mirror of
https://github.com/nzp-team/fteqw.git
synced 2024-11-10 22:51:57 +00:00
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
1230 lines
34 KiB
C
1230 lines
34 KiB
C
/*
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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 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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#include <dsound.h>
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#ifndef DECLSPEC_SELECTANY
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#define DECLSPEC_SELECTANY
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#endif
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#define FORCE_DEFINE_GUID(name, l, w1, w2, b1, b2, b3, b4, b5, b6, b7, b8) \
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EXTERN_C const GUID DECLSPEC_SELECTANY name \
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= { l, w1, w2, { b1, b2, b3, b4, b5, b6, b7, b8 } }
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#if _MSC_VER <= 1200
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DEFINE_GUID(IID_IKsPropertySet, 0x31efac30, 0x515c, 0x11d0, 0xa9, 0xaa, 0x00, 0xaa, 0x00, 0x61, 0xbe, 0x93);
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DEFINE_GUID(IID_IDirectSound, 0x279AFA83, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60);
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#else
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FORCE_DEFINE_GUID(IID_IDirectSound, 0x279AFA83, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60);
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FORCE_DEFINE_GUID(IID_IKsPropertySet, 0x31efac30, 0x515c, 0x11d0, 0xa9, 0xaa, 0x00, 0xaa, 0x00, 0x61, 0xbe, 0x93);
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#endif
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#ifdef AVAIL_DSOUND
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#define iDirectSoundCreate(a,b,c) pDirectSoundCreate(a,b,c)
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#define iDirectSoundEnumerate(a,b,c) pDirectSoundEnumerate(a,b)
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HRESULT (WINAPI *pDirectSoundCreate)(GUID FAR *lpGUID, LPDIRECTSOUND FAR *lplpDS, IUnknown FAR *pUnkOuter);
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HRESULT (WINAPI *pDirectSoundEnumerate)(LPDSENUMCALLBACKA lpCallback, LPVOID lpContext);
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#if defined(VOICECHAT)
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HRESULT (WINAPI *pDirectSoundCaptureCreate)(GUID FAR *lpGUID, LPDIRECTSOUNDCAPTURE FAR *lplpDS, IUnknown FAR *pUnkOuter);
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HRESULT (WINAPI *pDirectSoundCaptureEnumerate)(LPDSENUMCALLBACK lpDSEnumCallback, LPVOID lpContext);
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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 struct {
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LPDIRECTSOUND pDS;
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LPDIRECTSOUNDBUFFER pDSBuf;
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LPDIRECTSOUNDBUFFER pDSPBuf;
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DWORD gSndBufSize;
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DWORD mmstarttime;
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#ifdef _IKsPropertySet_
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LPKSPROPERTYSET EaxKsPropertiesSet;
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#endif
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} dshandle_t;
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HINSTANCE hInstDS;
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static void DSOUND_Restore(soundcardinfo_t *sc)
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{
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DWORD dwStatus;
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dshandle_t *dh = sc->handle;
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if (dh->pDSBuf->lpVtbl->GetStatus (dh->pDSBuf, &dwStatus) != ERROR_SUCCESS)
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Con_Printf ("Couldn't get sound buffer status\n");
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if (dwStatus & DSBSTATUS_BUFFERLOST)
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dh->pDSBuf->lpVtbl->Restore (dh->pDSBuf);
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if (!(dwStatus & DSBSTATUS_PLAYING))
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dh->pDSBuf->lpVtbl->Play(dh->pDSBuf, 0, 0, DSBPLAY_LOOPING);
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}
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static DWORD dsound_locksize;
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static void *DSOUND_Lock(soundcardinfo_t *sc, unsigned int *sampidx)
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{
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void *ret;
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int reps;
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DWORD dwSize2=0;
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DWORD *pbuf2;
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HRESULT hresult;
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dshandle_t *dh = sc->handle;
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dsound_locksize=0;
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reps = 0;
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while ((hresult = dh->pDSBuf->lpVtbl->Lock(dh->pDSBuf, 0, dh->gSndBufSize, (void**)&ret, &dsound_locksize,
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(void**)&pbuf2, &dwSize2, 0)) != DS_OK)
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{
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if (hresult != DSERR_BUFFERLOST)
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{
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Con_Printf ("S_TransferStereo16: DS::Lock Sound Buffer Failed\n");
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return NULL;
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}
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if (++reps > 10000)
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{
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Con_Printf ("S_TransferStereo16: DS: couldn't restore buffer\n");
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return NULL;
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}
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DSOUND_Restore(sc);
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}
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return ret;
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}
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//called when the mixer is done with it.
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static void DSOUND_Unlock(soundcardinfo_t *sc, void *buffer)
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{
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dshandle_t *dh = sc->handle;
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dh->pDSBuf->lpVtbl->Unlock(dh->pDSBuf, buffer, dsound_locksize, NULL, 0);
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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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//per device
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static void DSOUND_Shutdown_Internal (soundcardinfo_t *sc)
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{
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dshandle_t *dh = sc->handle;
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if (!dh)
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return;
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sc->handle = NULL;
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#ifdef _IKsPropertySet_
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if (dh->EaxKsPropertiesSet)
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{
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IKsPropertySet_Release(dh->EaxKsPropertiesSet);
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}
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#endif
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if (dh->pDSBuf)
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{
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dh->pDSBuf->lpVtbl->Stop(dh->pDSBuf);
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dh->pDSBuf->lpVtbl->Release(dh->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 (dh->pDSPBuf && (dh->pDSBuf != dh->pDSPBuf))
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{
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dh->pDSPBuf->lpVtbl->Release(dh->pDSPBuf);
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}
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if (dh->pDS)
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{
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dh->pDS->lpVtbl->SetCooperativeLevel (dh->pDS, mainwindow, DSSCL_NORMAL);
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dh->pDS->lpVtbl->Release(dh->pDS);
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}
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dh->pDS = NULL;
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dh->pDSBuf = NULL;
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dh->pDSPBuf = NULL;
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#ifdef _IKsPropertySet_
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dh->EaxKsPropertiesSet = NULL;
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#endif
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Z_Free(dh);
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}
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static void DSOUND_Shutdown (soundcardinfo_t *sc)
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{
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#ifdef MULTITHREAD
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if (sc->thread)
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{
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//thread does the actual closing.
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sc->selfpainting = false;
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Sys_WaitOnThread(sc->thread);
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sc->thread = NULL;
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}
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else
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#endif
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DSOUND_Shutdown_Internal(sc);
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}
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/*
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Direct Sound.
