eukara/fteqw
1
0
Fork 0
forked from fte/fteqw
Code Pull requests Activity
fteqw/engine/client/snd_directx.c
Spoike 75fb5f5398 The different aspects of directx are now activated independently from each other, allowing greater custom build control. 
Tweeked win64 options so as not to conflict with 32bit builds.
Win64 builds now with NO_LIBRARIES added. Yes, you need to provide 64bit libraries yourself if you want to use them (dinput/dsound should be part of the directx sdk and are, strictly speaking, not re-distributable). See bothdefs.h for how to activate individual 64bit libs.

git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@3138 fc73d0e0-1445-4013-8a0c-d673dee63da5
2009-03-07 04:37:24 +00:00

1051 lines
29 KiB
C
Raw Blame History

/*
Copyright (C) 1996-1997 Id Software, Inc.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
See the GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#include "quakedef.h"
#include "winquake.h"
#include <dsound.h>
#define SND_ERROR 0
#define SND_LOADED 1
#define SND_NOMORE 2 //like error, but doesn't try the next card.
#ifdef AVAIL_DSOUND
#define iDirectSoundCreate(a,b,c) pDirectSoundCreate(a,b,c)
#define iDirectSoundEnumerate(a,b,c) pDirectSoundEnumerate(a,b)
HRESULT (WINAPI *pDirectSoundCreate)(GUID FAR *lpGUID, LPDIRECTSOUND FAR *lplpDS, IUnknown FAR *pUnkOuter);
#if defined(VOICECHAT) && !defined(__MINGW32__)
HRESULT (WINAPI *pDirectSoundCaptureCreate)(GUID FAR *lpGUID, LPDIRECTSOUNDCAPTURE FAR *lplpDS, IUnknown FAR *pUnkOuter);
#endif
HRESULT (WINAPI *pDirectSoundEnumerate)(LPDSENUMCALLBACKA lpCallback, LPVOID lpContext );
// 64K is > 1 second at 16-bit, 22050 Hz
#define WAV_BUFFERS 64
#define WAV_MASK 0x3F
#define WAV_BUFFER_SIZE 0x0400
#define SECONDARY_BUFFER_SIZE 0x10000
typedef struct {
LPDIRECTSOUND pDS;
LPDIRECTSOUNDBUFFER pDSBuf;
LPDIRECTSOUNDBUFFER pDSPBuf;
DWORD gSndBufSize;
DWORD mmstarttime;
#ifdef _IKsPropertySet_
LPKSPROPERTYSET EaxKsPropertiesSet;
#endif
} dshandle_t;
HINSTANCE hInstDS;
static void DSOUND_Restore(soundcardinfo_t *sc)
{
DWORD dwStatus;
dshandle_t *dh = sc->handle;
if (dh->pDSBuf->lpVtbl->GetStatus (dh->pDSBuf, &dwStatus) != DD_OK)
Con_Printf ("Couldn't get sound buffer status\n");
if (dwStatus & DSBSTATUS_BUFFERLOST)
dh->pDSBuf->lpVtbl->Restore (dh->pDSBuf);
if (!(dwStatus & DSBSTATUS_PLAYING))
dh->pDSBuf->lpVtbl->Play(dh->pDSBuf, 0, 0, DSBPLAY_LOOPING);
}
DWORD dwSize;
static void *DSOUND_Lock(soundcardinfo_t *sc)
{
void *ret;
int reps;
DWORD dwSize2=0;
DWORD *pbuf2;
HRESULT hresult;
dshandle_t *dh = sc->handle;
dwSize=0;
reps = 0;
while ((hresult = dh->pDSBuf->lpVtbl->Lock(dh->pDSBuf, 0, dh->gSndBufSize, (void**)&ret, &dwSize,
(void**)&pbuf2, &dwSize2, 0)) != DS_OK)
{
if (hresult != DSERR_BUFFERLOST)
{
Con_Printf ("S_TransferStereo16: DS::Lock Sound Buffer Failed\n");
return NULL;
}
if (++reps > 10000)
{
Con_Printf ("S_TransferStereo16: DS: couldn't restore buffer\n");
return NULL;
}
DSOUND_Restore(sc);
}
return ret;
}
//called when the mixer is done with it.
