound::GetInNumBuf() { return iNewNumSndBufIn; }
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/******************************************************************************\
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* Wave in *
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\******************************************************************************/
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bool CSound::Read ( CVector<short>& psData )
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{
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int i;
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bool bError;
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// check if device must be opened or reinitialized
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if ( bChangParamIn )
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{
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// reinit sound interface
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Init();
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// reset flag
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bChangParamIn = false;
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}
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// wait until enough data is available
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int iWaitCount = 0;
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while ( iBufferPosCapture < iBufferSizeStereo )
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{
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if ( !bCaptureBufferOverrun )
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{
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// regular case
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WaitForSingleObject ( m_ASIOEvent, INFINITE );
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}
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else
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{
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// it seems that the buffers are too small, wait
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// just one time to avoid CPU to go up to 100% and
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// then leave this function
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if ( iWaitCount == 0 )
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{
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WaitForSingleObject ( m_ASIOEvent, INFINITE );
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iWaitCount++;
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}
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else
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{
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return true;
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}
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}
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}
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ASIOMutex.lock(); // get mutex lock
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{
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// check for buffer overrun in ASIO thread
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bError = bCaptureBufferOverrun;
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if ( bCaptureBufferOverrun )
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{
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// reset flag
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bCaptureBufferOverrun = false;
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}
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// copy data from sound card capture buffer in function output buffer
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for ( i = 0; i < iBufferSizeStereo; i++ )
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{
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psData[i] = psCaptureBuffer[i];
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}
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// move all other data in buffer
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const int iLenCopyRegion = iBufferPosCapture - iBufferSizeStereo;
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for ( i = 0; i < iLenCopyRegion; i++ )
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{
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psCaptureBuffer[i] = psCaptureBuffer[iBufferSizeStereo + i];
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}
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// adjust "current block to write" pointer
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iBufferPosCapture -= iBufferSizeStereo;
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// in case more than one buffer was ready, reset event
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ResetEvent ( m_ASIOEvent );
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}
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ASIOMutex.unlock();
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return bError;
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}
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void CSound::SetInNumBuf ( int iNewNum )
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{
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// check new parameter
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if ( ( iNewNum < MAX_SND_BUF_IN ) && ( iNewNum >= iMinNumSndBuf ) )
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{
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// change only if parameter is different
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if ( iNewNum != iNewNumSndBufIn )
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{
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iNewNumSndBufIn = iNewNum;
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bChangParamIn = true;
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}
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}
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}
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/******************************************************************************\
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* Wave out *
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\******************************************************************************/
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bool CSound::Write ( CVector<short>& psData )
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{
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bool bError;
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// check if device must be opened or reinitialized
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if ( bChangParamOut )
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{
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// reinit sound interface
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Init();
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// reset flag
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bChangParamOut = false;
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}
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ASIOMutex.lock(); // get mutex lock
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{
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// check for buffer underrun in ASIO thread
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bError = bPlayBufferUnderrun;
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if ( bPlayBufferUnderrun )
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{
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// reset flag
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bPlayBufferUnderrun = false;
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}
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// first check if enough data in buffer is available
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const int iPlayBufferLen = iCurNumSndBufOut * iBufferSizeStereo;
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if ( iBufferPosPlay + iBufferSizeStereo > iPlayBufferLen )
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{
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// buffer overrun, return error
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bError = true;
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}
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else
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{
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// copy stereo data from function input in soundcard play buffer
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for ( int i = 0; i < iBufferSizeStereo; i++ )
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{
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psPlayBuffer[iBufferPosPlay + i] = psData[i];
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}
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iBufferPosPlay += iBufferSizeStereo;
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}
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}
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ASIOMutex.unlock();
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return bError;
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}
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void CSound::SetOutNumBuf ( int iNewNum )
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{
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// check new parameter
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if ( ( iNewNum < MAX_SND_BUF_OUT ) && ( iNewNum >= iMinNumSndBuf ) )
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{
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// change only if parameter is different
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if ( iNewNum != iNewNumSndBufOut )
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{
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iNewNumSndBufOut = iNewNum;
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bChangParamOut = true;
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}
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}
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}
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/******************************************************************************\
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* Common *
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\******************************************************************************/
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void CSound::SetDev ( const int iNewDev )
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{
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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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// the new driver was not selected before, use default settings for
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// buffer sizes
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bSetNumSndBufToMinimumValue = true;
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// a device was already been initialized and is used, kill working
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// thread and clean up
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// stop driver
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ASIOStop();
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// set event to ensure that thread leaves the waiting function
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if ( m_ASIOEvent != NULL )
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{
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SetEvent ( m_ASIOEvent );
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}