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These following defs should be moved to winquake.h somewhere.
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We tell DS to use a different wave format. We do this to gain extra channels. >2
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We still use the old stuff too, when we can for compatability.
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EAX 2 is also supported.
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This is a global state. Once applied, it's applied for other programs too.
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We have to do a few special things to try to ensure support in all it's different versions.
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*/
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/* new formatTag:*/
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#ifndef WAVE_FORMAT_EXTENSIBLE
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# define WAVE_FORMAT_EXTENSIBLE (0xfffe)
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#endif
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/* Speaker Positions:*/
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# define SPEAKER_FRONT_LEFT 0x1
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# define SPEAKER_FRONT_RIGHT 0x2
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# define SPEAKER_FRONT_CENTER 0x4
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# define SPEAKER_LOW_FREQUENCY 0x8
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# define SPEAKER_BACK_LEFT 0x10
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# define SPEAKER_BACK_RIGHT 0x20
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# define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
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# define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
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# define SPEAKER_BACK_CENTER 0x100
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# define SPEAKER_SIDE_LEFT 0x200
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# define SPEAKER_SIDE_RIGHT 0x400
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# define SPEAKER_TOP_CENTER 0x800
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# define SPEAKER_TOP_FRONT_LEFT 0x1000
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# define SPEAKER_TOP_FRONT_CENTER 0x2000
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# define SPEAKER_TOP_FRONT_RIGHT 0x4000
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# define SPEAKER_TOP_BACK_LEFT 0x8000
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# define SPEAKER_TOP_BACK_CENTER 0x10000
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# define SPEAKER_TOP_BACK_RIGHT 0x20000
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/* Bit mask locations reserved for future use*/
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# define SPEAKER_RESERVED 0x7FFC0000
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/* Used to specify that any possible permutation of speaker configurations*/
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# define SPEAKER_ALL 0x80000000
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/* DirectSound Speaker Config*/
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# define KSAUDIO_SPEAKER_MONO (SPEAKER_FRONT_CENTER)
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# define KSAUDIO_SPEAKER_STEREO (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)
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# define KSAUDIO_SPEAKER_QUAD (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
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SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
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# define KSAUDIO_SPEAKER_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
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SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER)
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# define KSAUDIO_SPEAKER_5POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
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SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
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SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
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# define KSAUDIO_SPEAKER_7POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
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SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
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SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
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SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)
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typedef struct {
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WAVEFORMATEX Format;
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union {
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WORD wValidBitsPerSample; /* bits of precision */
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WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */
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WORD wReserved; /* If neither applies, set to */
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/* zero. */
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} Samples;
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DWORD dwChannelMask; /* which channels are */
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/* present in stream */
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GUID SubFormat;
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} QWAVEFORMATEX;
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const static GUID QKSDATAFORMAT_SUBTYPE_PCM = {0x00000001,0x0000,0x0010,
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{0x80,
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0x00,
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0x00,
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0xaa,
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0x00,
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0x38,
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0x9b,
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0x71}};
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#ifdef _IKsPropertySet_
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const static GUID CLSID_EAXDIRECTSOUND = {0x4ff53b81, 0x1ce0, 0x11d3,
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{0xaa, 0xb8, 0x0, 0xa0, 0xc9, 0x59, 0x49, 0xd5}};
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const static GUID DSPROPSETID_EAX20_LISTENERPROPERTIES = {0x306a6a8, 0xb224, 0x11d2,
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{0x99, 0xe5, 0x0, 0x0, 0xe8, 0xd8, 0xc7, 0x22}};
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typedef struct _EAXLISTENERPROPERTIES
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{
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long lRoom; // room effect level at low frequencies
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long lRoomHF; // room effect high-frequency level re. low frequency level
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float flRoomRolloffFactor; // like DS3D flRolloffFactor but for room effect
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float flDecayTime; // reverberation decay time at low frequencies
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float flDecayHFRatio; // high-frequency to low-frequency decay time ratio
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long lReflections; // early reflections level relative to room effect
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float flReflectionsDelay; // initial reflection delay time
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long lReverb; // late reverberation level relative to room effect
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float flReverbDelay; // late reverberation delay time relative to initial reflection
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unsigned long dwEnvironment; // sets all listener properties
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float flEnvironmentSize; // environment size in meters
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float flEnvironmentDiffusion; // environment diffusion
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float flAirAbsorptionHF; // change in level per meter at 5 kHz
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unsigned long dwFlags; // modifies the behavior of properties
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} EAXLISTENERPROPERTIES, *LPEAXLISTENERPROPERTIES;
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enum
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{
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EAX_ENVIRONMENT_GENERIC,
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EAX_ENVIRONMENT_PADDEDCELL,
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EAX_ENVIRONMENT_ROOM,
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EAX_ENVIRONMENT_BATHROOM,
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EAX_ENVIRONMENT_LIVINGROOM,
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EAX_ENVIRONMENT_STONEROOM,
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EAX_ENVIRONMENT_AUDITORIUM,
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EAX_ENVIRONMENT_CONCERTHALL,
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EAX_ENVIRONMENT_CAVE,
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EAX_ENVIRONMENT_ARENA,
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EAX_ENVIRONMENT_HANGAR,
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EAX_ENVIRONMENT_CARPETEDHALLWAY,
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EAX_ENVIRONMENT_HALLWAY,
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EAX_ENVIRONMENT_STONECORRIDOR,
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EAX_ENVIRONMENT_ALLEY,
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EAX_ENVIRONMENT_FOREST,
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EAX_ENVIRONMENT_CITY,
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EAX_ENVIRONMENT_MOUNTAINS,
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EAX_ENVIRONMENT_QUARRY,
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EAX_ENVIRONMENT_PLAIN,
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EAX_ENVIRONMENT_PARKINGLOT,
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EAX_ENVIRONMENT_SEWERPIPE,