static void DSOUND_Unlock(soundcardinfo_t *sc, void *buffer)
{
dshandle_t *dh = sc->handle;
dh->pDSBuf->lpVtbl->Unlock(dh->pDSBuf, buffer, dwSize, NULL, 0);
}
/*
==================
FreeSound
==================
*/
//per device
static void DSOUND_Shutdown (soundcardinfo_t *sc)
{
dshandle_t *dh = sc->handle;
if (!dh)
return;
sc->handle = NULL;
#ifdef _IKsPropertySet_
if (dh->EaxKsPropertiesSet)
{
IKsPropertySet_Release(dh->EaxKsPropertiesSet);
}
#endif
if (dh->pDSBuf)
{
dh->pDSBuf->lpVtbl->Stop(dh->pDSBuf);
dh->pDSBuf->lpVtbl->Release(dh->pDSBuf);
}
// only release primary buffer if it's not also the mixing buffer we just released
if (dh->pDSPBuf && (dh->pDSBuf != dh->pDSPBuf))
{
dh->pDSPBuf->lpVtbl->Release(dh->pDSPBuf);
}
if (dh->pDS)
{
dh->pDS->lpVtbl->SetCooperativeLevel (dh->pDS, mainwindow, DSSCL_NORMAL);
dh->pDS->lpVtbl->Release(dh->pDS);
}
dh->pDS = NULL;
dh->pDSBuf = NULL;
dh->pDSPBuf = NULL;
#ifdef _IKsPropertySet_
dh->EaxKsPropertiesSet = NULL;
#endif
Z_Free(dh);
}
const char *dsndcard;
GUID FAR *dsndguid;
int dsnd_guids;
int aimedforguid;
static BOOL (CALLBACK DSEnumCallback)(GUID FAR *guid, LPCSTR str1, LPCSTR str2, LPVOID parm)
{
if (guid == NULL)
return TRUE;
if (aimedforguid == dsnd_guids)
{
dsndcard = str1;
dsndguid = guid;
}
dsnd_guids++;
return TRUE;
}
/*
Direct Sound.
These following defs should be moved to winquake.h somewhere.
We tell DS to use a different wave format. We do this to gain extra channels. >2
We still use the old stuff too, when we can for compatability.
EAX 2 is also supported.
This is a global state. Once applied, it's applied for other programs too.
We have to do a few special things to try to ensure support in all it's different versions.
*/
/* new formatTag:*/
# define WAVE_FORMAT_EXTENSIBLE (0xfffe)
/* Speaker Positions:*/
# define SPEAKER_FRONT_LEFT 0x1
# define SPEAKER_FRONT_RIGHT 0x2
# define SPEAKER_FRONT_CENTER 0x4
# define SPEAKER_LOW_FREQUENCY 0x8
# define SPEAKER_BACK_LEFT 0x10
# define SPEAKER_BACK_RIGHT 0x20
# define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
# define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
# define SPEAKER_BACK_CENTER 0x100
# define SPEAKER_SIDE_LEFT 0x200
# define SPEAKER_SIDE_RIGHT 0x400
# define SPEAKER_TOP_CENTER 0x800
# define SPEAKER_TOP_FRONT_LEFT 0x1000
# define SPEAKER_TOP_FRONT_CENTER 0x2000
# define SPEAKER_TOP_FRONT_RIGHT 0x4000
# define SPEAKER_TOP_BACK_LEFT 0x8000
# define SPEAKER_TOP_BACK_CENTER 0x10000
# define SPEAKER_TOP_BACK_RIGHT 0x20000
/* Bit mask locations reserved for future use*/
# define SPEAKER_RESERVED 0x7FFC0000
/* Used to specify that any possible permutation of speaker configurations*/
# define SPEAKER_ALL 0x80000000
/* DirectSound Speaker Config*/