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// wait for the thread to terminate
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Sleep ( 500 );
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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 std::string strErrorMessage = LoadAndInitializeDriver ( iNewDev );
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if ( !strErrorMessage.empty() )
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{
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// The new driver initializing was not successful, try to preserve
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// the old buffer settings -> this is possible, if errornous driver
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// had failed before the buffer setting was done. If it failed after
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// setting the minimum buffer sizes, the following flag modification
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// does not have any effect which means the old settings cannot be
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// recovered anymore (TODO better solution)
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bSetNumSndBufToMinimumValue = false;
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// loading and initializing the new driver failed, go back to original
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// driver and display error message
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LoadAndInitializeDriver ( lCurDev );
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Init();
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throw CGenErr ( strErrorMessage.c_str() );
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}
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Init();
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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 try
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// to load the selected driver, if this fails, we try to load the first
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// available driver in the system. If this fails, too, we throw an error
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// that no driver is available -> it does not make sense to start the llcon
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// software if no audio hardware is available
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const std::string strErrorMessage = LoadAndInitializeDriver ( iNewDev );
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if ( !strErrorMessage.empty() )
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{
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// loading and initializing the new driver failed, try to find at
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// least one usable driver
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if ( !LoadAndInitializeFirstValidDriver() )
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{
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throw CGenErr ( "No usable ASIO audio device (driver) found." );
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}
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}
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}
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else
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{
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// the new driver was not selected before, use default settings for
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// buffer sizes
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bSetNumSndBufToMinimumValue = true;
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// try to find one usable driver (select the first valid driver)
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if ( !LoadAndInitializeFirstValidDriver() )
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{
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throw CGenErr ( "No usable ASIO audio device (driver) found." );
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}
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}
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}
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}
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std::string 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 "The audio driver could not be initialized.";
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}
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const std::string strStat = PrepareDriver();
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// store ID of selected driver if initialization was successful
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if ( strStat.empty() )
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{
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lCurDev = iDriverIdx;
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}
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return strStat;
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}
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bool CSound::LoadAndInitializeFirstValidDriver()
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{
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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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if ( loadAsioDriver ( cDriverNames[iCurDriverIdx] ) )
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{
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if ( ASIOInit ( &driverInfo ) == ASE_OK )
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{
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if ( PrepareDriver().empty() )
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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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}
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else
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{
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// driver could not be loaded, free memory
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asioDrivers->removeCurrentDriver();
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}
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}
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else
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{
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// driver could not be loaded, free memory
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asioDrivers->removeCurrentDriver();
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}
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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 bValidDriverDetected;
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}
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std::string CSound::PrepareDriver()
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{
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int i;
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int iDesiredBufferSizeMono;
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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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// clean up and return error string
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ASIOExit();
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asioDrivers->removeCurrentDriver();
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return "The audio device does not support the "
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"required number of channels.";
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}
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// set the sample rate and check if sample rate is supported
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ASIOSetSampleRate ( SND_CRD_SAMPLE_RATE );
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ASIOSampleRate sampleRate;
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ASIOGetSampleRate ( &sampleRate );
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if ( sampleRate != SND_CRD_SAMPLE_RATE )
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{
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// clean up and return error string
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ASIOExit();
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asioDrivers->removeCurrentDriver();
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return "The audio device does not support the "
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"required sample rate.";
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}
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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 the desired mono buffer size
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// TEST -> put this in the GUI and implement the code for the Linux driver, too
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// setting this variable to false sets the previous behaviour
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const bool bPreferPowerOfTwoAudioBufferSize = false;
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if ( bPreferPowerOfTwoAudioBufferSize )
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{
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// use next power of 2 for desired block size mono
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iDesiredBufferSizeMono = LlconMath().NextPowerOfTwo ( iBufferSizeMono );
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}
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else
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{
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iDesiredBufferSizeMono = iBufferSizeMono;
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}
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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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iASIOBufferSizeMono = 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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iASIOBufferSizeMono = HWBufferInfo.lMaxSize;
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}
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else
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{
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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 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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// set ASIO buffer size
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iASIOBufferSizeMono = iTrialBufSize;
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}
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}
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// calculate the minimum required number of soundcard buffers
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iMinNumSndBuf = static_cast<int> (
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ceil ( static_cast<double> ( iASIOBufferSizeMono ) / iBufferSizeMono ) );
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// TODO better solution
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// For some ASIO buffer sizes, the above calculation seems not to work although
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// it should be correct. Maybe there is a misunderstanding or a bug in the
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// sound interface implementation. As a workaround, we implement a table here, to
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// get working parameters for the most common ASIO buffer settings
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// Interesting observation: only 256 samples seems to be wrong, all other tested
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// buffer sizes like 192, 512, 384, etc. are correct...