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EAX_ENVIRONMENT_UNDERWATER,
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EAX_ENVIRONMENT_DRUGGED,
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EAX_ENVIRONMENT_DIZZY,
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EAX_ENVIRONMENT_PSYCHOTIC,
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EAX_ENVIRONMENT_COUNT
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};
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typedef enum
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{
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DSPROPERTY_EAXLISTENER_NONE,
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DSPROPERTY_EAXLISTENER_ALLPARAMETERS,
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DSPROPERTY_EAXLISTENER_ROOM,
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DSPROPERTY_EAXLISTENER_ROOMHF,
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DSPROPERTY_EAXLISTENER_ROOMROLLOFFFACTOR,
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DSPROPERTY_EAXLISTENER_DECAYTIME,
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DSPROPERTY_EAXLISTENER_DECAYHFRATIO,
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DSPROPERTY_EAXLISTENER_REFLECTIONS,
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DSPROPERTY_EAXLISTENER_REFLECTIONSDELAY,
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DSPROPERTY_EAXLISTENER_REVERB,
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DSPROPERTY_EAXLISTENER_REVERBDELAY,
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DSPROPERTY_EAXLISTENER_ENVIRONMENT,
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DSPROPERTY_EAXLISTENER_ENVIRONMENTSIZE,
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DSPROPERTY_EAXLISTENER_ENVIRONMENTDIFFUSION,
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DSPROPERTY_EAXLISTENER_AIRABSORPTIONHF,
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DSPROPERTY_EAXLISTENER_FLAGS
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} DSPROPERTY_EAX_LISTENERPROPERTY;
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const static GUID DSPROPSETID_EAX20_BUFFERPROPERTIES ={
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0x306a6a7,
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0xb224,
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0x11d2,
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{0x99, 0xe5, 0x0, 0x0, 0xe8, 0xd8, 0xc7, 0x22}};
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const static GUID CLSID_EAXDirectSound ={
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0x4ff53b81,
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0x1ce0,
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0x11d3,
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{0xaa, 0xb8, 0x0, 0xa0, 0xc9, 0x59, 0x49, 0xd5}};
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typedef struct _EAXBUFFERPROPERTIES
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{
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long lDirect; // direct path level
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long lDirectHF; // direct path level at high frequencies
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long lRoom; // room effect level
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long lRoomHF; // room effect level at high frequencies
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float flRoomRolloffFactor; // like DS3D flRolloffFactor but for room effect
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long lObstruction; // main obstruction control (attenuation at high frequencies)
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float flObstructionLFRatio; // obstruction low-frequency level re. main control
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long lOcclusion; // main occlusion control (attenuation at high frequencies)
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float flOcclusionLFRatio; // occlusion low-frequency level re. main control
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float flOcclusionRoomRatio; // occlusion room effect level re. main control
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long lOutsideVolumeHF; // outside sound cone level at high frequencies
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float flAirAbsorptionFactor; // multiplies DSPROPERTY_EAXLISTENER_AIRABSORPTIONHF
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unsigned long dwFlags; // modifies the behavior of properties
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} EAXBUFFERPROPERTIES, *LPEAXBUFFERPROPERTIES;
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typedef enum
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{
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DSPROPERTY_EAXBUFFER_NONE,
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DSPROPERTY_EAXBUFFER_ALLPARAMETERS,
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DSPROPERTY_EAXBUFFER_DIRECT,
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DSPROPERTY_EAXBUFFER_DIRECTHF,
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DSPROPERTY_EAXBUFFER_ROOM,
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DSPROPERTY_EAXBUFFER_ROOMHF,
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DSPROPERTY_EAXBUFFER_ROOMROLLOFFFACTOR,
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DSPROPERTY_EAXBUFFER_OBSTRUCTION,
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DSPROPERTY_EAXBUFFER_OBSTRUCTIONLFRATIO,
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DSPROPERTY_EAXBUFFER_OCCLUSION,
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DSPROPERTY_EAXBUFFER_OCCLUSIONLFRATIO,
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DSPROPERTY_EAXBUFFER_OCCLUSIONROOMRATIO,
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DSPROPERTY_EAXBUFFER_OUTSIDEVOLUMEHF,
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DSPROPERTY_EAXBUFFER_AIRABSORPTIONFACTOR,
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DSPROPERTY_EAXBUFFER_FLAGS
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} DSPROPERTY_EAX_BUFFERPROPERTY;
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#endif
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static void DSOUND_SetReverb(soundcardinfo_t *sc, size_t reverb)
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{
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#ifdef _IKsPropertySet_
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dshandle_t *dh = sc->handle;
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|
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//attempt at eax support.
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//EAX is a global thing. Get it going in a game and your media player will be doing it too.
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if (dh->EaxKsPropertiesSet) //only on ds cards.
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{
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EAXLISTENERPROPERTIES ListenerProperties = {0};
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|
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/* DWORD p;
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IKsPropertySet_Get(dh->EaxKsPropertiesSet, &DSPROPSETID_EAX20_LISTENERPROPERTIES,
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DSPROPERTY_EAXLISTENER_ALLPARAMETERS, 0, 0, &ListenerProperties,
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sizeof(ListenerProperties), &p);
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*/
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if (reverb)
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{
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#if 1 //phycotic.
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ListenerProperties.flEnvironmentSize = 2.8;
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ListenerProperties.flEnvironmentDiffusion = 0.240;
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ListenerProperties.lRoom = -374;
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ListenerProperties.lRoomHF = -150;
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ListenerProperties.flRoomRolloffFactor = 0;
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ListenerProperties.flAirAbsorptionHF = -5;
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ListenerProperties.lReflections = -10000;
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ListenerProperties.flReflectionsDelay = 0.053;
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ListenerProperties.lReverb = 625;
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|
ListenerProperties.flReverbDelay = 0.08;
|
|
ListenerProperties.flDecayTime = 5.096;
|
|
ListenerProperties.flDecayHFRatio = 0.910;
|
|
ListenerProperties.dwFlags = 0x3f;
|
|
ListenerProperties.dwEnvironment = EAX_ENVIRONMENT_PSYCHOTIC;
|
|
#else
|
|
ListenerProperties.flEnvironmentSize = 5.8;
|
|
ListenerProperties.flEnvironmentDiffusion = 0;
|
|
ListenerProperties.lRoom = -374;
|
|
ListenerProperties.lRoomHF = -2860;
|
|
ListenerProperties.flRoomRolloffFactor = 0;
|
|
ListenerProperties.flAirAbsorptionHF = -5;
|
|
ListenerProperties.lReflections = -889;
|
|
ListenerProperties.flReflectionsDelay = 0.024;
|
|
ListenerProperties.lReverb = 797;
|
|
ListenerProperties.flReverbDelay = 0.035;
|
|
ListenerProperties.flDecayTime = 5.568;
|
|
ListenerProperties.flDecayHFRatio = 0.100;
|
|
ListenerProperties.dwFlags = 0x3f;
|
|
ListenerProperties.dwEnvironment = EAX_ENVIRONMENT_UNDERWATER;
|
|
#endif
|
|
}
|
|
else
|
|
{
|
|
ListenerProperties.flEnvironmentSize = 1;
|
|
ListenerProperties.flEnvironmentDiffusion = 0;
|
|
ListenerProperties.lRoom = 0;
|
|
ListenerProperties.lRoomHF = 0;
|
|
ListenerProperties.flRoomRolloffFactor = 0;
|
|
ListenerProperties.flAirAbsorptionHF = 0;
|
|
ListenerProperties.lReflections = 1000;
|
|
ListenerProperties.flReflectionsDelay = 0;
|
|
ListenerProperties.lReverb = 813;
|
|
ListenerProperties.flReverbDelay = 0.00;
|
|
ListenerProperties.flDecayTime = 0.1;
|
|
ListenerProperties.flDecayHFRatio = 0.1;
|
|
ListenerProperties.dwFlags = 0x3f;
|
|
ListenerProperties.dwEnvironment = EAX_ENVIRONMENT_GENERIC;
|
|
}
|
|
|
|
// env = EAX_ENVIRONMENT_UNDERWATER;
|
|
|
|
if (FAILED(IKsPropertySet_Set(dh->EaxKsPropertiesSet, &DSPROPSETID_EAX20_LISTENERPROPERTIES,
|
|
DSPROPERTY_EAXLISTENER_ALLPARAMETERS, 0, 0, &ListenerProperties,
|
|
sizeof(ListenerProperties))))
|
|
Con_SafePrintf ("EAX set failed\n");
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/*
|
|
==============
|
|
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.
