1194 lines
46 KiB
C++
Executable file
1194 lines
46 KiB
C++
Executable file
/******************************************************************************\
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* Copyright (c) 2004-2020
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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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// pointer to our sound object
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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::LoadAndInitializeDriver ( int iDriverIdx,
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bool bOpenDriverSetup )
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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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// check if device is capable
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if ( strStat.isEmpty() )
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{
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// the device has changed, per definition we reset the channel
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// mapping to the defaults (first two available channels)
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ResetChannelMapping();
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// store ID of selected driver if initialization was successful
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lCurDev = iDriverIdx;
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}
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else
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{
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// if requested, open ASIO driver setup in case of an error
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if ( bOpenDriverSetup )
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{
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OpenDriverSetup();
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QMessageBox::question ( nullptr, APP_NAME, "Are you done with your ASIO driver settings of device " + GetDeviceName ( iDriverIdx ) + "?", QMessageBox::Yes );
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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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void CSound::UnloadCurrentDriver()
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{
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// clean up 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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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_HZ );
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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 ) + " Hz";
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}
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// check if sample rate can be set
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const ASIOError SetSaRateReturn = ASIOSetSampleRate ( SYSTEM_SAMPLE_RATE_HZ );
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if ( ( SetSaRateReturn == ASE_NoClock ) ||
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( SetSaRateReturn == ASE_InvalidMode ) ||
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( SetSaRateReturn == 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 to set the required sampling "
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"rate. This error can happen if you have an audio interface like the "
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"Roland UA-25EX where you set the sample rate with a hardware switch "
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"on the audio device. If this is the case, please change the sample rate "
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"to " ) + QString().setNum ( SYSTEM_SAMPLE_RATE_HZ ) + tr ( " Hz on the "
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"device and restart the " ) + APP_NAME + tr ( " software." );
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}
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// check the number of available channels
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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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"for input and output is: " ) +
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QString().setNum ( NUM_IN_OUT_CHANNELS );
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}
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// clip number of input/output channels to our maximum
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if ( lNumInChan > MAX_NUM_IN_OUT_CHANNELS )
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{
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lNumInChan = MAX_NUM_IN_OUT_CHANNELS;
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}
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if ( lNumOutChan > MAX_NUM_IN_OUT_CHANNELS )
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{
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lNumOutChan = MAX_NUM_IN_OUT_CHANNELS;
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}
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// query channel infos for all available input channels
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bool bInputChMixingSupported = true;
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for ( int i = 0; i < lNumInChan; i++ )
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{
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// setup for input channels
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channelInfosInput[i].isInput = ASIOTrue;
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channelInfosInput[i].channel = i;
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ASIOGetChannelInfo ( &channelInfosInput[i] );
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// Check supported sample formats.
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// Actually, it would be enough to have at least two channels which
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// support the required sample format. But since we have support for
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// all known sample types, the following check should always pass and
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// therefore we throw the error message on any channel which does not
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// fullfill the sample format requirement (quick hack solution).
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if ( !CheckSampleTypeSupported ( channelInfosInput[i].type ) )
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{
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// return error string
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return tr ( "Required audio sample format not available." );
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}
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// store the name of the channel and check if channel mixing is supported
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channelInputName[i] = channelInfosInput[i].name;
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if ( !CheckSampleTypeSupportedForCHMixing ( channelInfosInput[i].type ) )
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{
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bInputChMixingSupported = false;
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}
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}
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// query channel infos for all available output channels
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for ( int i = 0; i < lNumOutChan; i++ )
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{
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// setup for output channels
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channelInfosOutput[i].isInput = ASIOFalse;
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channelInfosOutput[i].channel = i;
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ASIOGetChannelInfo ( &channelInfosOutput[i] );
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// Check supported sample formats.
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// Actually, it would be enough to have at least two channels which
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// support the required sample format. But since we have support for
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// all known sample types, the following check should always pass and
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// therefore we throw the error message on any channel which does not
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// fullfill the sample format requirement (quick hack solution).
