mirror of
https://git.code.sf.net/p/quake/quakeforge
synced 2024-11-05 20:50:43 +00:00
f6d047712d
subsystem in favor of Cvars*. These new cvars are: o snd_device defaults to "" which selects the default device of the system (eg, plug:0,0 for ALSA 0.9 or /dev/dsp for OSS) o snd_rate defaults to 0 which selects the system default rate. o snd_bits defaults to 0 which selects the system default bit depth. o snd_stereo defaults to 1 (0 is mono) * actually, not that thorough: alsa and oss only. The rest have just ws :/
725 lines
15 KiB
C
725 lines
15 KiB
C
/*
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snd_win.c
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(description)
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Copyright (C) 1996-1997 Id Software, Inc.
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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:
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Free Software Foundation, Inc.
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59 Temple Place - Suite 330
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Boston, MA 02111-1307, USA
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$Id$
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#define CINTERFACE
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#include "winquake.h"
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#include "qargs.h"
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#include "console.h"
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#include "sound.h"
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#define iDirectSoundCreate(a,b,c) pDirectSoundCreate(a,b,c)
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HRESULT (WINAPI * pDirectSoundCreate) (GUID FAR * lpGUID,
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LPDIRECTSOUND FAR * lplpDS,
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IUnknown FAR * pUnkOuter);
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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_isdirect, snd_iswave;
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static qboolean primary_format_set;
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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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LPDIRECTSOUND pDS;
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LPDIRECTSOUNDBUFFER pDSBuf, pDSPBuf;
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HINSTANCE hInstDS;
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sndinitstat SNDDMA_InitDirect (void);
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qboolean SNDDMA_InitWav (void);
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/*
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S_BlockSound
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*/
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void
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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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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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S_UnblockSound
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*/
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void
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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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snd_blocked--;
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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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void
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FreeSound (void)
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{
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int i;
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if (pDSBuf) {
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IDirectSoundBuffer_Stop (pDSBuf);
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IDirectSound_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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IDirectSound_Release (pDSPBuf);
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}
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if (pDS) {
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IDirectSound_SetCooperativeLevel (pDS, mainwindow, DSSCL_NORMAL);
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IDirectSound_Release (pDS);
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}
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if (hWaveOut) {
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waveOutReset (hWaveOut);
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if (lpWaveHdr) {
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for (i = 0; i < WAV_BUFFERS; i++)
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waveOutUnprepareHeader (hWaveOut, lpWaveHdr + i,
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sizeof (WAVEHDR));
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}
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waveOutClose (hWaveOut);
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if (hWaveHdr) {
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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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GlobalUnlock (hData);
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GlobalFree (hData);
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}
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}
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pDS = NULL;
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pDSBuf = NULL;
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pDSPBuf = NULL;
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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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SNDDMA_InitDirect
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Direct-Sound support
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*/
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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 * format.wBitsPerSample / 8;
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format.cbSize = 0;
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format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign;
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if (!hInstDS) {
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hInstDS = LoadLibrary ("dsound.dll");
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if (hInstDS == NULL) {
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Con_Printf ("Couldn't load dsound.dll\n");
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return SIS_FAILURE;
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}
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pDirectSoundCreate =