|
|
===============
|
|
*/
|
|
static unsigned int DSOUND_GetDMAPos(soundcardinfo_t *sc)
|
|
{
|
|
DWORD mmtime;
|
|
int s;
|
|
DWORD dwWrite;
|
|
|
|
dshandle_t *dh = sc->handle;
|
|
|
|
dh->pDSBuf->lpVtbl->GetCurrentPosition(dh->pDSBuf, &mmtime, &dwWrite);
|
|
s = mmtime - dh->mmstarttime;
|
|
|
|
|
|
s >>= (sc->sn.samplebits/8) - 1;
|
|
|
|
s %= (sc->sn.samples);
|
|
|
|
return s;
|
|
}
|
|
|
|
/*
|
|
==============
|
|
SNDDMA_Submit
|
|
|
|
Send sound to device if buffer isn't really the dma buffer
|
|
===============
|
|
*/
|
|
static void DSOUND_Submit(soundcardinfo_t *sc, int start, int end)
|
|
{
|
|
}
|
|
|
|
static qboolean DSOUND_InitOutputLibrary(void)
|
|
{
|
|
if (!hInstDS)
|
|
hInstDS = LoadLibrary("dsound.dll");
|
|
if (!hInstDS)
|
|
{
|
|
Con_SafePrintf ("Couldn't load dsound.dll\n");
|
|
return false;
|
|
}
|
|
if (!pDirectSoundCreate)
|
|
pDirectSoundCreate = (void *)GetProcAddress(hInstDS,"DirectSoundCreate");
|
|
if (!pDirectSoundCreate)
|
|
{
|
|
Con_SafePrintf ("Couldn't get DS proc addr\n");
|
|
return false;
|
|
}
|
|
if (!pDirectSoundEnumerate)
|
|
pDirectSoundEnumerate = (void *)GetProcAddress(hInstDS,"DirectSoundEnumerateA");
|
|
return true;
|
|
}
|
|
/*
|
|
==================
|
|
SNDDMA_InitDirect
|
|
|
|
Direct-Sound support
|
|
==================
|
|
*/
|
|
static int DSOUND_InitCard_Internal (soundcardinfo_t *sc, char *cardname)
|
|
{
|
|
extern cvar_t snd_inactive;
|
|
#if _MSC_VER > 1200 //fixme err
|
|
#ifdef _IKsPropertySet_
|
|
extern cvar_t snd_eax;
|
|
#endif
|
|
#endif
|
|
DSBUFFERDESC dsbuf;
|
|
DSBCAPS dsbcaps;
|
|
DWORD dwSize, dwWrite;
|
|
DSCAPS dscaps;
|
|
QWAVEFORMATEX format, pformat;
|
|
HRESULT hresult;
|
|
int reps;
|
|
qboolean primary_format_set;
|
|
dshandle_t *dh;
|
|
char *buffer;
|
|
GUID guid, *dsguid;
|
|
|
|
|
|
memset (&format, 0, sizeof(format));
|
|
|
|
if (*sc->name)
|
|
{
|
|
wchar_t mssuck[128];
|
|
mbstowcs(mssuck, sc->name, sizeof(mssuck)/sizeof(mssuck[0])-1);
|
|
CLSIDFromString(mssuck, &guid);
|
|
dsguid = &guid;
|
|
}
|
|
else
|
|
{
|
|
memset(&guid, 0, sizeof(GUID));
|
|
dsguid = NULL;
|
|
}
|
|
|
|
if (sc->sn.numchannels >= 8) // 7.1 surround
|
|
{
|
|
format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
|
|
format.Format.cbSize = 22;
|
|
memcpy(&format.SubFormat, &QKSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID));
|
|
|
|
format.dwChannelMask = KSAUDIO_SPEAKER_7POINT1;
|
|
sc->sn.numchannels = 8;
|
|
}
|
|
else if (sc->sn.numchannels >= 6) //5.1 surround
|
|
{
|
|
format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
|
|
format.Format.cbSize = 22;
|
|
memcpy(&format.SubFormat, &QKSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID));
|
|
|
|
format.dwChannelMask = KSAUDIO_SPEAKER_5POINT1;
|
|
sc->sn.numchannels = 6;
|
|
}
|
|
else if (sc->sn.numchannels >= 4) //4 speaker quad
|
|
{
|
|
format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
|
|
format.Format.cbSize = 22;
|
|
memcpy(&format.SubFormat, &QKSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID));
|
|
|
|
format.dwChannelMask = KSAUDIO_SPEAKER_QUAD;
|
|
sc->sn.numchannels = 4;
|
|
}
|
|
else if (sc->sn.numchannels >= 2) //stereo
|
|
{
|
|
format.Format.wFormatTag = WAVE_FORMAT_PCM;
|
|
format.Format.cbSize = 0;
|
|
sc->sn.numchannels = 2;
|
|
}
|
|
else //mono time
|
|
{
|
|
format.Format.wFormatTag = WAVE_FORMAT_PCM;
|
|
format.Format.cbSize = 0;
|
|
sc->sn.numchannels = 1;
|
|
}
|
|
|
|
format.Format.nChannels = sc->sn.numchannels;
|
|
format.Format.wBitsPerSample = sc->sn.samplebits;
|
|
format.Format.nSamplesPerSec = sc->sn.speed;
|
|
format.Format.nBlockAlign = format.Format.nChannels
|
|
*format.Format.wBitsPerSample / 8;
|
|
format.Format.nAvgBytesPerSec = format.Format.nSamplesPerSec
|
|
*format.Format.nBlockAlign;
|
|
|
|
if (!DSOUND_InitOutputLibrary())
|
|
return false;
|
|
|
|
sc->handle = Z_Malloc(sizeof(dshandle_t));
|
|
dh = sc->handle;
|
|
//EAX attempt
|
|
