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https://git.code.sf.net/p/quake/quakeforge-old
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753 lines
16 KiB
C
753 lines
16 KiB
C
/*
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Copyright (C) 1996-1997 Id Software, Inc.
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Copyright (C) 1999,2000 contributors of the QuakeForge project
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Please see the file "AUTHORS" for a list of contributors
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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 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 "winquake.h"
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#ifdef HAVE_MMSYSTEM_H
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# include <mmsystem.h>
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#endif
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#ifdef HAVE_DSOUND
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#define iDirectSoundCreate(a,b,c) pDirectSoundCreate(a,b,c)
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HRESULT (WINAPI *pDirectSoundCreate)(GUID FAR *lpGUID, LPDIRECTSOUND FAR *lplpDS, IUnknown FAR *pUnkOuter);
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#endif
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// 64K is > 1 second at 16-bit, 22050 Hz
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#define WAV_BUFFERS 64
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#define WAV_MASK 0x3F
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#define WAV_BUFFER_SIZE 0x0400
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#define SECONDARY_BUFFER_SIZE 0x10000
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typedef enum {SIS_SUCCESS, SIS_FAILURE, SIS_NOTAVAIL} sndinitstat;
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static qboolean wavonly;
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static qboolean dsound_init;
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static qboolean wav_init;
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static qboolean snd_firsttime = true, snd_iswave;
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#ifdef HAVE_DSOUND
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static qboolean primary_format_set, snd_isdirect;
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#endif
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static int sample16;
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static int snd_sent, snd_completed;
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/*
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* Global variables. Must be visible to window-procedure function
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* so it can unlock and free the data block after it has been played.
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*/
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HANDLE hData;
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HPSTR lpData, lpData2;
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HGLOBAL hWaveHdr;
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LPWAVEHDR lpWaveHdr;
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HWAVEOUT hWaveOut;
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WAVEOUTCAPS wavecaps;
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DWORD gSndBufSize;
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MMTIME mmstarttime;
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#ifdef HAVE_DSOUND
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LPDIRECTSOUND pDS;
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LPDIRECTSOUNDBUFFER pDSBuf, pDSPBuf;
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#endif
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HINSTANCE hInstDS;
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qboolean SNDDMA_InitDirect (void);
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qboolean SNDDMA_InitWav (void);
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/*
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==================
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S_BlockSound
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==================
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*/
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void S_BlockSound (void)
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{
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// DirectSound takes care of blocking itself
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if (snd_iswave)
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{
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snd_blocked++;
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if (snd_blocked == 1)
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waveOutReset (hWaveOut);
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}
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}
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/*
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==================
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S_UnblockSound
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==================
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*/
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void S_UnblockSound (void)
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{
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// DirectSound takes care of blocking itself
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if (snd_iswave)
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{
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snd_blocked--;
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}
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}
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/*
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==================
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FreeSound
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==================
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*/
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void FreeSound (void)
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{
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int i;
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#ifdef HAVE_DSOUND
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if (pDSBuf)
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{
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pDSBuf->lpVtbl->Stop(pDSBuf);
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pDSBuf->lpVtbl->Release(pDSBuf);
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}
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// only release primary buffer if it's not also the mixing buffer we just released
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if (pDSPBuf && (pDSBuf != pDSPBuf))
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{
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pDSPBuf->lpVtbl->Release(pDSPBuf);
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}
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if (pDS)
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{
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pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_NORMAL);
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pDS->lpVtbl->Release(pDS);
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}
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#endif
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if (hWaveOut)
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{
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waveOutReset (hWaveOut);
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if (lpWaveHdr)
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{
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for (i=0 ; i< WAV_BUFFERS ; i++)
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waveOutUnprepareHeader (hWaveOut, lpWaveHdr+i, sizeof(WAVEHDR));
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}
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waveOutClose (hWaveOut);
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if (hWaveHdr)
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{
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GlobalUnlock(hWaveHdr);
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GlobalFree(hWaveHdr);
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}
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if (hData)
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{
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GlobalUnlock(hData);
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GlobalFree(hData);
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}
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}
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#ifdef HAVE_DSOUND
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pDS = NULL;
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pDSBuf = NULL;
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pDSPBuf = NULL;
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#endif
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hWaveOut = 0;
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hData = 0;
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hWaveHdr = 0;
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lpData = NULL;
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lpWaveHdr = NULL;
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dsound_init = false;
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wav_init = false;
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}
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/*
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==================
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SNDDMA_InitDirect