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(void *) GetProcAddress (hInstDS, "DirectSoundCreate");
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if (!pDirectSoundCreate) {
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Con_Printf ("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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if (hresult != DSERR_ALLOCATED) {
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Con_Printf ("DirectSound create failed\n");
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return SIS_FAILURE;
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}
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Con_Printf ("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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dscaps.dwSize = sizeof (dscaps);
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if (DS_OK != IDirectSound_GetCaps (pDS, &dscaps)) {
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Con_Printf ("Couldn't get DS caps\n");
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}
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if (dscaps.dwFlags & DSCAPS_EMULDRIVER) {
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Con_Printf ("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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if (DS_OK !=
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IDirectSound_SetCooperativeLevel (pDS, mainwindow, DSSCL_EXCLUSIVE)) {
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Con_Printf ("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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if (DS_OK ==
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IDirectSound_CreateSoundBuffer (pDS, &dsbuf, &pDSPBuf, NULL)) {
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pformat = format;
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if (DS_OK != IDirectSoundBuffer_SetFormat (pDSPBuf, &pformat)) {
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} else
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primary_format_set = true;
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}
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}
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if (!primary_format_set || !COM_CheckParm ("-primarysound")) {
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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 !=
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IDirectSound_CreateSoundBuffer (pDS, &dsbuf, &pDSBuf, NULL)) {
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Con_Printf ("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 != IDirectSound_GetCaps (pDSBuf, &dsbcaps)) {
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Con_Printf ("DS:GetCaps failed\n");
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FreeSound ();
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return SIS_FAILURE;
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}
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} else {
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if (DS_OK !=
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IDirectSound_SetCooperativeLevel (pDS, mainwindow,
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DSSCL_WRITEPRIMARY)) {
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Con_Printf ("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 != IDirectSound_GetCaps (pDSPBuf, &dsbcaps)) {
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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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}
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// Make sure mixer is active
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IDirectSoundBuffer_Play (pDSBuf, 0, 0, DSBPLAY_LOOPING);
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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 = IDirectSoundBuffer_Lock (pDSBuf, 0, gSndBufSize,
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(LPVOID *) & lpData, &dwSize,
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NULL, NULL, 0)) != DS_OK) {
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if (hresult != DSERR_BUFFERLOST) {
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Con_Printf ("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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Con_Printf ("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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IDirectSoundBuffer_Unlock (pDSBuf, lpData, dwSize, NULL, 0);
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/* we don't want anyone to access the buffer directly w/o locking it
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first. */
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lpData = NULL;
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IDirectSoundBuffer_Stop (pDSBuf);
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IDirectSoundBuffer_GetCurrentPosition (pDSBuf, &mmstarttime.u.sample,
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&dwWrite);
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IDirectSoundBuffer_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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/*
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SNDDM_InitWav
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Crappy windows multimedia base
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*/
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qboolean
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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 * format.wBitsPerSample / 8;
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format.cbSize = 0;
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format.nAvgBytesPerSec = format.nSamplesPerSec * 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, 0, 0L,
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CALLBACK_NULL)) != MMSYSERR_NOERROR) {
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if (hr != MMSYSERR_ALLOCATED) {
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Con_Printf ("waveOutOpen failed\n");
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return false;
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}
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Con_Printf ("waveOutOpen failure;\n" " hardware already in use\n");
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return false;
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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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Con_Printf ("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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Con_Printf ("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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Con_Printf ("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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Con_Printf ("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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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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Con_Printf ("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;