#if _MSC_VER > 1200
|
|
#ifdef _IKsPropertySet_
|
|
dh->pDS = NULL;
|
|
if (snd_eax.ival)
|
|
{
|
|
CoInitialize(NULL);
|
|
if (FAILED(CoCreateInstance( &CLSID_EAXDirectSound, NULL, CLSCTX_INPROC_SERVER, &IID_IDirectSound, (void **)&dh->pDS )))
|
|
dh->pDS=NULL;
|
|
else
|
|
{
|
|
IDirectSound_Initialize(dh->pDS, dsguid);
|
|
}
|
|
}
|
|
|
|
if (!dh->pDS)
|
|
#endif
|
|
#endif
|
|
{
|
|
while ((hresult = iDirectSoundCreate(dsguid, &dh->pDS, NULL)) != DS_OK)
|
|
{
|
|
if (hresult != DSERR_ALLOCATED)
|
|
{
|
|
Con_SafePrintf (": create failed\n");
|
|
return false;
|
|
}
|
|
|
|
// if (MessageBox (NULL,
|
|
// "The sound hardware is in use by another app.\n\n"
|
|
// "Select Retry to try to start sound again or Cancel to run Quake with no sound.",
|
|
// "Sound not available",
|
|
// MB_RETRYCANCEL | MB_SETFOREGROUND | MB_ICONEXCLAMATION) != IDRETRY)
|
|
// {
|
|
Con_SafePrintf (": failure\n"
|
|
" hardware already in use\n"
|
|
" Close the other app then use snd_restart\n");
|
|
return false;
|
|
// }
|
|
}
|
|
}
|
|
|
|
#ifdef FTE_SDL
|
|
#define mainwindow GetDesktopWindow()
|
|
#endif
|
|
if (DS_OK != dh->pDS->lpVtbl->SetCooperativeLevel (dh->pDS, mainwindow, DSSCL_EXCLUSIVE))
|
|
{
|
|
Con_SafePrintf ("Set coop level failed\n");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
|
|
dscaps.dwSize = sizeof(dscaps);
|
|
|
|
if (DS_OK != dh->pDS->lpVtbl->GetCaps (dh->pDS, &dscaps))
|
|
{
|
|
Con_SafePrintf ("Couldn't get DS caps\n");
|
|
}
|
|
|
|
if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
|
|
{
|
|
Con_SafePrintf ("No DirectSound driver installed\n");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
|
|
|
|
// get access to the primary buffer, if possible, so we can set the
|
|
// sound hardware format
|
|
memset (&dsbuf, 0, sizeof(dsbuf));
|
|
dsbuf.dwSize = sizeof(DSBUFFERDESC);
|
|
dsbuf.dwFlags = DSBCAPS_PRIMARYBUFFER|DSBCAPS_CTRLVOLUME;
|
|
dsbuf.dwBufferBytes = 0;
|
|
dsbuf.lpwfxFormat = NULL;
|
|
|
|
#ifdef DSBCAPS_GLOBALFOCUS
|
|
#ifndef FTE_SDL
|
|
if (snd_inactive.ival || sys_parentwindow
|
|
) /*always inactive if we have a parent window, because we can't tell properly otherwise*/
|
|
#endif
|
|
{
|
|
dsbuf.dwFlags |= DSBCAPS_GLOBALFOCUS;
|
|
sc->inactive_sound = true;
|
|
}
|
|
#endif
|
|
|
|
memset(&dsbcaps, 0, sizeof(dsbcaps));
|
|
dsbcaps.dwSize = sizeof(dsbcaps);
|
|
primary_format_set = false;
|
|
|
|
if (!COM_CheckParm ("-snoforceformat"))
|
|
{
|
|
if (DS_OK == dh->pDS->lpVtbl->CreateSoundBuffer(dh->pDS, &dsbuf, &dh->pDSPBuf, NULL))
|
|
{
|
|
pformat = format;
|
|
|
|
if (DS_OK != dh->pDSPBuf->lpVtbl->SetFormat (dh->pDSPBuf, (WAVEFORMATEX *)&pformat))
|
|
{
|
|
// if (snd_firsttime)
|
|
// Con_SafePrintf ("Set primary sound buffer format: no\n");
|
|
}
|
|
else
|
|
// {
|
|
// if (snd_firsttime)
|
|
// Con_SafePrintf ("Set primary sound buffer format: yes\n");
|
|
|
|
primary_format_set = true;
|
|
// }
|
|
}
|
|
}
|
|
|
|
if (!primary_format_set || !COM_CheckParm ("-primarysound"))
|
|
{
|
|
// create the secondary buffer we'll actually work with
|
|
memset (&dsbuf, 0, sizeof(dsbuf));
|
|
dsbuf.dwSize = sizeof(DSBUFFERDESC);
|
|
dsbuf.dwFlags = DSBCAPS_CTRLFREQUENCY|DSBCAPS_LOCSOFTWARE; //dmw 29 may, 2003 removed locsoftware
|
|
#ifdef DSBCAPS_GLOBALFOCUS
|
|
if (snd_inactive.ival)
|
|
{
|
|
dsbuf.dwFlags |= DSBCAPS_GLOBALFOCUS;
|
|
sc->inactive_sound = true;
|
|
}
|
|
#endif
|
|
dsbuf.dwBufferBytes = sc->sn.samples / format.Format.nChannels;
|
|
if (!dsbuf.dwBufferBytes)
|
|
{
|
|
dsbuf.dwBufferBytes = SECONDARY_BUFFER_SIZE;
|
|
// the fast rates will need a much bigger buffer
|
|
if (format.Format.nSamplesPerSec > 48000)
|
|
dsbuf.dwBufferBytes *= 4;
|
|
}
|
|
dsbuf.lpwfxFormat = (WAVEFORMATEX *)&format;
|
|
|
|
memset(&dsbcaps, 0, sizeof(dsbcaps));
|
|
dsbcaps.dwSize = sizeof(dsbcaps);
|
|
|
|
if (DS_OK != dh->pDS->lpVtbl->CreateSoundBuffer(dh->pDS, &dsbuf, &dh->pDSBuf, NULL))
|
|
{
|
|