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Direct-Sound support
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==================
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*/
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#ifdef HAVE_DSOUND
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sndinitstat SNDDMA_InitDirect (void)
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{
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DSBUFFERDESC dsbuf;
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DSBCAPS dsbcaps;
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DWORD dwSize, dwWrite;
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DSCAPS dscaps;
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WAVEFORMATEX format, pformat;
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HRESULT hresult;
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int reps;
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memset ((void *)&sn, 0, sizeof (sn));
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shm = &sn;
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shm->channels = 2;
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shm->samplebits = 16;
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shm->speed = 11025;
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memset (&format, 0, sizeof(format));
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format.wFormatTag = WAVE_FORMAT_PCM;
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format.nChannels = shm->channels;
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format.wBitsPerSample = shm->samplebits;
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format.nSamplesPerSec = shm->speed;
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format.nBlockAlign = format.nChannels
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*format.wBitsPerSample / 8;
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format.cbSize = 0;
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format.nAvgBytesPerSec = format.nSamplesPerSec
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*format.nBlockAlign;
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if (!hInstDS)
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{
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hInstDS = LoadLibrary("dsound.dll");
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if (hInstDS == NULL)
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{
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Con_SafePrintf ("Couldn't load dsound.dll\n");
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return SIS_FAILURE;
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}
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pDirectSoundCreate = (void *)GetProcAddress(hInstDS,"DirectSoundCreate");
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if (!pDirectSoundCreate)
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{
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Con_SafePrintf ("Couldn't get DS proc addr\n");
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return SIS_FAILURE;
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}
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}
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while ((hresult = iDirectSoundCreate(NULL, &pDS, NULL)) != DS_OK)
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{
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if (hresult != DSERR_ALLOCATED)
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{
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Con_SafePrintf ("DirectSound create failed\n");
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return SIS_FAILURE;
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}
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if (MessageBox (NULL,
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"The sound hardware is in use by another app.\n\n"
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"Select Retry to try to start sound again or Cancel to run Quake with no sound.",
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"Sound not available",
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MB_RETRYCANCEL | MB_SETFOREGROUND | MB_ICONEXCLAMATION) != IDRETRY)
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{
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Con_SafePrintf ("DirectSoundCreate failure\n"
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" hardware already in use\n");
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return SIS_NOTAVAIL;
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}
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}
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dscaps.dwSize = sizeof(dscaps);
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if (DS_OK != pDS->lpVtbl->GetCaps (pDS, &dscaps))
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{
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Con_SafePrintf ("Couldn't get DS caps\n");
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}
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if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
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{
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Con_SafePrintf ("No DirectSound driver installed\n");
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FreeSound ();
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return SIS_FAILURE;
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}
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// Changed DSSCL_EXCLUSIVE to DSSCL_NORMAL
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// -- Robert S. Elsner <sockman@ngfc.com>
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if (DS_OK != pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_NORMAL))
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// if (DS_OK != pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_EXCLUSIVE))
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{
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Con_SafePrintf ("Set coop level failed\n");
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FreeSound ();
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return SIS_FAILURE;
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}
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// get access to the primary buffer, if possible, so we can set the
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// sound hardware format
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memset (&dsbuf, 0, sizeof(dsbuf));
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dsbuf.dwSize = sizeof(DSBUFFERDESC);
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dsbuf.dwFlags = DSBCAPS_PRIMARYBUFFER;
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dsbuf.dwBufferBytes = 0;
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dsbuf.lpwfxFormat = NULL;
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memset(&dsbcaps, 0, sizeof(dsbcaps));
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dsbcaps.dwSize = sizeof(dsbcaps);
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primary_format_set = false;
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if (!COM_CheckParm ("-snoforceformat"))
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{
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if (DS_OK == pDS->lpVtbl->CreateSoundBuffer(pDS, &dsbuf, &pDSPBuf, NULL))
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{
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pformat = format;
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if (DS_OK != pDSPBuf->lpVtbl->SetFormat (pDSPBuf, &pformat))
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{
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// if (snd_firsttime)
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// Con_SafePrintf ("Set primary sound buffer format: no\n");
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}
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else
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// {
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// if (snd_firsttime)
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// Con_SafePrintf ("Set primary sound buffer format: yes\n");
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primary_format_set = true;
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// }
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}
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}
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if (!primary_format_set || !COM_CheckParm ("-primarysound"))
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{
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// create the secondary buffer we'll actually work with
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memset (&dsbuf, 0, sizeof(dsbuf));
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dsbuf.dwSize = sizeof(DSBUFFERDESC);
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dsbuf.dwFlags = DSBCAPS_CTRLFREQUENCY | DSBCAPS_LOCSOFTWARE;
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dsbuf.dwBufferBytes = SECONDARY_BUFFER_SIZE;
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dsbuf.lpwfxFormat = &format;
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memset(&dsbcaps, 0, sizeof(dsbcaps));
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dsbcaps.dwSize = sizeof(dsbcaps);
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if (DS_OK != pDS->lpVtbl->CreateSoundBuffer(pDS, &dsbuf, &pDSBuf, NULL))
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{
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Con_SafePrintf ("DS:CreateSoundBuffer Failed");
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FreeSound ();
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return SIS_FAILURE;
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}
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shm->channels = format.nChannels;