# define KSAUDIO_SPEAKER_MONO (SPEAKER_FRONT_CENTER)
# define KSAUDIO_SPEAKER_STEREO (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)
# define KSAUDIO_SPEAKER_QUAD (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
# define KSAUDIO_SPEAKER_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER)
# define KSAUDIO_SPEAKER_5POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
# define KSAUDIO_SPEAKER_7POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)
typedef struct {
WAVEFORMATEX Format;
union {
WORD wValidBitsPerSample; /* bits of precision */
WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */
WORD wReserved; /* If neither applies, set to */
/* zero. */
} Samples;
DWORD dwChannelMask; /* which channels are */
/* present in stream */
GUID SubFormat;
} QWAVEFORMATEX;
const static GUID KSDATAFORMAT_SUBTYPE_PCM = {0x00000001,0x0000,0x0010,
{0x80,
0x00,
0x00,
0xaa,
0x00,
0x38,
0x9b,
0x71}};
#ifdef _IKsPropertySet_
const static GUID CLSID_EAXDIRECTSOUND = {0x4ff53b81, 0x1ce0, 0x11d3,
{0xaa, 0xb8, 0x0, 0xa0, 0xc9, 0x59, 0x49, 0xd5}};
const static GUID DSPROPSETID_EAX20_LISTENERPROPERTIES = {0x306a6a8, 0xb224, 0x11d2,
{0x99, 0xe5, 0x0, 0x0, 0xe8, 0xd8, 0xc7, 0x22}};
typedef struct _EAXLISTENERPROPERTIES
{
long lRoom; // room effect level at low frequencies
long lRoomHF; // room effect high-frequency level re. low frequency level
float flRoomRolloffFactor; // like DS3D flRolloffFactor but for room effect
float flDecayTime; // reverberation decay time at low frequencies
float flDecayHFRatio; // high-frequency to low-frequency decay time ratio
long lReflections; // early reflections level relative to room effect
float flReflectionsDelay; // initial reflection delay time
long lReverb; // late reverberation level relative to room effect
float flReverbDelay; // late reverberation delay time relative to initial reflection
unsigned long dwEnvironment; // sets all listener properties
float flEnvironmentSize; // environment size in meters
float flEnvironmentDiffusion; // environment diffusion
float flAirAbsorptionHF; // change in level per meter at 5 kHz
unsigned long dwFlags; // modifies the behavior of properties
} EAXLISTENERPROPERTIES, *LPEAXLISTENERPROPERTIES;
enum
{
EAX_ENVIRONMENT_GENERIC,
EAX_ENVIRONMENT_PADDEDCELL,
EAX_ENVIRONMENT_ROOM,
EAX_ENVIRONMENT_BATHROOM,
EAX_ENVIRONMENT_LIVINGROOM,
EAX_ENVIRONMENT_STONEROOM,
EAX_ENVIRONMENT_AUDITORIUM,
EAX_ENVIRONMENT_CONCERTHALL,
EAX_ENVIRONMENT_CAVE,
EAX_ENVIRONMENT_ARENA,
EAX_ENVIRONMENT_HANGAR,
EAX_ENVIRONMENT_CARPETEDHALLWAY,
EAX_ENVIRONMENT_HALLWAY,
EAX_ENVIRONMENT_STONECORRIDOR,
EAX_ENVIRONMENT_ALLEY,
EAX_ENVIRONMENT_FOREST,
EAX_ENVIRONMENT_CITY,
EAX_ENVIRONMENT_MOUNTAINS,
EAX_ENVIRONMENT_QUARRY,
EAX_ENVIRONMENT_PLAIN,
EAX_ENVIRONMENT_PARKINGLOT,
EAX_ENVIRONMENT_SEWERPIPE,
EAX_ENVIRONMENT_UNDERWATER,
EAX_ENVIRONMENT_DRUGGED,
EAX_ENVIRONMENT_DIZZY,
EAX_ENVIRONMENT_PSYCHOTIC,
EAX_ENVIRONMENT_COUNT
};
typedef enum
{
DSPROPERTY_EAXLISTENER_NONE,
DSPROPERTY_EAXLISTENER_ALLPARAMETERS,
DSPROPERTY_EAXLISTENER_ROOM,