Con_SafePrintf ("DS:CreateSoundBuffer Failed");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
|
|
sc->sn.numchannels = format.Format.nChannels;
|
|
sc->sn.samplebits = format.Format.wBitsPerSample;
|
|
sc->sn.speed = format.Format.nSamplesPerSec;
|
|
|
|
if (DS_OK != dh->pDSBuf->lpVtbl->GetCaps (dh->pDSBuf, &dsbcaps))
|
|
{
|
|
Con_SafePrintf ("DS:GetCaps failed\n");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
|
|
// if (snd_firsttime)
|
|
// Con_SafePrintf ("Using secondary sound buffer\n");
|
|
}
|
|
else
|
|
{
|
|
if (DS_OK != dh->pDS->lpVtbl->SetCooperativeLevel (dh->pDS, mainwindow, DSSCL_WRITEPRIMARY))
|
|
{
|
|
Con_SafePrintf ("Set coop level failed\n");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
|
|
if (DS_OK != dh->pDSPBuf->lpVtbl->GetCaps (dh->pDSPBuf, &dsbcaps))
|
|
{
|
|
Con_Printf ("DS:GetCaps failed\n");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
|
|
dh->pDSBuf = dh->pDSPBuf;
|
|
// Con_SafePrintf ("Using primary sound buffer\n");
|
|
}
|
|
|
|
dh->gSndBufSize = dsbcaps.dwBufferBytes;
|
|
|
|
#if 1
|
|
// Make sure mixer is active
|
|
dh->pDSBuf->lpVtbl->Play(dh->pDSBuf, 0, 0, DSBPLAY_LOOPING);
|
|
|
|
/* if (snd_firsttime)
|
|
Con_SafePrintf(" %d channel(s)\n"
|
|
" %d bits/sample\n"
|
|
" %d bytes/sec\n",
|
|
shm->channels, shm->samplebits, shm->speed);*/
|
|
|
|
|
|
// initialize the buffer
|
|
reps = 0;
|
|
|
|
while ((hresult = dh->pDSBuf->lpVtbl->Lock(dh->pDSBuf, 0, dh->gSndBufSize, (void**)&buffer, &dwSize, NULL, NULL, 0)) != DS_OK)
|
|
{
|
|
if (hresult != DSERR_BUFFERLOST)
|
|
{
|
|
Con_SafePrintf ("SNDDMA_InitDirect: DS::Lock Sound Buffer Failed\n");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
|
|
if (++reps > 10000)
|
|
{
|
|
Con_SafePrintf ("SNDDMA_InitDirect: DS: couldn't restore buffer\n");
|
|
DSOUND_Shutdown_Internal (sc);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
memset(buffer, 0, dwSize);
|
|
// lpData[4] = lpData[5] = 0x7f; // force a pop for debugging
|
|
|
|
// Sleep(500);
|
|
|
|
dh->pDSBuf->lpVtbl->Unlock(dh->pDSBuf, buffer, dwSize, NULL, 0);
|
|
|
|
|
|
dh->pDSBuf->lpVtbl->Stop(dh->pDSBuf);
|
|
#endif
|
|
dh->pDSBuf->lpVtbl->GetCurrentPosition(dh->pDSBuf, &dh->mmstarttime, &dwWrite);
|
|
dh->pDSBuf->lpVtbl->Play(dh->pDSBuf, 0, 0, DSBPLAY_LOOPING);
|
|
|
|
sc->sn.samples = dh->gSndBufSize/(sc->sn.samplebits/8);
|
|
sc->sn.samplepos = 0;
|
|
sc->sn.buffer = NULL;
|
|
|
|
|
|
sc->Lock = DSOUND_Lock;
|
|
sc->Unlock = DSOUND_Unlock;
|
|
sc->SetEnvironmentReverb = DSOUND_SetReverb;
|
|
sc->Submit = DSOUND_Submit;
|
|
sc->Shutdown = DSOUND_Shutdown;
|
|
sc->GetDMAPos = DSOUND_GetDMAPos;
|
|
sc->Restore = DSOUND_Restore;
|
|
|
|
#if _MSC_VER > 1200
|
|
#ifdef _IKsPropertySet_
|
|
//attempt at eax support
|
|
if (snd_eax.ival)
|
|
{
|
|
int r;
|
|
DWORD support;
|
|
|
|
if (SUCCEEDED(IDirectSoundBuffer_QueryInterface(dh->pDSBuf, &IID_IKsPropertySet, (void*)&dh->EaxKsPropertiesSet)))
|
|
{
|
|
r = IKsPropertySet_QuerySupport(dh->EaxKsPropertiesSet, &DSPROPSETID_EAX20_LISTENERPROPERTIES, DSPROPERTY_EAXLISTENER_ALLPARAMETERS, &support);
|
|
if(!SUCCEEDED(r) || (support&(KSPROPERTY_SUPPORT_GET|KSPROPERTY_SUPPORT_SET))
|
|
!= (KSPROPERTY_SUPPORT_GET|KSPROPERTY_SUPPORT_SET))
|
|
{
|
|
IKsPropertySet_Release(dh->EaxKsPropertiesSet);
|
|
dh->EaxKsPropertiesSet = NULL;
|
|
Con_SafePrintf ("EAX 2 not supported\n");
|
|
return true;//otherwise successful. It can be used for normal sound anyway.
|
|
}
|
|
|
|
//worked. EAX is supported.
|
|
}
|
|
else
|
|
{
|
|
Con_SafePrintf ("Couldn't get extended properties\n");
|
|
dh->EaxKsPropertiesSet = NULL;
|
|
}
|
|
}
|
|
#endif
|
|
#endif
|
|
|
|
return true;
|
|
}
|
|
|
|
|
|
|
|
|
|
#ifdef MULTITHREAD
|
|
static int DSOUND_Thread(void *arg)
|
|
{
|
|
soundcardinfo_t *sc = arg;
|
|
void *cond = sc->handle;
|
|
sc->handle = NULL;
|
|
|
|
//once creating the thread, the main thread will wait for us to signal that we have inited the dsound device.