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if ( !CheckSampleTypeSupported ( channelInfosOutput[i].type ) )
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{
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// return error string
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return tr ( "Required audio sample format not available." );
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}
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}
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// special case with 4 input channels: support adding channels
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if ( ( lNumInChan == 4 ) && bInputChMixingSupported )
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{
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// add four mixed channels (i.e. 4 normal, 4 mixed channels)
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lNumInChanPlusAddChan = 8;
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for ( int iCh = 0; iCh < lNumInChanPlusAddChan; iCh++ )
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{
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int iSelCH, iSelAddCH;
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GetSelCHAndAddCH ( iCh, lNumInChan, iSelCH, iSelAddCH );
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if ( iSelAddCH >= 0 )
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{
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// for mixed channels, show both audio channel names to be mixed
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channelInputName[iCh] =
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channelInputName[iSelCH] + " + " + channelInputName[iSelAddCH];
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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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// regular case: no mixing input channels used
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lNumInChanPlusAddChan = lNumInChan;
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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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void CSound::SetLeftInputChannel ( const int iNewChan )
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{
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// apply parameter after input parameter check
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if ( ( iNewChan >= 0 ) && ( iNewChan < lNumInChanPlusAddChan ) )
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{
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vSelectedInputChannels[0] = iNewChan;
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}
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}
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void CSound::SetRightInputChannel ( const int iNewChan )
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{
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// apply parameter after input parameter check
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if ( ( iNewChan >= 0 ) && ( iNewChan < lNumInChanPlusAddChan ) )
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{
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vSelectedInputChannels[1] = iNewChan;
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}
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}
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void CSound::SetLeftOutputChannel ( const int iNewChan )
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{
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// apply parameter after input parameter check
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if ( ( iNewChan >= 0 ) && ( iNewChan < lNumOutChan ) )
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{
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vSelectedOutputChannels[0] = iNewChan;
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}
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}
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void CSound::SetRightOutputChannel ( const int iNewChan )
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{
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// apply parameter after input parameter check
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if ( ( iNewChan >= 0 ) && ( iNewChan < lNumOutChan ) )
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{
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vSelectedOutputChannels[1] = iNewChan;
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}
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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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/*
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// TEST
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#include <QMessageBox>
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QMessageBox::information ( 0, "APP_NAME", QString("lMinSize: %1, lMaxSize: %2, lPreferredSize: %3, lGranularity: %4").
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arg(HWBufferInfo.lMinSize).arg(HWBufferInfo.lMaxSize).arg(HWBufferInfo.lPreferredSize).arg(HWBufferInfo.lGranularity) );
