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fteqw/engine/client/snd_win.c
Spoike 1755c401ea getting bored. Bulk commit of 6 files 
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@465 fc73d0e0-1445-4013-8a0c-d673dee63da5
2004-11-17 18:10:46 +00:00

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/*
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"
#ifndef NODIRECTX
#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);
HRESULT (WINAPI *pDirectSoundCaptureCreate)(GUID FAR *lpGUID, LPDIRECTSOUNDCAPTURE FAR *lplpDS, IUnknown FAR *pUnkOuter);
HRESULT (WINAPI *pDirectSoundEnumerate)(LPDSENUMCALLBACKA lpCallback, LPVOID lpContext );
#endif
// 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 enum {SIS_SUCCESS, SIS_FAILURE, SIS_NOTAVAIL, SIS_NOMORE} sndinitstat;
static qboolean wavonly;
//static qboolean dsound_init;
//static qboolean wav_init;
qboolean snd_firsttime = true;
static qboolean primary_format_set;
static int sample16;
//static int snd_sent, snd_completed;
/*
* Global variables. Must be visible to window-procedure function
* so it can unlock and free the data block after it has been played.
*/
/*
HANDLE hData;
HPSTR lpData, lpData2;
HGLOBAL hWaveHdr;
LPWAVEHDR lpWaveHdr;
HWAVEOUT hWaveOut;
WAVEOUTCAPS wavecaps;
DWORD gSndBufSize;
MMTIME mmstarttime;
LPDIRECTSOUND pDS;
LPDIRECTSOUNDBUFFER pDSBuf, pDSPBuf;
*/
HINSTANCE hInstDS;
qboolean SNDDMA_InitDirect (soundcardinfo_t *sc);
qboolean SNDDMA_InitWav (soundcardinfo_t *sc);
soundcardinfo_t *sndcardinfo;
/*
==================
S_BlockSound
==================
*/
//all devices
void S_BlockSound (void)
{
soundcardinfo_t *sc;
snd_blocked++;
for (sc = sndcardinfo; sc; sc=sc->next)
{
if (sc->snd_iswave && !sc->inactive_sound)
{
if (snd_blocked == 1)
waveOutReset (sc->hWaveOut);
}
}
}
/*
==================
S_UnblockSound
==================
*/
//all devices
void S_UnblockSound (void)
{
snd_blocked--;
}
/*
==================
FreeSound
==================
*/
//per device
void FreeSound (soundcardinfo_t *sc)
{
int i;
#ifndef NODIRECTX
if (sc->EaxKsPropertiesSet)
{
IKsPropertySet_Release(sc->EaxKsPropertiesSet);
}
if (sc->pDSBuf)
{
sc->pDSBuf->lpVtbl->Stop(sc->pDSBuf);
sc->pDSBuf->lpVtbl->Release(sc->pDSBuf);
}
// only release primary buffer if it's not also the mixing buffer we just released
if (sc->pDSPBuf && (sc->pDSBuf != sc->pDSPBuf))
{
sc->pDSPBuf->lpVtbl->Release(sc->pDSPBuf);
}
if (sc->pDS)
{
sc->pDS->lpVtbl->SetCooperativeLevel (sc->pDS, mainwindow, DSSCL_NORMAL);
sc->pDS->lpVtbl->Release(sc->pDS);
}
#endif
if (sc->hWaveOut)
{
waveOutReset (sc->hWaveOut);
if (sc->lpWaveHdr)
{
for (i=0 ; i< WAV_BUFFERS ; i++)
waveOutUnprepareHeader (sc->hWaveOut, sc->lpWaveHdr+i, sizeof(WAVEHDR));
}
waveOutClose (sc->hWaveOut);
if (sc->hWaveHdr)
{
GlobalUnlock(sc->hWaveHdr);
GlobalFree(sc->hWaveHdr);
}
if (sc->hData)
{
GlobalUnlock(sc->hData);
GlobalFree(sc->hData);
}
}
#ifndef NODIRECTX
sc->pDS = NULL;
sc->pDSBuf = NULL;
sc->pDSPBuf = NULL;
sc->EaxKsPropertiesSet = NULL;
sc->dsound_init = false;
#endif
sc->hWaveOut = 0;
sc->hData = 0;
sc->hWaveHdr = 0;
sc->lpData = NULL;
sc->lpWaveHdr = NULL;
sc->wav_init = false;
}
#ifndef NODIRECTX
const char *dsndcard;
GUID FAR *dsndguid;
int dsnd_guids;
int aimedforguid;