|
|
if (!DSOUND_InitCard_Internal(sc, sc->name))
|
|
sc->selfpainting = false;
|
|
|
|
//wake up the main thread.
|
|
Sys_ConditionSignal(cond);
|
|
|
|
while(sc->selfpainting)
|
|
{
|
|
S_MixerThread(sc);
|
|
/* Quote:
|
|
On NT (Win2K and XP) the cursors in SW buffers (and HW buffers on some devices) move in 10ms increments, so calling GetCurrentPosition() every 10ms is ideal.
|
|
Calling it more often than every 5ms will cause some perf degradation.
|
|
*/
|
|
Sleep(9);
|
|
}
|
|
|
|
//we created the device, we need to kill it.
|
|
DSOUND_Shutdown_Internal(sc);
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
static qboolean QDECL DSOUND_InitCard (soundcardinfo_t *sc, const char *device)
|
|
{
|
|
if (COM_CheckParm("-wavonly"))
|
|
return false;
|
|
|
|
Q_strncpyz(sc->name, device?device:"", sizeof(sc->name));
|
|
|
|
#ifdef MULTITHREAD
|
|
if (snd_mixerthread.ival)
|
|
{
|
|
void *cond;
|
|
sc->selfpainting = true;
|
|
sc->handle = cond = Sys_CreateConditional();
|
|
Sys_LockConditional(cond);
|
|
sc->thread = Sys_CreateThread("dsoundmixer", DSOUND_Thread, sc, THREADP_HIGHEST, 0);
|
|
if (!sc->thread)
|
|
{
|
|
Con_SafePrintf ("Unable to create sound mixing thread\n");
|
|
return false;
|
|
}
|
|
|
|
//wait for the thread to finish (along with all its error con printfs etc
|
|
if (!Sys_ConditionWait(cond))
|
|
Con_SafePrintf ("Looks like the sound thread isn't starting up\n");
|
|
Sys_UnlockConditional(cond);
|
|
Sys_DestroyConditional(cond);
|
|
|
|
if (!sc->selfpainting)
|
|
{
|
|
Sys_WaitOnThread(sc->thread);
|
|
sc->thread = NULL;
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
else
|
|
#endif
|
|
return DSOUND_InitCard_Internal(sc, sc->name);
|
|
}
|
|
|
|
#define SDRVNAME "DSound"
|
|
static BOOL (CALLBACK DSound_EnumCallback)(GUID FAR *guid, LPCSTR str1, LPCSTR str2, LPVOID parm)
|
|
{
|
|
char guidbuf[128];
|
|
wchar_t mssuck[128];
|
|
void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename) = parm;
|
|
if (guid == NULL) //we don't care about the (dupe) default device
|
|
return TRUE;
|
|
|
|
StringFromGUID2(guid, mssuck, sizeof(mssuck)/sizeof(mssuck[0]));
|
|
wcstombs(guidbuf, mssuck, sizeof(guidbuf));
|
|
callback(SDRVNAME, guidbuf, va("DS: %s", str1));
|
|
return TRUE;
|
|
}
|
|
static qboolean QDECL DSOUND_Enumerate(void (QDECL *cb) (const char *drivername, const char *devicecode, const char *readablename))
|
|
{
|
|
if (!DSOUND_InitOutputLibrary())
|
|
return false;
|
|
if (pDirectSoundEnumerate)
|
|
{
|
|
pDirectSoundEnumerate(&DSound_EnumCallback, cb);
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
sounddriver_t DSOUND_Output =
|
|
{
|
|
SDRVNAME,
|
|
DSOUND_InitCard,
|
|
DSOUND_Enumerate
|
|
};
|
|
|
|
#endif
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
#if defined(VOICECHAT) && defined(AVAIL_DSOUND)
|
|
|
|
|
|
typedef struct
|
|
{
|
|
LPDIRECTSOUNDCAPTURE DSCapture;
|
|
LPDIRECTSOUNDCAPTUREBUFFER DSCaptureBuffer;
|
|
long lastreadpos;
|
|
} dsndcapture_t;
|
|
static const long bufferbytes = 1024*1024;
|
|
|
|
static const long inputwidth = 2;
|
|
|
|
static BOOL CALLBACK dsound_capture_enumerate_ds(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext)
|
|
{
|
|
char guidbuf[128];
|
|
wchar_t mssuck[128];
|
|
void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename) = lpContext;
|
|
|
|
if (lpGuid == NULL) //we don't care about the (dupe) default device
|
|
return TRUE;
|
|
|
|
StringFromGUID2(lpGuid, mssuck, sizeof(mssuck)/sizeof(mssuck[0]));
|
|
wcstombs(guidbuf, mssuck, sizeof(guidbuf));
|
|
callback(SDRVNAME, guidbuf, lpcstrDescription);
|
|
return TRUE;
|
|
}
|
|
|
|
static qboolean QDECL DSOUND_Capture_Enumerate (void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename))
|
|
{
|
|
if (!pDirectSoundCaptureEnumerate)
|
|
{
|
|
/*make sure its loaded*/
|
|
if (!hInstDS)
|
|
hInstDS = LoadLibrary("dsound.dll");
|
|
if (hInstDS)
|
|
pDirectSoundCaptureEnumerate = (void *)GetProcAddress(hInstDS,"DirectSoundCaptureEnumerateA");
|
|
|
|
if (!pDirectSoundCaptureEnumerate)
|
|
return false;
|
|
}
|
|
|
|
if (!FAILED(pDirectSoundCaptureEnumerate(dsound_capture_enumerate_ds, callback)))
|
|
return true;
|
|
return false;
|
|
}
|
|
|
|
static void *QDECL DSOUND_Capture_Init (int rate, const char *device)
|
|
{
|
|
dsndcapture_t *result;
|
|
DSCBUFFERDESC bufdesc;
|
|
|
|
WAVEFORMATEX wfxFormat;
|
|
GUID *dsguid, guid;
|
|
|
|
wfxFormat.wFormatTag = WAVE_FORMAT_PCM;