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_exit(1);
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*/
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// TODO see https://github.com/EddieRingle/portaudio/blob/master/src/hostapi/asio/pa_asio.cpp#L1654 (SelectHostBufferSizeForUnspecifiedUserFramesPerBuffer)
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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 = GetActualBufferSize ( iNewPrefMonoBufferSize );
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// init base class
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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_HZ );
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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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// prepare input channels
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for ( int i = 0; i < lNumInChan; 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;
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}
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// prepare output channels
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for ( int i = 0; i < lNumOutChan; i++ )
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{
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bufferInfos[lNumInChan + i].isInput = ASIOFalse;
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bufferInfos[lNumInChan + i].channelNum = i;
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bufferInfos[lNumInChan + i].buffers[0] = 0;
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bufferInfos[lNumInChan + i].buffers[1] = 0;
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}
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ASIOCreateBuffers ( bufferInfos, lNumInChan + lNumOutChan,
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iASIOBufferSizeMono, &asioCallbacks );
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// query the latency of the driver
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long lInputLatency = 0;
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long lOutputLatency = 0;
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if ( ASIOGetLatencies ( &lInputLatency, &lOutputLatency ) != ASE_NotPresent )
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{
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// add the input and output latencies (returned in number of
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// samples) and calculate the time in ms
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dInOutLatencyMs =
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( static_cast<double> ( lInputLatency ) + lOutputLatency ) *
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1000 / SYSTEM_SAMPLE_RATE_HZ;
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}
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else
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{
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// no latency available
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dInOutLatencyMs = 0.0;
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}
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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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// make sure the working thread is actually done
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// (by checking the locked state)
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if ( ASIOMutex.tryLock ( 5000 ) )
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{
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ASIOMutex.unlock();
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}
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}
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CSound::CSound ( void (*fpNewCallback) ( CVector<int16_t>& psData, void* arg ),
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void* arg,
|
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const int iCtrlMIDIChannel,
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const bool bNoAutoJackConnect) :
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CSoundBase ( "ASIO", true, fpNewCallback, arg, iCtrlMIDIChannel, bNoAutoJackConnect ),
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lNumInChan ( 0 ),
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lNumInChanPlusAddChan ( 0 ),
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lNumOutChan ( 0 ),
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dInOutLatencyMs ( 0.0 ), // "0.0" means that no latency value is available
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vSelectedInputChannels ( NUM_IN_OUT_CHANNELS ),
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vSelectedOutputChannels ( NUM_IN_OUT_CHANNELS )
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{
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int i;
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// init pointer to our sound object
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pSound = this;