DSPROPERTY_EAXLISTENER_ROOMHF,
DSPROPERTY_EAXLISTENER_ROOMROLLOFFFACTOR,
DSPROPERTY_EAXLISTENER_DECAYTIME,
DSPROPERTY_EAXLISTENER_DECAYHFRATIO,
DSPROPERTY_EAXLISTENER_REFLECTIONS,
DSPROPERTY_EAXLISTENER_REFLECTIONSDELAY,
DSPROPERTY_EAXLISTENER_REVERB,
DSPROPERTY_EAXLISTENER_REVERBDELAY,
DSPROPERTY_EAXLISTENER_ENVIRONMENT,
DSPROPERTY_EAXLISTENER_ENVIRONMENTSIZE,
DSPROPERTY_EAXLISTENER_ENVIRONMENTDIFFUSION,
DSPROPERTY_EAXLISTENER_AIRABSORPTIONHF,
DSPROPERTY_EAXLISTENER_FLAGS
} DSPROPERTY_EAX_LISTENERPROPERTY;
const static GUID DSPROPSETID_EAX20_BUFFERPROPERTIES ={
0x306a6a7,
0xb224,
0x11d2,
{0x99, 0xe5, 0x0, 0x0, 0xe8, 0xd8, 0xc7, 0x22}};
const static GUID CLSID_EAXDirectSound ={
0x4ff53b81,
0x1ce0,
0x11d3,
{0xaa, 0xb8, 0x0, 0xa0, 0xc9, 0x59, 0x49, 0xd5}};
typedef struct _EAXBUFFERPROPERTIES
{
long lDirect; // direct path level
long lDirectHF; // direct path level at high frequencies
long lRoom; // room effect level
long lRoomHF; // room effect level at high frequencies
float flRoomRolloffFactor; // like DS3D flRolloffFactor but for room effect
long lObstruction; // main obstruction control (attenuation at high frequencies)
float flObstructionLFRatio; // obstruction low-frequency level re. main control
long lOcclusion; // main occlusion control (attenuation at high frequencies)
float flOcclusionLFRatio; // occlusion low-frequency level re. main control
float flOcclusionRoomRatio; // occlusion room effect level re. main control
long lOutsideVolumeHF; // outside sound cone level at high frequencies
float flAirAbsorptionFactor; // multiplies DSPROPERTY_EAXLISTENER_AIRABSORPTIONHF
unsigned long dwFlags; // modifies the behavior of properties
} EAXBUFFERPROPERTIES, *LPEAXBUFFERPROPERTIES;
typedef enum
{
DSPROPERTY_EAXBUFFER_NONE,
DSPROPERTY_EAXBUFFER_ALLPARAMETERS,
DSPROPERTY_EAXBUFFER_DIRECT,
DSPROPERTY_EAXBUFFER_DIRECTHF,
DSPROPERTY_EAXBUFFER_ROOM,
DSPROPERTY_EAXBUFFER_ROOMHF,
DSPROPERTY_EAXBUFFER_ROOMROLLOFFFACTOR,
DSPROPERTY_EAXBUFFER_OBSTRUCTION,
DSPROPERTY_EAXBUFFER_OBSTRUCTIONLFRATIO,
DSPROPERTY_EAXBUFFER_OCCLUSION,
DSPROPERTY_EAXBUFFER_OCCLUSIONLFRATIO,
DSPROPERTY_EAXBUFFER_OCCLUSIONROOMRATIO,
DSPROPERTY_EAXBUFFER_OUTSIDEVOLUMEHF,
DSPROPERTY_EAXBUFFER_AIRABSORPTIONFACTOR,
DSPROPERTY_EAXBUFFER_FLAGS
} DSPROPERTY_EAX_BUFFERPROPERTY;
#endif
static void DSOUND_SetUnderWater(soundcardinfo_t *sc, qboolean underwater)
{
#ifdef _IKsPropertySet_
dshandle_t *dh = sc->handle;
//attempt at eax support.
//EAX is a global thing. Get it going in a game and your media player will be doing it too.
if (dh->EaxKsPropertiesSet) //only on ds cards.
{
EAXLISTENERPROPERTIES ListenerProperties = {0};
/* DWORD p;
IKsPropertySet_Get(dh->EaxKsPropertiesSet, &DSPROPSETID_EAX20_LISTENERPROPERTIES,
DSPROPERTY_EAXLISTENER_ALLPARAMETERS, 0, 0, &ListenerProperties,
sizeof(ListenerProperties), &p);
*/
if (underwater)
{
#if 1 //phycotic.
