/* 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 #define FORCE_DEFINE_GUID(name, l, w1, w2, b1, b2, b3, b4, b5, b6, b7, b8) \ EXTERN_C const GUID DECLSPEC_SELECTANY name \ = { l, w1, w2, { b1, b2, b3, b4, b5, b6, b7, b8 } } // SDL fix, seems SDL builds complain about multiple definitions of those 2 #ifndef _SDL FORCE_DEFINE_GUID(IID_IDirectSound, 0x279AFA83, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60); FORCE_DEFINE_GUID(IID_IKsPropertySet, 0x31efac30, 0x515c, 0x11d0, 0xa9, 0xaa, 0x00, 0xaa, 0x00, 0x61, 0xbe, 0x93); #endif #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:*/ #ifndef WAVE_FORMAT_EXTENSIBLE # define WAVE_FORMAT_EXTENSIBLE (0xfffe) #endif /* 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.ival) //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.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, 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.ival) { 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 (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.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 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.ival) { 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