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
/*
==================
SNDDMA_InitDirect
Direct-Sound support
==================
*/
sndinitstat SNDDMA_InitDirect (soundcardinfo_t *sc)
{
extern cvar_t snd_khz, snd_eax, snd_speakers, snd_inactive;
DSBUFFERDESC dsbuf;
DSBCAPS dsbcaps;
DWORD dwSize, dwWrite;
DSCAPS dscaps;
QWAVEFORMATEX format, pformat;
HRESULT hresult;
int reps;
memset ((void *)&sc->sn, 0, sizeof (sc->sn));
sc->sn.numchannels = 2;
sc->sn.samplebits = 16;
if (!sc->sn.speed)
{
if (snd_khz.value >= 45)
sc->sn.speed = 48000;
else if (snd_khz.value >= 30) //set by a slider
sc->sn.speed = 44100;
else if (snd_khz.value >= 20)
sc->sn.speed = 22050;
else
sc->sn.speed = 11025;
}
memset (&format, 0, sizeof(format));
if (snd_speakers.value >= 5) //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 (snd_speakers.value >= 3) //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 (snd_speakers.value >= 1.5) //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 SIS_FAILURE;
}
pDirectSoundCreate = (void *)GetProcAddress(hInstDS,"DirectSoundCreate");
if (!pDirectSoundCreate)
{
Con_SafePrintf ("Couldn't get DS proc addr\n");
return SIS_FAILURE;
}
pDirectSoundEnumerate = (void *)GetProcAddress(hInstDS,"DirectSoundEnumerateA");
}
dsnd_guids=0;
dsndguid=NULL;
dsndcard="DirectSound";
if (snd_multipledevices)
if (pDirectSoundEnumerate)
pDirectSoundEnumerate(&DSEnumCallback, NULL);
aimedforguid++;
if (!dsndguid) //no more...
if (aimedforguid != 1) //not the first device.
return SIS_NOMORE;
//EAX attempt
#ifndef MINIMAL
CoInitialize(NULL);
if (FAILED(CoCreateInstance( &CLSID_EAXDirectSound, NULL, CLSCTX_INPROC_SERVER, &IID_IDirectSound, (void **)&sc->pDS )))
sc->pDS=NULL;
else
IDirectSound_Initialize(sc->pDS, dsndguid);
if (!sc->pDS)
#endif
{
while ((hresult = iDirectSoundCreate(dsndguid, &sc->pDS, NULL)) != DS_OK)
{
if (hresult != DSERR_ALLOCATED)
{
Con_SafePrintf (": create failed\n");
return SIS_FAILURE;
}
// 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 app then use snd_restart\n");
return SIS_NOTAVAIL;
// }
}
}
Q_strncpyz(sc->name, dsndcard, sizeof(sc->name));
dscaps.dwSize = sizeof(dscaps);
if (DS_OK != sc->pDS->lpVtbl->GetCaps (sc->pDS, &dscaps))
{
Con_SafePrintf ("Couldn't get DS caps\n");
}
if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
{
Con_SafePrintf ("No DirectSound driver installed\n");
FreeSound (sc);
return SIS_FAILURE;
}
if (DS_OK != sc->pDS->lpVtbl->SetCooperativeLevel (sc->pDS, mainwindow, DSSCL_EXCLUSIVE))
{
Con_SafePrintf ("Set coop level failed\n");
FreeSound (sc);
return SIS_FAILURE;
}
// 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;
memset(&dsbcaps, 0, sizeof(dsbcaps));
dsbcaps.dwSize = sizeof(dsbcaps);
primary_format_set = false;
if (!COM_CheckParm ("-snoforceformat"))
{
if (DS_OK == sc->pDS->lpVtbl->CreateSoundBuffer(sc->pDS, &dsbuf, &sc->pDSPBuf, NULL))
{
pformat = format;
if (DS_OK != sc->pDSPBuf->lpVtbl->SetFormat (sc->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; //dmw 29 may, 2003 removed locsoftware
if (snd_inactive.value)
{
dsbuf.dwFlags |= DSBCAPS_GLOBALFOCUS;
sc->inactive_sound = true;
}
dsbuf.dwBufferBytes = SECONDARY_BUFFER_SIZE;
dsbuf.lpwfxFormat = (WAVEFORMATEX *)&format;
memset(&dsbcaps, 0, sizeof(dsbcaps));
