mirror of
https://github.com/nzp-team/fteqw.git
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27a59a0cbc
vulkan, wasapi, quake injector features added. irc, avplug, cef plugins/drivers reworked/updated/added openal reverb, doppler effects added. 'dir' console command now attempts to view clicked files. lots of warning fixes, should now only be deprecation warnings for most targets (depending on compiler version anyway...). SendEntity finally reworked to use flags properly. effectinfo improved, other smc-targetted fixes. mapcluster stuff now has support for linux. .basebone+.baseframe now exist in ssqc. qcc: -Fqccx supports qccx syntax, including qccx hacks. don't expect these to work in fteqw nor dp though. qcc: rewrote function call handling to use refs rather than defs. this makes struct passing more efficient and makes the __out keyword usable with fields etc. qccgui: can cope a little better with non-unicode files. can now represent most quake chars. qcc: suppressed warnings from *extensions.qc git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5000 fc73d0e0-1445-4013-8a0c-d673dee63da5
253 lines
6.3 KiB
C
253 lines
6.3 KiB
C
/*
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Copyright (C) 2001-2002 A Nourai
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Copyright (C) 2006 Jacek Piszczek (Mac OSX port)
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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See the included (GNU.txt) GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#include "quakedef.h"
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#include "sound.h"
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#include <CoreServices/CoreServices.h>
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#include <AudioUnit/AudioUnit.h>
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// Jacek:
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// coreaudio is poorly documented so I'm not 100% sure the code below
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// is correct :(
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struct MacOSSound_Private
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{
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AudioUnit gOutputUnit;
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unsigned int readpos;
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};
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static OSStatus AudioRender(void *inRefCon,
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AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp,
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UInt32 inBusNumber,
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UInt32 inNumberFrames,
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AudioBufferList *ioData)
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{
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soundcardinfo_t *sc = inRefCon;
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struct MacOSSound_Private *pdata = sc->handle;
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int start = pdata->readpos;
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int buffersize = sc->sn.samples * (sc->sn.samplebits/8);
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int bytes = ioData->mBuffers[0].mDataByteSize;
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int remaining;
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start %= buffersize;
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if (start + bytes > buffersize)
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{
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remaining = bytes;
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bytes = buffersize - start;
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remaining -= bytes;
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}
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else
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{
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remaining = 0;
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}
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memcpy(ioData->mBuffers[0].mData, sc->sn.buffer + start, bytes);
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memcpy((char*)ioData->mBuffers[0].mData+bytes, sc->sn.buffer, remaining);
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pdata->readpos += inNumberFrames*sc->sn.numchannels * (sc->sn.samplebits/8);
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return noErr;
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}
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static void MacOS_Shutdown(soundcardinfo_t *sc)
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{
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struct MacOSSound_Private *pdata = sc->handle;
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sc->handle = NULL;
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if (!pdata)
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return;
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// stop playback
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AudioOutputUnitStop (pdata->gOutputUnit);
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// release the unit
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AudioUnitUninitialize (pdata->gOutputUnit);
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// free the unit
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CloseComponent (pdata->gOutputUnit);
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// free the buffer memory
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Z_Free(sc->sn.buffer);
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Z_Free(pdata);
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}
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static unsigned int MacOS_GetDMAPos(soundcardinfo_t *sc)
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{
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struct MacOSSound_Private *pdata = sc->handle;
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sc->sn.samplepos = pdata->readpos/(sc->sn.samplebits/8);
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return sc->sn.samplepos;
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}
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static void MacOS_Submit(soundcardinfo_t *sc)
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{
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}
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static void *MacOS_Lock(soundcardinfo_t *sc, unsigned int *sampidx)
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{
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return sc->sn.buffer;
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}
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static void MacOS_Unlock(soundcardinfo_t *sc, void *buffer)
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{
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}
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static int MacOS_InitCard(soundcardinfo_t *sc, int cardnum)
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{
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ComponentResult err = noErr;
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if (cardnum)
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return 2; /* no more */
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struct MacOSSound_Private *pdata = Z_Malloc(sizeof(*pdata));
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if (!pdata)
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return FALSE;
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// Open the default output unit
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ComponentDescription desc;
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desc.componentType = kAudioUnitType_Output;
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desc.componentSubType = kAudioUnitSubType_DefaultOutput;
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desc.componentManufacturer = kAudioUnitManufacturer_Apple;
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desc.componentFlags = 0;
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desc.componentFlagsMask = 0;
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Component comp = FindNextComponent(NULL, &desc);