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shm->samplebits = format.wBitsPerSample;
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shm->speed = format.nSamplesPerSec;
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if (DS_OK != pDSBuf->lpVtbl->GetCaps (pDSBuf, &dsbcaps))
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{
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Con_SafePrintf ("DS:GetCaps failed\n");
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FreeSound ();
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return SIS_FAILURE;
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}
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// if (snd_firsttime)
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// Con_SafePrintf ("Using secondary sound buffer\n");
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}
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else
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{
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// Removed the option for -primarysound
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// -- Robert S. Elsner <sockman@ngfc.com>
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#if 0
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if (DS_OK != pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_WRITEPRIMARY))
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{
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Con_SafePrintf ("Set coop level failed\n");
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FreeSound ();
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return SIS_FAILURE;
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}
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if (DS_OK != pDSPBuf->lpVtbl->GetCaps (pDSPBuf, &dsbcaps))
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{
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Con_Printf ("DS:GetCaps failed\n");
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return SIS_FAILURE;
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}
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pDSBuf = pDSPBuf;
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// Con_SafePrintf ("Using primary sound buffer\n");
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#endif
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}
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// Make sure mixer is active
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pDSBuf->lpVtbl->Play(pDSBuf, 0, 0, DSBPLAY_LOOPING);
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/* if (snd_firsttime)
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Con_SafePrintf(" %d channel(s)\n"
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" %d bits/sample\n"
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" %d bytes/sec\n",
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shm->channels, shm->samplebits, shm->speed);*/
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gSndBufSize = dsbcaps.dwBufferBytes;
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// initialize the buffer
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reps = 0;
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while ((hresult = pDSBuf->lpVtbl->Lock(pDSBuf, 0, gSndBufSize, &lpData, &dwSize, NULL, NULL, 0)) != DS_OK)
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{
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if (hresult != DSERR_BUFFERLOST)
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{
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Con_SafePrintf ("SNDDMA_InitDirect: DS::Lock Sound Buffer Failed\n");
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FreeSound ();
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return SIS_FAILURE;
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}
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if (++reps > 10000)
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{
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Con_SafePrintf ("SNDDMA_InitDirect: DS: couldn't restore buffer\n");
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FreeSound ();
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return SIS_FAILURE;
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}
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}
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memset(lpData, 0, dwSize);
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// lpData[4] = lpData[5] = 0x7f; // force a pop for debugging
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pDSBuf->lpVtbl->Unlock(pDSBuf, lpData, dwSize, NULL, 0);
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/* we don't want anyone to access the buffer directly w/o locking it first. */
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lpData = NULL;
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pDSBuf->lpVtbl->Stop(pDSBuf);
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pDSBuf->lpVtbl->GetCurrentPosition(pDSBuf, &mmstarttime.u.sample, &dwWrite);
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pDSBuf->lpVtbl->Play(pDSBuf, 0, 0, DSBPLAY_LOOPING);
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shm->soundalive = true;
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shm->splitbuffer = false;
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shm->samples = gSndBufSize/(shm->samplebits/8);
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shm->samplepos = 0;
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shm->submission_chunk = 1;
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shm->buffer = (unsigned char *) lpData;
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sample16 = (shm->samplebits/8) - 1;
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dsound_init = true;
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return SIS_SUCCESS;
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}
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#endif /* HAVE_DSOUND */
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/*
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==================
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SNDDM_InitWav
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Crappy windows multimedia base
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==================
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*/
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qboolean SNDDMA_InitWav (void)
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{
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WAVEFORMATEX format;
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int i;
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HRESULT hr;
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snd_sent = 0;
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snd_completed = 0;
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shm = &sn;
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shm->channels = 2;
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shm->samplebits = 16;
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shm->speed = 11025;
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memset (&format, 0, sizeof(format));
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format.wFormatTag = WAVE_FORMAT_PCM;
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format.nChannels = shm->channels;
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format.wBitsPerSample = shm->samplebits;
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format.nSamplesPerSec = shm->speed;
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format.nBlockAlign = format.nChannels
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*format.wBitsPerSample / 8;
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format.cbSize = 0;
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format.nAvgBytesPerSec = format.nSamplesPerSec
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*format.nBlockAlign;
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/* Open a waveform device for output using window callback. */
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while ((hr = waveOutOpen((LPHWAVEOUT)&hWaveOut, WAVE_MAPPER,
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&format,
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0, 0L, CALLBACK_NULL)) != MMSYSERR_NOERROR)
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{
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if (hr != MMSYSERR_ALLOCATED)
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{
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Con_SafePrintf ("waveOutOpen failed\n");
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return false;
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}
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if (MessageBox (NULL,
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"The sound hardware is in use by another app.\n\n"
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"Select Retry to try to start sound again or Cancel to run Quake with no sound.",
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"Sound not available",
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MB_RETRYCANCEL | MB_SETFOREGROUND | MB_ICONEXCLAMATION) != IDRETRY)
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{
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Con_SafePrintf ("waveOutOpen failure;\n"
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" hardware already in use\n");
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return false;
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}
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}
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/*
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* Allocate and lock memory for the waveform data. The memory
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* for waveform data must be globally allocated with
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* GMEM_MOVEABLE and GMEM_SHARE flags.