ListenerProperties.flEnvironmentSize = 2.8;
ListenerProperties.flEnvironmentDiffusion = 0.240;
ListenerProperties.lRoom = -374;
ListenerProperties.lRoomHF = -150;
ListenerProperties.flRoomRolloffFactor = 0;
ListenerProperties.flAirAbsorptionHF = -5;
ListenerProperties.lReflections = -10000;
ListenerProperties.flReflectionsDelay = 0.053;
ListenerProperties.lReverb = 625;
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 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)
{
}
/*
==================
SNDDMA_InitDirect
Direct-Sound support
==================
*/
int DSOUND_InitCard (soundcardinfo_t *sc, int cardnum)
{
extern cvar_t snd_eax, snd_inactive;
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;
if (COM_CheckParm("-wavonly"))
return SND_NOMORE;
memset (&format, 0, sizeof(format));
if (sc->sn.numchannels >= 6) //5.1 surround
{
format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
format.Format.cbSize = 22;
memcpy(&format.SubFormat, &KSDATAFORMAT_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, &KSDATAFORMAT_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 (!hInstDS)
{
hInstDS = LoadLibrary("dsound.dll");
if (hInstDS == NULL)
{
Con_SafePrintf ("Couldn't load dsound.dll\n");
return SND_ERROR;
}
pDirectSoundCreate = (void *)GetProcAddress(hInstDS,"DirectSoundCreate");
if (!pDirectSoundCreate)
{
Con_SafePrintf ("Couldn't get DS proc addr\n");
return SND_ERROR;
}
pDirectSoundEnumerate = (void *)GetProcAddress(hInstDS,"DirectSoundEnumerateA");
}
dsnd_guids=0;
dsndguid=NULL;
dsndcard="DirectSound";
if (pDirectSoundEnumerate)
pDirectSoundEnumerate(&DSEnumCallback, NULL);
if (!snd_usemultipledevices.value) //if only one device, ALWAYS use the default.
dsndguid=NULL;
aimedforguid++;
if (!dsndguid) //no more...
if (aimedforguid != 1) //not the first device.
return SND_NOMORE;
sc->handle = Z_Malloc(sizeof(dshandle_t));
dh = sc->handle;
//EAX attempt
#ifndef MINIMAL
#ifdef _IKsPropertySet_
dh->pDS = NULL;
if (snd_eax.value)
{
CoInitialize(NULL);
if (FAILED(CoCreateInstance( &CLSID_EAXDirectSound, NULL, CLSCTX_INPROC_SERVER, &IID_IDirectSound, (void **)&dh->pDS )))
dh->pDS=NULL;
else
IDirectSound_Initialize(dh->pDS, dsndguid);
}
if (!dh->pDS)
#endif
#endif
{
while ((hresult = iDirectSoundCreate(dsndguid, &dh->pDS, NULL)) != DS_OK)
{
if (hresult != DSERR_ALLOCATED)
{
Con_SafePrintf (": create failed\n");
return SND_ERROR;
}
// 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 SND_ERROR;
// }
}
}
Q_strncpyz(sc->name, dsndcard, sizeof(sc->name));
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 (sc);
return SND_ERROR;
}
if (DS_OK != dh->pDS->lpVtbl->SetCooperativeLevel (dh->pDS, mainwindow, DSSCL_EXCLUSIVE))
{
Con_SafePrintf ("Set coop level failed\n");
DSOUND_Shutdown (sc);
return SND_ERROR;
}
// 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
if (snd_inactive.value)
{
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.value)
{
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 (sc);
return SND_ERROR;
}
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 (sc);
return SND_ERROR;
}
// 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 (sc);
return SND_ERROR;
}
if (DS_OK != dh->pDSPBuf->lpVtbl->GetCaps (dh->pDSPBuf, &dsbcaps))
{
Con_Printf ("DS:GetCaps failed\n");
DSOUND_Shutdown (sc);
return SND_ERROR;
}
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 (sc);
return SND_ERROR;
}
if (++reps > 10000)
{
Con_SafePrintf ("SNDDMA_InitDirect: DS: couldn't restore buffer\n");
DSOUND_Shutdown (sc);
return SND_ERROR;
}
}
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->SetWaterDistortion = DSOUND_SetUnderWater;
sc->Submit = DSOUND_Submit;
sc->Shutdown = DSOUND_Shutdown;
sc->GetDMAPos = DSOUND_GetDMAPos;
sc->Restore = DSOUND_Restore;