dsbcaps.dwSize = sizeof(dsbcaps);
if (DS_OK != sc->pDS->lpVtbl->CreateSoundBuffer(sc->pDS, &dsbuf, &sc->pDSBuf, NULL))
{
Con_SafePrintf ("DS:CreateSoundBuffer Failed");
FreeSound (sc);
return SIS_FAILURE;
}
sc->sn.numchannels = format.Format.nChannels;
sc->sn.samplebits = format.Format.wBitsPerSample;
sc->sn.speed = format.Format.nSamplesPerSec;
if (DS_OK != sc->pDSBuf->lpVtbl->GetCaps (sc->pDSBuf, &dsbcaps))
{
Con_SafePrintf ("DS:GetCaps failed\n");
FreeSound (sc);
return SIS_FAILURE;
}
// if (snd_firsttime)
// Con_SafePrintf ("Using secondary sound buffer\n");
}
else
{
if (DS_OK != sc->pDS->lpVtbl->SetCooperativeLevel (sc->pDS, mainwindow, DSSCL_WRITEPRIMARY))
{
Con_SafePrintf ("Set coop level failed\n");
FreeSound (sc);
return SIS_FAILURE;
}
if (DS_OK != sc->pDSPBuf->lpVtbl->GetCaps (sc->pDSPBuf, &dsbcaps))
{
Con_Printf ("DS:GetCaps failed\n");
return SIS_FAILURE;
}
sc->pDSBuf = sc->pDSPBuf;
// Con_SafePrintf ("Using primary sound buffer\n");
}
sc->gSndBufSize = dsbcaps.dwBufferBytes;
#if 1
// Make sure mixer is active
sc->pDSBuf->lpVtbl->Play(sc->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 = sc->pDSBuf->lpVtbl->Lock(sc->pDSBuf, 0, sc->gSndBufSize, (void**)&sc->lpData, &dwSize, NULL, NULL, 0)) != DS_OK)
{
if (hresult != DSERR_BUFFERLOST)
{
Con_SafePrintf ("SNDDMA_InitDirect: DS::Lock Sound Buffer Failed\n");
FreeSound (sc);
return SIS_FAILURE;
}
if (++reps > 10000)
{
Con_SafePrintf ("SNDDMA_InitDirect: DS: couldn't restore buffer\n");
FreeSound (sc);
return SIS_FAILURE;
}
}
memset(sc->lpData, 0, dwSize);
// lpData[4] = lpData[5] = 0x7f; // force a pop for debugging
// Sleep(500);
sc->pDSBuf->lpVtbl->Unlock(sc->pDSBuf, sc->lpData, dwSize, NULL, 0);
/* we don't want anyone to access the buffer directly w/o locking it first. */
sc->lpData = NULL;
sc->pDSBuf->lpVtbl->Stop(sc->pDSBuf);
#endif
sc->pDSBuf->lpVtbl->GetCurrentPosition(sc->pDSBuf, &sc->mmstarttime, &dwWrite);
sc->pDSBuf->lpVtbl->Play(sc->pDSBuf, 0, 0, DSBPLAY_LOOPING);
sc->sn.soundalive = true;
sc->sn.splitbuffer = false;
sc->sn.samples = sc->gSndBufSize/(sc->sn.samplebits/8);
sc->sn.samplepos = 0;
sc->sn.submission_chunk = 1;
sc->sn.buffer = (unsigned char *) sc->lpData;
sample16 = (sc->sn.samplebits/8) - 1;
sc->dsound_init = true;
#ifdef _IKsPropertySet_
//attempt at eax support
if (snd_eax.value)
{
int r;
DWORD support;
if (SUCCEEDED(IDirectSoundBuffer_QueryInterface(sc->pDSBuf, &IID_IKsPropertySet, (void*)&sc->EaxKsPropertiesSet)))
{
r = IKsPropertySet_QuerySupport(sc->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(sc->EaxKsPropertiesSet);
sc->EaxKsPropertiesSet = NULL;
Con_SafePrintf ("EAX 2 not supported\n");
return SIS_SUCCESS;//otherwise successful. It can be used for normal sound anyway.
}
//worked. EAX is supported.
}
else
{
Con_SafePrintf ("Couldn't get extended properties\n");
sc->EaxKsPropertiesSet = NULL;
}
}
#endif
return SIS_SUCCESS;
}
#endif
void SNDDMA_SetUnderWater(qboolean underwater)
{
#ifndef NODIRECTX
soundcardinfo_t *sc;
#ifdef _IKsPropertySet_
//attempt at eax support.
//EAX is a global thing. Get it going in a game and your media player will be doing it too.
for (sc = sndcardinfo; sc; sc = sc->next)
if (sc->EaxKsPropertiesSet) //only on ds cards.