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*/
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gSndBufSize = WAV_BUFFERS*WAV_BUFFER_SIZE;
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hData = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, gSndBufSize);
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if (!hData)
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{
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Con_SafePrintf ("Sound: Out of memory.\n");
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FreeSound ();
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return false;
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}
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lpData = GlobalLock(hData);
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if (!lpData)
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{
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Con_SafePrintf ("Sound: Failed to lock.\n");
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FreeSound ();
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return false;
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}
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memset (lpData, 0, gSndBufSize);
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/*
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* Allocate and lock memory for the header. This memory must
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* also be globally allocated with GMEM_MOVEABLE and
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* GMEM_SHARE flags.
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*/
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hWaveHdr = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE,
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(DWORD) sizeof(WAVEHDR) * WAV_BUFFERS);
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if (hWaveHdr == NULL)
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{
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Con_SafePrintf ("Sound: Failed to Alloc header.\n");
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FreeSound ();
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return false;
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}
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lpWaveHdr = (LPWAVEHDR) GlobalLock(hWaveHdr);
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if (lpWaveHdr == NULL)
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{
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Con_SafePrintf ("Sound: Failed to lock header.\n");
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FreeSound ();
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return false;
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}
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memset (lpWaveHdr, 0, sizeof(WAVEHDR) * WAV_BUFFERS);
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/* After allocation, set up and prepare headers. */
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for (i=0 ; i<WAV_BUFFERS ; i++)
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{
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lpWaveHdr[i].dwBufferLength = WAV_BUFFER_SIZE;
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lpWaveHdr[i].lpData = lpData + i*WAV_BUFFER_SIZE;
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if (waveOutPrepareHeader(hWaveOut, lpWaveHdr+i, sizeof(WAVEHDR)) !=
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MMSYSERR_NOERROR)
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{
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Con_SafePrintf ("Sound: failed to prepare wave headers\n");
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FreeSound ();
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return false;
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}
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}
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shm->soundalive = true;
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shm->splitbuffer = false;
|
|
shm->samples = gSndBufSize/(shm->samplebits/8);
|
|
shm->samplepos = 0;
|
|
shm->submission_chunk = 1;
|
|
shm->buffer = (unsigned char *) lpData;
|
|
sample16 = (shm->samplebits/8) - 1;
|
|
|
|
wav_init = true;
|
|
|
|
return true;
|
|
}
|
|
|
|
/*
|
|
==================
|
|
SNDDMA_Init
|
|
|
|
Try to find a sound device to mix for.
|
|
Returns false if nothing is found.