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if ( iASIOBufferSizeMono == 256 )
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{
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iMinNumSndBuf = 4;
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}
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Q_ASSERT ( iMinNumSndBuf < MAX_SND_BUF_IN );
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Q_ASSERT ( iMinNumSndBuf < MAX_SND_BUF_OUT );
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// set or just check the sound card buffer sizes
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if ( bSetNumSndBufToMinimumValue )
|
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{
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// use minimum buffer sizes as default
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iNewNumSndBufIn = iMinNumSndBuf;
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iNewNumSndBufOut = iMinNumSndBuf;
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}
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else
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{
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// correct number of sound card buffers if required
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iNewNumSndBufIn = max ( iMinNumSndBuf, iNewNumSndBufIn );
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iNewNumSndBufOut = max ( iMinNumSndBuf, iNewNumSndBufOut );
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}
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iCurNumSndBufIn = iNewNumSndBufIn;
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iCurNumSndBufOut = iNewNumSndBufOut;
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// display warning in case the ASIO buffer is too big
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if ( iMinNumSndBuf > 6 )
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{
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QMessageBox::critical ( 0, APP_NAME,
|
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QString ( "The ASIO buffer size of the selected audio driver is ") +
|
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QString().number ( iASIOBufferSizeMono ) +
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QString ( " samples which is too large. Please try to modify "
|
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"the ASIO buffer size value in your ASIO driver settings (most ASIO "
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"drivers like ASIO4All or kx driver allow to change the ASIO buffer size). "
|
|
"Recommended settings are 96 or 128 samples. Please make sure that "
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"before you try to change the ASIO driver buffer size all ASIO "
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"applications including llcon are closed." ), "Ok", 0 );
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}
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|
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// prepare input channels
|
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for ( i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
|
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{
|
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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;
|
|
}
|
|
|
|
// prepare output channels
|
|
for ( i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
|
|
{
|
|
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;
|
|
}
|
|
|
|
// create and activate ASIO buffers (buffer size in samples)
|
|
ASIOCreateBuffers ( bufferInfos, 2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
|
|
iASIOBufferSizeMono, &asioCallbacks );
|
|
|
|
// now get some buffer details
|
|
for ( i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
|
|
{
|
|
channelInfos[i].channel = bufferInfos[i].channelNum;
|
|
channelInfos[i].isInput = bufferInfos[i].isInput;
|
|
ASIOGetChannelInfo ( &channelInfos[i] );
|
|
|
|