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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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wav_init = true;
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return true;
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}
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/*
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SNDDMA_Init
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Try to find a sound device to mix for.
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Returns false if nothing is found.
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*/
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qboolean
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SNDDMA_Init (void)
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{
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sndinitstat stat;
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if (COM_CheckParm ("-wavonly"))
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wavonly = true;
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dsound_init = wav_init = 0;
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stat = SIS_FAILURE; // assume DirectSound won't
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// initialize
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/* Init DirectSound */
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if (!wavonly) {
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if (snd_firsttime || snd_isdirect) {
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stat = SNDDMA_InitDirect ();;
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if (stat == SIS_SUCCESS) {
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snd_isdirect = true;
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if (snd_firsttime)
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Con_Printf ("DirectSound initialized\n");
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} else {
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snd_isdirect = false;
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Con_Printf ("DirectSound failed to init\n");
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}
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}
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}
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// if DirectSound didn't succeed in initializing, try to initialize
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// waveOut sound, unless DirectSound failed because the hardware is
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// already allocated (in which case the user has already chosen not
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// to have sound)
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if (!dsound_init && (stat != SIS_NOTAVAIL)) {
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if (snd_firsttime || snd_iswave) {
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snd_iswave = SNDDMA_InitWav ();
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if (snd_iswave) {
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if (snd_firsttime)
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Con_Printf ("Wave sound initialized\n");
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} else {
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Con_Printf ("Wave sound failed to init\n");
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}
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}
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}
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snd_firsttime = false;
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if (!dsound_init && !wav_init) {
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if (snd_firsttime)
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Con_Printf ("No sound device initialized\n");
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return 0;
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}
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return 1;
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}
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/*
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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)
|
|
{
|
|
MMTIME mmtime;
|
|
int s = 0;
|
|
DWORD dwWrite;
|
|
|
|
if (dsound_init) {
|
|
mmtime.wType = TIME_SAMPLES;
|
|
IDirectSoundBuffer_GetCurrentPosition (pDSBuf, &mmtime.u.sample,
|
|
&dwWrite);
|
|
s = mmtime.u.sample - mmstarttime.u.sample;
|
|
} else 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_Printf ("Failed to write block to device\n");
|
|
FreeSound ();
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
SNDDMA_Shutdown
|
|
|
|
Reset the sound device for exiting
|
|
*/
|
|
void
|
|
SNDDMA_Shutdown (void)
|
|
{
|
|
FreeSound ();
|
|
}
|
|
|
|
DWORD *
|
|
DSOUND_LockBuffer (qboolean lockit)
|
|
{
|
|
int reps;
|
|
|
|
static DWORD dwSize;
|
|
static DWORD dwSize2;
|
|
static DWORD *pbuf1;
|
|
static DWORD *pbuf2;
|
|
HRESULT hresult;
|
|
|
|
if (!pDSBuf)
|
|
return NULL;
|
|
|
|
if (lockit) {
|
|
reps = 0;
|
|
while ((hresult = IDirectSoundBuffer_Lock (pDSBuf, 0, gSndBufSize,
|
|
(LPVOID *) & pbuf1, &dwSize,
|
|
(LPVOID *) & pbuf2, &dwSize2,
|
|
0)) != DS_OK) {
|
|
if (hresult != DSERR_BUFFERLOST) {
|
|
Con_Printf
|
|
("S_TransferStereo16: DS::Lock Sound Buffer Failed\n");
|
|
S_Shutdown ();
|
|
S_Startup ();
|
|
return NULL;
|
|
}
|
|
|
|
if (++reps > 10000) {
|
|
Con_Printf
|
|
("S_TransferStereo16: DS: couldn't restore buffer\n");
|
|
S_Shutdown ();
|
|
S_Startup ();
|
|
return NULL;
|
|
}
|
|
}
|
|
} else {
|
|
IDirectSoundBuffer_Unlock (pDSBuf, pbuf1, dwSize, NULL, 0);
|
|
pbuf1 = NULL;
|
|
pbuf2 = NULL;
|
|
dwSize = 0;
|
|
dwSize2 = 0;
|
|
}
|
|
return (pbuf1);
|
|
}
|
|
|
|
void
|
|
DSOUND_ClearBuffer (int clear)
|
|
{
|
|
DWORD *pData;
|
|
|
|
// FIXME: this should be called with 2nd pbuf2 = NULL, dwsize =0
|
|
pData = DSOUND_LockBuffer (true);
|
|
memset (pData, clear, shm->samples * shm->samplebits / 8);
|
|
DSOUND_LockBuffer (false);
|
|
}
|
|
|
|
void
|
|
DSOUND_Restore (void)
|
|
{
|
|
// if the buffer was lost or stopped, restore it and/or restart it
|
|
DWORD dwStatus;
|
|
|
|
if (!pDSBuf)
|
|
return;
|
|
|
|
if (IDirectSoundBuffer_GetStatus (pDSBuf, &dwStatus) != DD_OK)
|
|
Con_Printf ("Couldn't get sound buffer status\n");
|
|
|
|
if (dwStatus & DSBSTATUS_BUFFERLOST)
|
|
IDirectSoundBuffer_Restore (pDSBuf);
|
|
|
|
if (!(dwStatus & DSBSTATUS_PLAYING))
|
|
IDirectSoundBuffer_Play (pDSBuf, 0, 0, DSBPLAY_LOOPING);
|
|
|
|
return;
|
|
}
|