|
|
==================
|
|
*/
|
|
|
|
qboolean SNDDMA_Init(void)
|
|
{
|
|
sndinitstat stat;
|
|
|
|
if (COM_CheckParm ("-wavonly"))
|
|
wavonly = true;
|
|
|
|
dsound_init = wav_init = 0;
|
|
|
|
stat = SIS_FAILURE; // assume DirectSound won't initialize
|
|
|
|
#ifdef HAVE_DSOUND
|
|
/* Init DirectSound */
|
|
if (!wavonly)
|
|
{
|
|
if (snd_firsttime || snd_isdirect)
|
|
{
|
|
stat = SNDDMA_InitDirect ();;
|
|
|
|
if (stat == SIS_SUCCESS)
|
|
{
|
|
snd_isdirect = true;
|
|
|
|
if (snd_firsttime)
|
|
Con_SafePrintf ("DirectSound initialized\n");
|
|
}
|
|
else
|
|
{
|
|
snd_isdirect = false;
|
|
Con_SafePrintf ("DirectSound failed to init\n");
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
|
|
// if DirectSound didn't succeed in initializing, try to initialize
|
|
// waveOut sound, unless DirectSound failed because the hardware is
|
|
// already allocated (in which case the user has already chosen not
|
|
// to have sound)
|
|
if (!dsound_init && (stat != SIS_NOTAVAIL))
|
|
{
|
|
if (snd_firsttime || snd_iswave)
|
|
{
|
|
|
|
snd_iswave = SNDDMA_InitWav ();
|
|
|
|
if (snd_iswave)
|
|
{
|
|
if (snd_firsttime)
|
|
Con_SafePrintf ("Wave sound initialized\n");
|
|
}
|
|
else
|
|
{
|
|
Con_SafePrintf ("Wave sound failed to init\n");
|
|
}
|
|
}
|
|
}
|
|
|
|
snd_firsttime = false;
|
|
|
|
if (!dsound_init && !wav_init)
|
|
{
|
|
if (snd_firsttime)
|
|
Con_SafePrintf ("No sound device initialized\n");
|
|
|
|
return 0;
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
/*
|
|
==============
|
|
SNDDMA_GetDMAPos
|
|
|
|
return the current sample position (in mono samples read)
|
|
inside the recirculating dma buffer, so the mixing code will know
|
|
how many sample are required to fill it up.
|
|
===============
|
|
*/
|
|
int SNDDMA_GetDMAPos(void)
|
|
{
|
|
int s;
|
|
#ifdef HAVE_DSOUND
|
|
MMTIME mmtime;
|
|
DWORD dwWrite;
|
|
|
|
if (dsound_init)
|
|
{
|
|
mmtime.wType = TIME_SAMPLES;
|
|
pDSBuf->lpVtbl->GetCurrentPosition(pDSBuf, &mmtime.u.sample, &dwWrite);
|
|
s = mmtime.u.sample - mmstarttime.u.sample;
|
|
} else
|
|
#endif /* HAVE_DSOUND */
|
|
if (wav_init) {
|
|
s = snd_sent * WAV_BUFFER_SIZE;
|
|
}
|
|
|
|
|
|
s >>= sample16;
|
|
|
|
s &= (shm->samples-1);
|
|
|
|
return s;
|
|
}
|
|
|
|
/*
|
|
==============
|
|
SNDDMA_Submit
|
|
|
|
Send sound to device if buffer isn't really the dma buffer
|
|
===============
|
|
*/
|
|
void SNDDMA_Submit(void)
|
|
{
|
|
LPWAVEHDR h;
|
|
int wResult;
|
|
|
|
if (!wav_init)
|
|
return;
|
|
|
|
//
|
|
// find which sound blocks have completed
|
|
//
|
|
while (1)
|
|
{
|
|
if ( snd_completed == snd_sent )
|
|
{
|
|
Con_DPrintf ("Sound overrun\n");
|
|
break;
|
|
}
|
|
|
|
if ( ! (lpWaveHdr[ snd_completed & WAV_MASK].dwFlags & WHDR_DONE) )
|
|
{
|
|
break;
|
|
}
|
|
|
|
snd_completed++; // this buffer has been played
|
|
}
|
|
|
|
//
|
|
// submit two new sound blocks
|
|
//
|
|
while (((snd_sent - snd_completed) >> sample16) < 4)
|
|
{
|
|
h = lpWaveHdr + ( snd_sent&WAV_MASK );
|
|
|
|
snd_sent++;
|
|
/*
|
|
* Now the data block can be sent to the output device. The
|
|
* waveOutWrite function returns immediately and waveform
|
|
* data is sent to the output device in the background.
|
|
*/
|
|
wResult = waveOutWrite(hWaveOut, h, sizeof(WAVEHDR));
|
|
|
|
if (wResult != MMSYSERR_NOERROR)
|
|
{
|
|
Con_SafePrintf ("Failed to write block to device\n");
|
|
FreeSound ();
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
==============
|
|
SNDDMA_Shutdown
|
|
|
|
Reset the sound device for exiting
|
|
===============
|
|
*/
|
|
void SNDDMA_Shutdown(void)
|
|
{
|
|
FreeSound ();
|
|
}
|
|
|