// only 16/24/32 LSB is supported
|
|
if ( ( channelInfos[i].type != ASIOSTInt16LSB ) &&
|
|
( channelInfos[i].type != ASIOSTInt24LSB ) &&
|
|
( channelInfos[i].type != ASIOSTInt32LSB ) )
|
|
{
|
|
// clean up and return error string
|
|
ASIODisposeBuffers();
|
|
ASIOExit();
|
|
asioDrivers->removeCurrentDriver();
|
|
return "Required audio sample format not available (16/24/32 bit LSB).";
|
|
}
|
|
}
|
|
|
|
// 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 );
|
|
|
|
return "";
|
|
}
|
|
|
|
void CSound::Init()
|
|
{
|
|
// first, stop audio and dispose ASIO buffers
|
|
ASIOStop();
|
|
|
|
ASIOMutex.lock(); // get mutex lock
|
|
{
|
|
// store new buffer number values
|
|
iCurNumSndBufIn = iNewNumSndBufIn;
|
|
iCurNumSndBufOut = iNewNumSndBufOut;
|
|
|
|
// initialize write block pointer in and overrun flag
|
|
iBufferPosCapture = 0;
|
|
bCaptureBufferOverrun = false;
|
|
|
|
// create memory for capture buffer
|
|
if ( psCaptureBuffer != NULL )
|
|
{
|
|
delete[] psCaptureBuffer;
|
|
}
|
|
psCaptureBuffer = new short[iCurNumSndBufIn * iBufferSizeStereo];
|
|
|
|
// initialize write block pointer out and underrun flag
|
|
iBufferPosPlay = 0;
|
|
bPlayBufferUnderrun = false;
|
|
|
|
// create memory for play buffer
|
|
if ( psPlayBuffer != NULL )
|
|
{
|
|
delete[] psPlayBuffer;
|
|
}
|
|
psPlayBuffer = new short[iCurNumSndBufOut * iBufferSizeStereo];
|
|
|
|
// clear new buffer
|
|
for ( int i = 0; i < iCurNumSndBufOut * iBufferSizeStereo; i++ )
|
|
{
|
|
psPlayBuffer[i] = 0;
|
|
}
|
|
|
|
// reset event
|
|
ResetEvent ( m_ASIOEvent );
|
|
}
|
|
ASIOMutex.unlock();
|
|
|
|
// initialization is done, (re)start audio
|
|
ASIOStart();
|
|
}
|
|
|
|
void CSound::Close()
|
|
{
|
|
// stop driver
|
|
ASIOStop();
|
|
|
|
// set event to ensure that thread leaves the waiting function
|
|
if ( m_ASIOEvent != NULL )
|
|
{
|
|
SetEvent ( m_ASIOEvent );
|
|
}
|
|
|
|
// wait for the thread to terminate
|
|
Sleep ( 500 );
|
|
|
|
// set flag to open devices the next time it is initialized
|
|
bChangParamIn = true;
|
|
bChangParamOut = true;
|
|
}
|
|
|
|
CSound::CSound ( const int iNewBufferSizeStereo,
|
|
void (*fpNewCallback) ( CVector<short>& psData, void* arg ), void* arg ) :
|
|
CSoundBase ( iNewBufferSizeStereo, fpNewCallback, arg )
|
|
{
|
|
// set internal buffer size value and calculate mono buffer size
|
|
iBufferSizeStereo = iNewBufferSizeStereo;
|
|
iBufferSizeMono = iBufferSizeStereo / 2;
|
|
|
|
// init number of sound buffers
|
|
iNewNumSndBufIn = NUM_SOUND_BUFFERS_IN;
|
|
iCurNumSndBufIn = NUM_SOUND_BUFFERS_IN;
|
|
iNewNumSndBufOut = NUM_SOUND_BUFFERS_OUT;
|
|
iCurNumSndBufOut = NUM_SOUND_BUFFERS_OUT;
|
|
iMinNumSndBuf = 1;
|
|
|
|
// should be initialized because an error can occur during init
|
|
m_ASIOEvent = NULL;
|
|
|
|
// get available ASIO driver names in system
|
|
for ( int 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 ( "No ASIO audio device (driver) found." );
|
|
}
|
|
|
|
asioDrivers->removeCurrentDriver();
|
|
|
|
// init device index with illegal value to show that driver is not initialized
|
|
lCurDev = -1;
|
|
|
|
// set up the asioCallback structure
|
|
asioCallbacks.bufferSwitch = &bufferSwitch;
|
|
asioCallbacks.sampleRateDidChange = &sampleRateChanged;
|
|
asioCallbacks.asioMessage = &asioMessages;
|
|
asioCallbacks.bufferSwitchTimeInfo = &bufferSwitchTimeInfo;
|
|
|
|
// init buffer pointer to zero
|
|
psCaptureBuffer = NULL;
|
|
psPlayBuffer = NULL;
|
|
|
|
// we use an event controlled structure
|
|
// create event
|
|
m_ASIOEvent = CreateEvent ( NULL, FALSE, FALSE, NULL );
|
|
|
|
// init flags
|
|
bChangParamIn = false;
|
|
bChangParamOut = false;
|
|
bSetNumSndBufToMinimumValue = false;
|
|
}
|
|
|
|
CSound::~CSound()
|
|
{
|
|
// cleanup ASIO stuff
|
|
ASIOStop();
|
|
ASIODisposeBuffers();
|
|
ASIOExit();
|
|
asioDrivers->removeCurrentDriver();
|
|
|
|
// delete allocated memory
|
|
if ( psCaptureBuffer != NULL )
|
|
{
|
|
delete[] psCaptureBuffer;
|
|
}
|
|
if ( psPlayBuffer != NULL )
|
|
{
|
|
delete[] psPlayBuffer;
|
|
}
|
|
|
|
// close the handle for the event