{
EAXLISTENERPROPERTIES ListenerProperties = {0};
/* DWORD p;
IKsPropertySet_Get(sc->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(sc->EaxKsPropertiesSet, &DSPROPSETID_EAX20_LISTENERPROPERTIES,
DSPROPERTY_EAXLISTENER_ALLPARAMETERS, 0, 0, &ListenerProperties,
sizeof(ListenerProperties))))
Con_SafePrintf ("EAX set failed\n");
}
#endif
#endif
}
/*
==================
SNDDM_InitWav
Crappy windows multimedia base
==================
*/
qboolean SNDDMA_InitWav (soundcardinfo_t *sc)
{
extern cvar_t snd_khz;
WAVEFORMATEX format;
int i;
HRESULT hr;
sc->snd_sent = 0;
sc->snd_completed = 0;
sc->sn.numchannels = 2;
sc->sn.samplebits = 16;
if (!sc->sn.speed)
{
if (snd_khz.value == 48)
sc->sn.speed = 48000;
else if (snd_khz.value == 44 || snd_khz.value == 44.1)
sc->sn.speed = 44100;
else if (snd_khz.value == 22 || snd_khz.value == 22.05)
sc->sn.speed = 22050;
else
sc->sn.speed = 11025;
}
memset (&format, 0, sizeof(format));
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = sc->sn.numchannels;
format.wBitsPerSample = sc->sn.samplebits;
format.nSamplesPerSec = sc->sn.speed;
format.nBlockAlign = format.nChannels
*format.wBitsPerSample / 8;
format.cbSize = 0;
format.nAvgBytesPerSec = format.nSamplesPerSec
*format.nBlockAlign;
/* Open a waveform device for output using window callback. */
while ((hr = waveOutOpen((LPHWAVEOUT)&sc->hWaveOut, WAVE_MAPPER,
&format,
0, 0L, CALLBACK_NULL)) != MMSYSERR_NOERROR)
{
if (hr != MMSYSERR_ALLOCATED)
{
Con_SafePrintf ("waveOutOpen 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 ("waveOutOpen failure;\n"
" hardware already in use\nclose the app, then try using snd_restart\n");
return false;
// }
}
/*
* Allocate and lock memory for the waveform data. The memory
* for waveform data must be globally allocated with
* GMEM_MOVEABLE and GMEM_SHARE flags.
*/
sc->gSndBufSize = WAV_BUFFERS*WAV_BUFFER_SIZE;
sc->hData = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, sc->gSndBufSize);
if (!sc->hData)
{
Con_SafePrintf ("Sound: Out of memory.\n");
FreeSound (sc);
return false;
}
sc->lpData = GlobalLock(sc->hData);
if (!sc->lpData)
{
Con_SafePrintf ("Sound: Failed to lock.\n");
FreeSound (sc);
return false;
}
memset (sc->lpData, 0, sc->gSndBufSize);
/*
* Allocate and lock memory for the header. This memory must
* also be globally allocated with GMEM_MOVEABLE and
* GMEM_SHARE flags.
*/
sc->hWaveHdr = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE,
(DWORD) sizeof(WAVEHDR) * WAV_BUFFERS);
if (sc->hWaveHdr == NULL)
{
Con_SafePrintf ("Sound: Failed to Alloc header.\n");
FreeSound (sc);
return false;
}
sc->lpWaveHdr = (LPWAVEHDR) GlobalLock(sc->hWaveHdr);
if (sc->lpWaveHdr == NULL)
{
Con_SafePrintf ("Sound: Failed to lock header.\n");
FreeSound (sc);
return false;
}
memset (sc->lpWaveHdr, 0, sizeof(WAVEHDR) * WAV_BUFFERS);
/* After allocation, set up and prepare headers. */
for (i=0 ; i<WAV_BUFFERS ; i++)
{
sc->lpWaveHdr[i].dwBufferLength = WAV_BUFFER_SIZE;
sc->lpWaveHdr[i].lpData = sc->lpData + i*WAV_BUFFER_SIZE;
if (waveOutPrepareHeader(sc->hWaveOut, sc->lpWaveHdr+i, sizeof(WAVEHDR)) !=
MMSYSERR_NOERROR)
{
Con_SafePrintf ("Sound: failed to prepare wave headers\n");
FreeSound (sc);
return false;
}
}
sc->sn.soundalive = true;
sc->sn.splitbuffer = false;
sc->sn.samples = sc->gSndBufSize/(sc->sn.samplebits/8);
sc->sn.samplepos = 0;
sc->sn.submission_chunk = 1;
sc->sn.buffer = (unsigned char *) sc->lpData;
sample16 = (sc->sn.samplebits/8) - 1;
sc->wav_init = true;
return true;
}
/*
==================
SNDDMA_Init
Try to find a sound device to mix for.