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// get available ASIO driver names in system
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for ( i = 0; i < MAX_NUMBER_SOUND_CARDS; i++ )
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{
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// allocate memory for driver names
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cDriverNames[i] = new char[32];
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}
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char cDummyName[] = "dummy";
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loadAsioDriver ( cDummyName ); // to initialize external object
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lNumDevs = asioDrivers->getDriverNames ( cDriverNames, MAX_NUMBER_SOUND_CARDS );
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|
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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>"
|
|
"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 "
|
|
"Windows audio interface and therefore a special audio driver is "
|
|
"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 driver." ) );
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}
|
|
asioDrivers->removeCurrentDriver();
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|
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// copy driver names to base class but internally we still have to use
|
|
// the char* variable because of the ASIO API :-(
|
|
for ( i = 0; i < lNumDevs; i++ )
|
|
{
|
|
strDriverNames[i] = cDriverNames[i];
|
|
}
|
|
|
|
// init device index as not initialized (invalid)
|
|
lCurDev = INVALID_SNC_CARD_DEVICE;
|
|
|
|
// init channel mapping
|
|
ResetChannelMapping();
|
|
|
|
// set up the asioCallback structure
|
|
asioCallbacks.bufferSwitch = &bufferSwitch;
|
|
asioCallbacks.sampleRateDidChange = &sampleRateChanged;
|
|
asioCallbacks.asioMessage = &asioMessages;
|
|
asioCallbacks.bufferSwitchTimeInfo = &bufferSwitchTimeInfo;
|
|
}
|
|
|
|
void CSound::ResetChannelMapping()
|
|
{
|
|
// init selected channel numbers with defaults: use first available
|
|
// channels for input and output
|
|
vSelectedInputChannels[0] = 0;
|
|
vSelectedInputChannels[1] = 1;
|
|
vSelectedOutputChannels[0] = 0;
|
|
vSelectedOutputChannels[1] = 1;
|
|
}
|
|
|
|
|
|
// ASIO callbacks -------------------------------------------------------------
|
|
ASIOTime* CSound::bufferSwitchTimeInfo ( ASIOTime*,
|
|
long index,
|
|
ASIOBool processNow )
|
|
{
|
|
bufferSwitch ( index, processNow );
|
|
return 0L;
|
|
}
|
|
|
|
bool CSound::CheckSampleTypeSupported ( const ASIOSampleType SamType )
|
|
{
|
|
// check for supported sample types
|
|
return ( ( SamType == ASIOSTInt16LSB ) ||
|
|
( SamType == ASIOSTInt24LSB ) ||
|
|
( SamType == ASIOSTInt32LSB ) ||
|
|
( SamType == ASIOSTFloat32LSB ) ||
|
|
( SamType == ASIOSTFloat64LSB ) ||
|
|
( SamType == ASIOSTInt32LSB16 ) ||
|
|
( SamType == ASIOSTInt32LSB18 ) ||
|
|
( SamType == ASIOSTInt32LSB20 ) ||
|
|
( SamType == ASIOSTInt32LSB24 ) ||
|
|
( SamType == ASIOSTInt16MSB ) ||
|
|
( SamType == ASIOSTInt24MSB ) ||
|
|
( SamType == ASIOSTInt32MSB ) ||
|
|
( SamType == ASIOSTFloat32MSB ) ||
|
|
( SamType == ASIOSTFloat64MSB ) ||
|
|
( SamType == ASIOSTInt32MSB16 ) ||
|
|
( SamType == ASIOSTInt32MSB18 ) ||
|
|
( SamType == ASIOSTInt32MSB20 ) ||
|
|
( SamType == ASIOSTInt32MSB24 ) );
|
|
}
|
|
|
|
bool CSound::CheckSampleTypeSupportedForCHMixing ( const ASIOSampleType SamType )
|
|
{
|
|
// check for supported sample types for audio channel mixing (see bufferSwitch)
|
|
return ( ( SamType == ASIOSTInt16LSB ) ||
|
|
( SamType == ASIOSTInt24LSB ) ||
|
|
( SamType == ASIOSTInt32LSB ) );
|
|
}
|
|
|
|
void CSound::bufferSwitch ( long index, ASIOBool )
|
|
{
|
|
int iCurSample;
|
|
|
|
// get references to class members
|
|
int& iASIOBufferSizeMono = pSound->iASIOBufferSizeMono;
|
|
CVector<int16_t>& vecsTmpAudioSndCrdStereo = pSound->vecsTmpAudioSndCrdStereo;
|
|
|
|
// perform the processing for input and output
|
|
pSound->ASIOMutex.lock(); // get mutex lock
|
|
{
|
|
// CAPTURE -------------------------------------------------------------
|
|
for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
|
|
{
|
|
int iSelCH, iSelAddCH;
|
|
|
|
GetSelCHAndAddCH ( pSound->vSelectedInputChannels[i], pSound->lNumInChan,
|
|
iSelCH, iSelAddCH );
|
|
|
|
// copy new captured block in thread transfer buffer (copy
|
|
// mono data interleaved in stereo buffer)
|
|
switch ( pSound->channelInfosInput[iSelCH].type )
|
|
{
|
|
case ASIOSTInt16LSB:
|
|
{
|
|
// no type conversion required, just copy operation
|
|
int16_t* pASIOBuf = static_cast<int16_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = pASIOBuf[iCurSample];
|
|
}
|
|
|
|
if ( iSelAddCH >= 0 )
|
|
{
|
|
// mix input channels case:
|
|
int16_t* pASIOBufAdd = static_cast<int16_t*> ( pSound->bufferInfos[iSelAddCH].buffers[index] );
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
|
|
Double2Short ( (double) vecsTmpAudioSndCrdStereo[2 * iCurSample + i] +
|
|
(double) pASIOBufAdd[iCurSample] );
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
case ASIOSTInt24LSB:
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
int iCurSam = 0;
|
|
memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
|
|
iCurSam >>= 8;
|
|
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = static_cast<int16_t> ( iCurSam );
|
|
}
|
|
|
|
if ( iSelAddCH >= 0 )
|
|
{
|
|
// mix input channels case:
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
int iCurSam = 0;
|
|
memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelAddCH].buffers[index] ) + iCurSample * 3, 3 );
|
|
iCurSam >>= 8;
|
|
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
|
|
Double2Short ( (double) vecsTmpAudioSndCrdStereo[2 * iCurSample + i] +
|
|
(double) static_cast<int16_t> ( iCurSam ) );
|
|
}
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt32LSB:
|
|
{
|
|
int32_t* pASIOBuf = static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
|
|
static_cast<int16_t> ( pASIOBuf[iCurSample] >> 16 );
|
|
}
|
|
|
|
if ( iSelAddCH >= 0 )
|
|
{
|
|
// mix input channels case:
|
|
int32_t* pASIOBufAdd = static_cast<int32_t*> ( pSound->bufferInfos[iSelAddCH].buffers[index] );
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
|
|
Double2Short ( (double) vecsTmpAudioSndCrdStereo[2 * iCurSample + i] +
|
|
(double) static_cast<int16_t> ( pASIOBufAdd[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 + i] =
|
|
static_cast<int16_t> ( static_cast<float*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( static_cast<double*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt16MSB:
|
|
// NOT YET TESTED
|
|
// flip bits
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
|
|
Flip16Bits ( ( static_cast<int16_t*> (
|
|
pSound->bufferInfos[iSelCH].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*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
|
|
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
|
|
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 + i] =
|
|
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( static_cast<float> (
|
|
Flip32Bits ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( static_cast<double> (
|
|
Flip64Bits ( static_cast<int64_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].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 + i] =
|
|
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
|
|
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
// call processing callback function
|
|
pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
|
|
|
|
|
|
// PLAYBACK ------------------------------------------------------------
|
|
for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
|
|
{
|
|
const int iSelCH = pSound->lNumInChan + pSound->vSelectedOutputChannels[i];
|
|
|
|
// copy data from sound card in output buffer (copy
|
|
// interleaved stereo data in mono sound card buffer)
|
|
switch ( pSound->channelInfosOutput[pSound->vSelectedOutputChannels[i]].type )
|
|
{
|
|
case ASIOSTInt16LSB:
|
|
{
|
|
// no type conversion required, just copy operation
|
|
int16_t* pASIOBuf = static_cast<int16_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
pASIOBuf[iCurSample] = vecsTmpAudioSndCrdStereo[2 * iCurSample + i];
|
|
}
|
|
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 + i] );
|
|
|
|
iCurSam <<= 8;
|
|
|
|
memcpy ( ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt32LSB:
|
|
{
|
|
int32_t* pASIOBuf = static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
|
|
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
// convert to 32 bit
|
|
const int32_t iCurSam = static_cast<int32_t> (
|
|
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
|
|
|
|
pASIOBuf[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 + i] );
|
|
|
|
static_cast<float*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<double*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
|
|
( iCurSam << 8 );
|
|
}
|
|
break;
|
|
|
|
case ASIOSTInt16MSB:
|
|
// NOT YET TESTED
|
|
// flip bits
|
|
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
|
|
{
|
|
( (int16_t*) pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
|
|
Flip16Bits ( vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
|
|
}
|
|
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 + i] ) );
|
|
|
|
memcpy ( ( (char*) pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<float*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<float*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].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 + i] );
|
|
|
|
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
|
|
Flip32Bits ( iCurSam << 8 );
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
// Finally if the driver supports the ASIOOutputReady() optimization,
|
|
// do it here, all data are in place -----------------------------------
|
|
if ( pSound->bASIOPostOutput )
|
|
{
|
|
ASIOOutputReady();
|
|
}
|
|
}
|
|
pSound->ASIOMutex.unlock();
|
|
}
|
|
|
|
long CSound::asioMessages ( long selector,
|
|
long,
|
|
void*,
|
|
double* )
|
|
{
|
|
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:
|
|
pSound->EmitReinitRequestSignal ( RS_ONLY_RESTART_AND_INIT );
|
|
ret = 1L; // 1L if request is accepted or 0 otherwise
|
|
break;
|
|
|
|
case kAsioResetRequest:
|
|
pSound->EmitReinitRequestSignal ( RS_RELOAD_RESTART_AND_INIT );
|
|
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;
|
|
}
|