|
|
if ( m_ASIOEvent != NULL )
|
|
{
|
|
CloseHandle ( m_ASIOEvent );
|
|
}
|
|
}
|
|
|
|
// 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
|
|
{
|
|
// first check buffer state of capture and play buffers
|
|
const int iCaptureBufferLen = iCurNumSndBufIn * iBufferSizeStereo;
|
|
|
|
bCaptureBufferOverrun =
|
|
( iBufferPosCapture + 2 * iASIOBufferSizeMono > iCaptureBufferLen );
|
|
|
|
bPlayBufferUnderrun = ( 2 * iASIOBufferSizeMono > iBufferPosPlay );
|
|
|
|
// 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 -----------------------------------------------------
|
|
// first check if space in buffer is available
|
|
if ( !bCaptureBufferOverrun )
|
|
{
|
|
// copy new captured block in thread transfer buffer (copy
|
|
// mono data interleaved in stereo buffer)
|
|
switch ( channelInfos[i].type )
|
|
{
|
|
case ASIOSTInt16LSB:
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
psCaptureBuffer[iBufferPosCapture +
|
|
2 * iCurSample + bufferInfos[i].channelNum] =
|
|
( (short*) bufferInfos[i].buffers[index] )[iCurSample];
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt24LSB:
|
|
|
|
// not yet tested, horrible things might happen with the following code ;-)
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
// convert current sample in 16 bit format
|
|
int iCurSam = 0;
|
|
memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
|
|
iCurSam >>= 8;
|
|
|
|
psCaptureBuffer[iBufferPosCapture +
|
|
2 * iCurSample + bufferInfos[i].channelNum] = static_cast<short> ( iCurSam );
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt32LSB:
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
// convert to 16 bit
|
|
psCaptureBuffer[iBufferPosCapture +
|
|
2 * iCurSample + bufferInfos[i].channelNum] =
|
|
(((int*) bufferInfos[i].buffers[index])[iCurSample] >> 16);
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// PLAYBACK ----------------------------------------------------
|
|
if ( !bPlayBufferUnderrun )
|
|
{
|
|
// copy data from sound card in output buffer (copy
|
|
// interleaved stereo data in mono sound card buffer)
|
|
switch ( channelInfos[i].type )
|
|
{
|
|
case ASIOSTInt16LSB:
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
( (short*) bufferInfos[i].buffers[index] )[iCurSample] =
|
|
psPlayBuffer[2 * iCurSample + bufferInfos[i].channelNum];
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt24LSB:
|
|
|
|
// not yet tested, horrible things might happen with the following code ;-)
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
// convert current sample in 24 bit format
|
|
int iCurSam = static_cast<int> ( psPlayBuffer[2 * iCurSample + bufferInfos[i].channelNum] );
|
|
iCurSam <<= 8;
|
|
|
|
memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt32LSB:
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
// convert to 32 bit
|
|
int iCurSam = static_cast<int> ( psPlayBuffer[2 * iCurSample + bufferInfos[i].channelNum] );
|
|
( (int*) bufferInfos[i].buffers[index] )[iCurSample] = ( iCurSam << 16 );
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
// Manage thread interface buffers for input and output ----------------
|
|
// capture
|
|
if ( !bCaptureBufferOverrun )
|
|
{
|
|
iBufferPosCapture += 2 * iASIOBufferSizeMono;
|
|
}
|
|
|
|
// play
|
|
if ( !bPlayBufferUnderrun )
|
|
{
|
|
// move all other data in play buffer
|
|
const int iLenCopyRegion = iBufferPosPlay - 2 * iASIOBufferSizeMono;
|
|
for ( iCurSample = 0; iCurSample < iLenCopyRegion; iCurSample++ )
|
|
{
|
|
psPlayBuffer[iCurSample] =
|
|
psPlayBuffer[2 * iASIOBufferSizeMono + iCurSample];
|
|
}
|
|
|
|
// adjust "current block to write" pointer
|
|
iBufferPosPlay -= 2 * iASIOBufferSizeMono;
|
|
}
|
|
|
|
|
|
// finally if the driver supports the ASIOOutputReady() optimization,
|
|
// do it here, all data are in place -----------------------------------
|
|
if ( bASIOPostOutput )
|
|
{
|
|
ASIOOutputReady();
|
|
}
|
|
|
|
// set event
|
|
SetEvent ( m_ASIOEvent );
|
|
}
|
|
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;
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|