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if (comp == NULL)
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{
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Con_Printf("FindNextComponent failed\n");
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Z_Free(pdata);
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return FALSE;
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}
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err = OpenAComponent(comp, &pdata->gOutputUnit);
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if (comp == NULL)
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{
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Con_Printf("OpenAComponent failed\n");
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Z_Free(pdata);
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return FALSE;
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}
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// Set up a callback function to generate output to the output unit
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AURenderCallbackStruct input;
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input.inputProc = AudioRender;
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input.inputProcRefCon = sc;
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err = AudioUnitSetProperty ( pdata->gOutputUnit,
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kAudioUnitProperty_SetRenderCallback,
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kAudioUnitScope_Input,
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0,
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&input,
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sizeof(input));
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if (err)
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{
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Con_Printf("AudioUnitSetProperty failed\n");
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CloseComponent(pdata->gOutputUnit);
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Z_Free(pdata);
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return FALSE;
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}
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// describe our audio data
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AudioStreamBasicDescription streamFormat;
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streamFormat.mSampleRate = sc->sn.speed;
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streamFormat.mFormatID = kAudioFormatLinearPCM;
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streamFormat.mFormatFlags = kAudioFormatFlagsNativeEndian
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| kLinearPCMFormatFlagIsPacked;
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//| kAudioFormatFlagIsNonInterleaved;
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streamFormat.mFramesPerPacket = 1;
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streamFormat.mChannelsPerFrame = 2;
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streamFormat.mBitsPerChannel = 16;
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if (streamFormat.mBitsPerChannel >= 16)
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streamFormat.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
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else
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streamFormat.mFormatFlags |= 0;
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streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame * (streamFormat.mBitsPerChannel/8);
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streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame * streamFormat.mFramesPerPacket;
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err = AudioUnitSetProperty (pdata->gOutputUnit,
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kAudioUnitProperty_StreamFormat,
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kAudioUnitScope_Input,
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0,
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&streamFormat,
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sizeof(AudioStreamBasicDescription));
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if (err)
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{
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Con_Printf("AudioUnitSetProperty failed\n");
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CloseComponent(pdata->gOutputUnit);
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Z_Free(pdata);
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return FALSE;
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}
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// set the shm structure
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sc->sn.speed = streamFormat.mSampleRate;
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sc->sn.samplebits = streamFormat.mBitsPerChannel;
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sc->sn.numchannels = streamFormat.mChannelsPerFrame;
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sc->sn.samples = 256 * 1024;
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sc->sn.buffer = Z_Malloc(sc->sn.samples*sc->sn.samplebits/8);
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int i;
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for (i = 0; i < sc->sn.samples*sc->sn.samplebits/8; i++)
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sc->sn.buffer[i] = rand();
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if (sc->sn.buffer == 0)
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{
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Con_Printf("Malloc failed - cannot allocate sound buffer\n");
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CloseComponent(pdata->gOutputUnit);
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Z_Free(pdata);
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return FALSE;
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}
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// Initialize unit
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err = AudioUnitInitialize(pdata->gOutputUnit);
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if (err)
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{
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Con_Printf("AudioOutputInitialize failed\n");
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CloseComponent(pdata->gOutputUnit);
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Z_Free(sc->sn.buffer);
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Z_Free(pdata);
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return FALSE;
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}
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// start playing :)
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err = AudioOutputUnitStart (pdata->gOutputUnit);
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if (err)
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{
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Con_Printf("AudioOutputUnitStart failed\n");
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AudioUnitUninitialize (pdata->gOutputUnit);
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CloseComponent(pdata->gOutputUnit);
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Z_Free(sc->sn.buffer);
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Z_Free(pdata);
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return FALSE;
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}
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sc->handle = pdata;
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sc->Lock = MacOS_Lock;
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sc->Unlock = MacOS_Unlock;
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sc->Submit = MacOS_Submit;
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sc->GetDMAPos = MacOS_GetDMAPos;
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sc->Shutdown = MacOS_Shutdown;
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Con_Printf("Sound initialised\n");
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return TRUE;
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}
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sounddriver pMacOS_InitCard = &MacOS_InitCard;
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