567 lines
19 KiB
C++
Executable File
567 lines
19 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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// 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<short> 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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std::string CSound::SetDev ( const int iNewDev )
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{
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std::string strReturn = ""; // init with no error
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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 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, go back to
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// original driver and display error message
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LoadAndInitializeDriver ( lCurDev );
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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 ).empty() )
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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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if ( !LoadAndInitializeFirstValidDriver() )
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{
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throw CGenErr ( "No usable ASIO audio device "
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"(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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// 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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return strReturn;
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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 ( LoadAndInitializeDriver ( iCurDriverIdx ).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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// 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::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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// check device capabilities if it fullfills our requirements
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const std::string strStat = CheckDeviceCapabilities();
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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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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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std::string 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 "The audio device does not support the "
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"required sample rate.";
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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 "The audio device does not support the "
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"required number of 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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// only 16/24/32 LSB is supported
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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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{
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// return error string
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return "Required audio sample format not "
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"available (16/24/32 bit LSB).";
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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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// 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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iActualBufferSizeMono = iTrialBufSize;
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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<short>& 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 ( "No ASIO audio device (driver) found." );
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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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// 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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{
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bufferSwitch ( index, processNow );
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return 0L;
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}
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void CSound::bufferSwitch ( long index, ASIOBool processNow )
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{
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int iCurSample;
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ASIOMutex.lock(); // get mutex lock
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{
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// perform the processing for input and output
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for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ ) // stereo
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{
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if ( bufferInfos[i].isInput == ASIOTrue )
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{
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// CAPTURE -----------------------------------------------------
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// copy new captured block in thread transfer buffer (copy
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// mono data interleaved in stereo buffer)
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switch ( channelInfos[i].type )
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{
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case ASIOSTInt16LSB:
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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( (short*) bufferInfos[i].buffers[index] )[iCurSample];
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}
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break;
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case ASIOSTInt24LSB:
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// not yet tested, horrible things might happen with the following code ;-)
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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// convert current sample in 16 bit format
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int iCurSam = 0;
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memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
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iCurSam >>= 8;
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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static_cast<short> ( iCurSam );
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}
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break;
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case ASIOSTInt32LSB:
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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// convert to 16 bit
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
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(((int*) bufferInfos[i].buffers[index])[iCurSample] >> 16);
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}
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break;
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}
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}
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}
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// call processing callback function
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pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
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// perform the processing for input and output
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for ( int i = 0; i < 2 * NUM_IN_OUT_CHANNELS; i++ ) // stereo
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{
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if ( bufferInfos[i].isInput != ASIOTrue )
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{
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// PLAYBACK ----------------------------------------------------
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// copy data from sound card in output buffer (copy
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// interleaved stereo data in mono sound card buffer)
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switch ( channelInfos[i].type )
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{
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case ASIOSTInt16LSB:
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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( (short*) bufferInfos[i].buffers[index] )[iCurSample] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum];
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}
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break;
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case ASIOSTInt24LSB:
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// not yet tested, horrible things might happen with the following code ;-)
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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// convert current sample in 24 bit format
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int iCurSam = static_cast<int> ( vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
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iCurSam <<= 8;
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memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
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}
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break;
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case ASIOSTInt32LSB:
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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// convert to 32 bit
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int iCurSam = static_cast<int> ( vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
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( (int*) bufferInfos[i].buffers[index] )[iCurSample] = ( iCurSam << 16 );
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}
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break;
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}
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}
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}
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// finally if the driver supports the ASIOOutputReady() optimization,
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// do it here, all data are in place -----------------------------------
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if ( bASIOPostOutput )
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{
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ASIOOutputReady();
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}
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}
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ASIOMutex.unlock();
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}
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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;
|
|
}
|