|
|
wfxFormat.nChannels = 1;
|
|
wfxFormat.nSamplesPerSec = rate;
|
|
wfxFormat.wBitsPerSample = 8*inputwidth;
|
|
wfxFormat.nBlockAlign = wfxFormat.nChannels * (wfxFormat.wBitsPerSample / 8);
|
|
wfxFormat.nAvgBytesPerSec = wfxFormat.nSamplesPerSec * wfxFormat.nBlockAlign;
|
|
wfxFormat.cbSize = 0;
|
|
|
|
bufdesc.dwSize = sizeof(bufdesc);
|
|
bufdesc.dwBufferBytes = bufferbytes;
|
|
bufdesc.dwFlags = 0;
|
|
bufdesc.dwReserved = 0;
|
|
bufdesc.lpwfxFormat = &wfxFormat;
|
|
|
|
if (device && *device)
|
|
{
|
|
wchar_t mssuck[128];
|
|
mbstowcs(mssuck, device, sizeof(mssuck)/sizeof(mssuck[0])-1);
|
|
CLSIDFromString(mssuck, &guid);
|
|
dsguid = &guid;
|
|
}
|
|
else
|
|
{
|
|
memset(&guid, 0, sizeof(GUID));
|
|
dsguid = NULL;
|
|
}
|
|
|
|
/*probably already inited*/
|
|
if (!hInstDS)
|
|
{
|
|
hInstDS = LoadLibrary("dsound.dll");
|
|
|
|
if (hInstDS == NULL)
|
|
{
|
|
Con_SafePrintf ("Couldn't load dsound.dll\n");
|
|
return NULL;
|
|
}
|
|
}
|
|
/*global pointer, used only in this function*/
|
|
if (!pDirectSoundCaptureCreate)
|
|
{
|
|
pDirectSoundCaptureCreate = (void *)GetProcAddress(hInstDS,"DirectSoundCaptureCreate");
|
|
|
|
if (!pDirectSoundCaptureCreate)
|
|
{
|
|
Con_SafePrintf ("Couldn't get DS proc addr\n");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
result = Z_Malloc(sizeof(*result));
|
|
if (!FAILED(pDirectSoundCaptureCreate(dsguid, &result->DSCapture, NULL)))
|
|
{
|
|
if (!FAILED(IDirectSoundCapture_CreateCaptureBuffer(result->DSCapture, &bufdesc, &result->DSCaptureBuffer, NULL)))
|
|
{
|
|
return result;
|
|
}
|
|
IDirectSoundCapture_Release(result->DSCapture);
|
|
Con_SafePrintf ("Couldn't create a capture buffer\n");
|
|
}
|
|
Z_Free(result);
|
|
return NULL;
|
|
}
|
|
|
|
static void QDECL DSOUND_Capture_Start(void *ctx)
|
|
{
|
|
DWORD capturePos;
|
|
dsndcapture_t *c = ctx;
|
|
IDirectSoundCaptureBuffer_Start(c->DSCaptureBuffer, DSBPLAY_LOOPING);
|
|
|
|
c->lastreadpos = 0;
|
|
IDirectSoundCaptureBuffer_GetCurrentPosition(c->DSCaptureBuffer, &capturePos, &c->lastreadpos);
|
|
}
|
|
|
|
static void QDECL DSOUND_Capture_Stop(void *ctx)
|
|
{
|
|
dsndcapture_t *c = ctx;
|
|
IDirectSoundCaptureBuffer_Stop(c->DSCaptureBuffer);
|
|
}
|
|
|
|
static void QDECL DSOUND_Capture_Shutdown(void *ctx)
|
|
{
|
|
dsndcapture_t *c = ctx;
|
|
if (c->DSCaptureBuffer)
|
|
{
|
|
IDirectSoundCaptureBuffer_Stop(c->DSCaptureBuffer);
|
|
IDirectSoundCaptureBuffer_Release(c->DSCaptureBuffer);
|
|
}
|
|
if (c->DSCapture)
|
|
{
|
|
IDirectSoundCapture_Release(c->DSCapture);
|
|
}
|
|
Z_Free(ctx);
|
|
}
|
|
|
|
/*minsamples is a hint*/
|
|
static unsigned int QDECL DSOUND_Capture_Update(void *ctx, unsigned char *buffer, unsigned int minbytes, unsigned int maxbytes)
|
|
{
|
|
dsndcapture_t *c = ctx;
|
|
HRESULT hr;
|
|
LPBYTE lpbuf1 = NULL;
|
|
LPBYTE lpbuf2 = NULL;
|
|
DWORD dwsize1 = 0;
|
|
DWORD dwsize2 = 0;
|
|
|
|
DWORD capturePos;
|
|
DWORD readPos;
|
|
long filled;
|
|
|
|
// Query to see how much data is in buffer.
|
|
hr = IDirectSoundCaptureBuffer_GetCurrentPosition(c->DSCaptureBuffer, &capturePos, &readPos);
|
|
if (hr != DS_OK)
|
|
{
|
|
return 0;
|
|
}
|
|
filled = readPos - c->lastreadpos;
|
|
if (filled < 0)
|
|
filled += bufferbytes; // unwrap offset
|
|
|
|
if (filled > maxbytes) //figure out how much we need to empty it by, and if that's enough to be worthwhile.
|
|
filled = maxbytes;
|
|
else if (filled < minbytes)
|
|
return 0;
|
|
|
|
// filled /= inputwidth;
|
|
// filled *= inputwidth;
|
|
|
|
// Lock free space in the DS
|
|
hr = IDirectSoundCaptureBuffer_Lock(c->DSCaptureBuffer, c->lastreadpos, filled, (void **) &lpbuf1, &dwsize1, (void **) &lpbuf2, &dwsize2, 0);
|
|
if (hr == DS_OK)
|
|
{
|
|
// Copy from DS to the buffer
|
|
memcpy(buffer, lpbuf1, dwsize1);
|
|
if(lpbuf2 != NULL)
|
|
{
|
|
memcpy(buffer+dwsize1, lpbuf2, dwsize2);
|
|
}
|
|
// Update our buffer offset and unlock sound buffer
|
|
c->lastreadpos = (c->lastreadpos + dwsize1 + dwsize2) % bufferbytes;
|
|
IDirectSoundCaptureBuffer_Unlock(c->DSCaptureBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2);
|
|
}
|
|
else
|
|
{
|
|
return 0;
|
|
}
|
|
return filled;
|
|
}
|
|
snd_capture_driver_t DSOUND_Capture =
|
|
{
|
|
1,
|
|
SDRVNAME,
|
|
DSOUND_Capture_Enumerate,
|
|
DSOUND_Capture_Init,
|
|
DSOUND_Capture_Start,
|
|
DSOUND_Capture_Update,
|
|
DSOUND_Capture_Stop,
|
|
DSOUND_Capture_Shutdown
|
|
};
|
|
#endif
|