#ifdef _IKsPropertySet_
//attempt at eax support
if (snd_eax.value)
{
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 SND_LOADED;//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
return SND_LOADED;
}
int (*pDSOUND_InitCard) (soundcardinfo_t *sc, int cardnum) = &DSOUND_InitCard;
#endif
#if defined(VOICECHAT) && defined(AVAIL_DSOUND) && !defined(__MINGW32__)
LPDIRECTSOUNDCAPTURE DSCapture;
LPDIRECTSOUNDCAPTUREBUFFER DSCaptureBuffer;
long lastreadpos;
long bufferbytes = 1024*1024;
long inputwidth = 2;
static WAVEFORMATEX wfxFormat;
int SNDDMA_InitCapture (void)
{
DSCBUFFERDESC bufdesc;
wfxFormat.wFormatTag = WAVE_FORMAT_PCM;
wfxFormat.nChannels = 1;
wfxFormat.nSamplesPerSec = 11025;
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 (DSCaptureBuffer)
{
IDirectSoundCaptureBuffer_Stop(DSCaptureBuffer);
IDirectSoundCaptureBuffer_Release(DSCaptureBuffer);
DSCaptureBuffer=NULL;
}
if (DSCapture)
{
IDirectSoundCapture_Release(DSCapture);
DSCapture=NULL;
}
if (!hInstDS)
{
hInstDS = LoadLibrary("dsound.dll");
if (hInstDS == NULL)
{
Con_SafePrintf ("Couldn't load dsound.dll\n");
return SIS_FAILURE;
}
}
if (!pDirectSoundCaptureCreate)
{
pDirectSoundCaptureCreate = (void *)GetProcAddress(hInstDS,"DirectSoundCaptureCreate");
if (!pDirectSoundCaptureCreate)
{
Con_SafePrintf ("Couldn't get DS proc addr\n");
return SIS_FAILURE;
}
// pDirectSoundCaptureEnumerate = (void *)GetProcAddress(hInstDS,"DirectSoundCaptureEnumerateA");
}
pDirectSoundCaptureCreate(NULL, &DSCapture, NULL);
if (FAILED(IDirectSoundCapture_CreateCaptureBuffer(DSCapture, &bufdesc, &DSCaptureBuffer, NULL)))
{
Con_SafePrintf ("Couldn't create a capture buffer\n");
IDirectSoundCapture_Release(DSCapture);
DSCapture=NULL;
return SIS_FAILURE;
}
IDirectSoundCaptureBuffer_Start(DSCaptureBuffer, DSBPLAY_LOOPING);
lastreadpos = 0;
return SIS_SUCCESS;
}
void SNDVC_Submit(qbyte *buffer, int samples, int freq, int width);
void DSOUND_UpdateCapture(void)
{
HRESULT hr;
LPBYTE lpbuf1 = NULL;
LPBYTE lpbuf2 = NULL;
DWORD dwsize1 = 0;
DWORD dwsize2 = 0;
DWORD capturePos;
DWORD readPos;
long filled;
static int update;
char *pBuffer;
// return;
if (!snd_capture.value)
{
if (DSCaptureBuffer)
{
IDirectSoundCaptureBuffer_Stop(DSCaptureBuffer);
IDirectSoundCaptureBuffer_Release(DSCaptureBuffer);
DSCaptureBuffer=NULL;
}
if (DSCapture)
{
IDirectSoundCapture_Release(DSCapture);
DSCapture=NULL;
}
return;
}
else if (!DSCaptureBuffer)
{
SNDDMA_InitCapture();
return;
}
// Query to see how much data is in buffer.
hr = IDirectSoundCaptureBuffer_GetCurrentPosition( DSCaptureBuffer, &capturePos, &readPos );
if( hr != DS_OK )
{
return;
}
filled = readPos - lastreadpos;
if( filled < 0 ) filled += bufferbytes; // unwrap offset
if (filled > 1400) //figure out how much we need to empty it by, and if that's enough to be worthwhile.
filled = 1400;
else if (filled < 1400)
return;
if ((filled/inputwidth) & 1) //force even numbers of samples
filled -= inputwidth;
pBuffer = BZ_Malloc(filled*inputwidth);
// Lock free space in the DS
hr = IDirectSoundCaptureBuffer_Lock ( DSCaptureBuffer, lastreadpos, filled, (void **) &lpbuf1, &dwsize1,
(void **) &lpbuf2, &dwsize2, 0);
if (hr == DS_OK)
{
// Copy from DS to the buffer
memcpy( pBuffer, lpbuf1, dwsize1);
if(lpbuf2 != NULL)
{
memcpy( pBuffer+dwsize1, lpbuf2, dwsize2);
}
// Update our buffer offset and unlock sound buffer
lastreadpos = (lastreadpos + dwsize1 + dwsize2) % bufferbytes;
IDirectSoundCaptureBuffer_Unlock ( DSCaptureBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2);
}
else
{
BZ_Free(pBuffer);
return;
}
SNDVC_MicInput(pBuffer, filled, wfxFormat.nSamplesPerSec, inputwidth);
BZ_Free(pBuffer);
}
void (*pDSOUND_UpdateCapture) (void) = &DSOUND_UpdateCapture;
#endif
View git blame Copy permalink