Returns false if nothing is found.
==================
*/
int SNDDMA_Init(soundcardinfo_t *sc)
{
sndinitstat stat;
if (COM_CheckParm ("-wavonly"))
wavonly = true;
sc->inactive_sound = false; //don't generate sound when the window is inactive.
#ifndef NODIRECTX
sc->dsound_init =
#endif
sc->wav_init = 0;
stat = SIS_FAILURE; // assume DirectSound won't initialize
#ifndef NODIRECTX
/* Init DirectSound */
if (!wavonly)
{
if (snd_firsttime || sc->snd_isdirect)
{
stat = SNDDMA_InitDirect (sc);
if (stat == SIS_SUCCESS)
{
sc->snd_isdirect = true;
if (snd_firsttime)
Con_SafePrintf ("%s initialized\n", sc->name);
return 1;
}
else if (stat == SIS_NOMORE)
return 2;
else
{
sc->snd_isdirect = false;
Con_SafePrintf ("DirectSound failed to init\n");
}
}
}
#endif
// if DirectSound didn't succeed in initializing, try to initialize
// waveOut sound, unless DirectSound failed because the hardware is
// already allocated (in which case the user has already chosen not
// to have sound)
#ifndef NODIRECTX
if (!sc->dsound_init && (stat != SIS_NOTAVAIL))
#endif
{
if (snd_firsttime || sc->snd_iswave)
{
sc->snd_iswave = SNDDMA_InitWav (sc);
if (sc->snd_iswave)
{
if (snd_firsttime)
Con_SafePrintf ("Wave sound initialized\n");
}
else
{
Con_SafePrintf ("Wave sound failed to init\n");
}
}
}
snd_firsttime = false;
#ifndef NODIRECTX
if (!sc->dsound_init && !sc->wav_init)
#endif
{
if (snd_firsttime)
Con_SafePrintf ("No sound device initialized\n");
return 2;
}
return 1;
}
/*
==============
SNDDMA_GetDMAPos
return the current sample position (in mono samples read)
inside the recirculating dma buffer, so the mixing code will know
how many sample are required to fill it up.
===============
*/
int SNDDMA_GetDMAPos(soundcardinfo_t *sc)
{
DWORD mmtime;
int s;
DWORD dwWrite;
#ifndef NODIRECTX
if (sc->dsound_init)
{
sc->pDSBuf->lpVtbl->GetCurrentPosition(sc->pDSBuf, &mmtime, &dwWrite);
s = mmtime - sc->mmstarttime;
}
else
#endif
if (sc->wav_init)
{
s = sc->snd_sent * WAV_BUFFER_SIZE;
}
else
s = 0;
s >>= sample16;
// s = (s/shm->numchannels % (shm->samples-1))*shm->numchannels;
return s;
}
/*
==============
SNDDMA_Submit
Send sound to device if buffer isn't really the dma buffer
===============
*/
void SNDDMA_Submit(soundcardinfo_t *sc)
{
LPWAVEHDR h;
int wResult;
if (!sc->wav_init)
return;
//
// find which sound blocks have completed
//
while (1)
{
if ( sc->snd_completed == sc->snd_sent )
{
Con_DPrintf ("Sound overrun\n");
break;
}
if ( ! (sc->lpWaveHdr[ sc->snd_completed & WAV_MASK].dwFlags & WHDR_DONE) )
{
break;
}
sc->snd_completed++; // this buffer has been played
}
//
// submit two new sound blocks
//
while (((sc->snd_sent - sc->snd_completed) >> sample16) < 4)
{
h = sc->lpWaveHdr + ( sc->snd_sent&WAV_MASK );
sc->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(sc->hWaveOut, h, sizeof(WAVEHDR));
if (wResult != MMSYSERR_NOERROR)
{
Con_SafePrintf ("Failed to write block to device\n");
FreeSound (sc);
return;
}
}
}
/*
==============
SNDDMA_Shutdown
Reset the sound device for exiting
===============
*/
void SNDDMA_Shutdown(soundcardinfo_t *sc)
{
FreeSound (sc);
}
#if !defined(NODIRECTX) && defined(VOICECHAT)
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 (!pDirectSoundCreate)
{
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 S_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);
}
#else
void S_UpdateCapture(void)
{
}
#endif
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