quakeforge-old/common/snd_win.c
2000-01-06 13:48:07 +00:00

753 lines
16 KiB
C

/*
Copyright (C) 1996-1997 Id Software, Inc.
Copyright (C) 1999,2000 contributors of the QuakeForge project
Please see the file "AUTHORS" for a list of contributors
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"
#ifdef HAVE_MMSYSTEM_H
# include <mmsystem.h>
#endif
#ifdef HAVE_DSOUND
#define iDirectSoundCreate(a,b,c) pDirectSoundCreate(a,b,c)
HRESULT (WINAPI *pDirectSoundCreate)(GUID FAR *lpGUID, LPDIRECTSOUND FAR *lplpDS, IUnknown FAR *pUnkOuter);
#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} sndinitstat;
static qboolean wavonly;
static qboolean dsound_init;
static qboolean wav_init;
static qboolean snd_firsttime = true, snd_iswave;
#ifdef HAVE_DSOUND
static qboolean primary_format_set, snd_isdirect;
#endif
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;
#ifdef HAVE_DSOUND
LPDIRECTSOUND pDS;
LPDIRECTSOUNDBUFFER pDSBuf, pDSPBuf;
#endif
HINSTANCE hInstDS;
qboolean SNDDMA_InitDirect (void);
qboolean SNDDMA_InitWav (void);
/*
==================
S_BlockSound
==================
*/
void S_BlockSound (void)
{
// DirectSound takes care of blocking itself
if (snd_iswave)
{
snd_blocked++;
if (snd_blocked == 1)
waveOutReset (hWaveOut);
}
}
/*
==================
S_UnblockSound
==================
*/
void S_UnblockSound (void)
{
// DirectSound takes care of blocking itself
if (snd_iswave)
{
snd_blocked--;
}
}
/*
==================
FreeSound
==================
*/
void FreeSound (void)
{
int i;
#ifdef HAVE_DSOUND
if (pDSBuf)
{
pDSBuf->lpVtbl->Stop(pDSBuf);
pDSBuf->lpVtbl->Release(pDSBuf);
}
// only release primary buffer if it's not also the mixing buffer we just released
if (pDSPBuf && (pDSBuf != pDSPBuf))
{
pDSPBuf->lpVtbl->Release(pDSPBuf);
}
if (pDS)
{
pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_NORMAL);
pDS->lpVtbl->Release(pDS);
}
#endif
if (hWaveOut)
{
waveOutReset (hWaveOut);
if (lpWaveHdr)
{
for (i=0 ; i< WAV_BUFFERS ; i++)
waveOutUnprepareHeader (hWaveOut, lpWaveHdr+i, sizeof(WAVEHDR));
}
waveOutClose (hWaveOut);
if (hWaveHdr)
{
GlobalUnlock(hWaveHdr);
GlobalFree(hWaveHdr);
}
if (hData)
{
GlobalUnlock(hData);
GlobalFree(hData);
}
}
#ifdef HAVE_DSOUND
pDS = NULL;
pDSBuf = NULL;
pDSPBuf = NULL;
#endif
hWaveOut = 0;
hData = 0;
hWaveHdr = 0;
lpData = NULL;
lpWaveHdr = NULL;
dsound_init = false;
wav_init = false;
}
/*
==================
SNDDMA_InitDirect
Direct-Sound support
==================
*/
#ifdef HAVE_DSOUND
sndinitstat SNDDMA_InitDirect (void)
{
DSBUFFERDESC dsbuf;
DSBCAPS dsbcaps;
DWORD dwSize, dwWrite;
DSCAPS dscaps;
WAVEFORMATEX format, pformat;
HRESULT hresult;
int reps;
memset ((void *)&sn, 0, sizeof (sn));
shm = &sn;
shm->channels = 2;
shm->samplebits = 16;
shm->speed = 11025;
memset (&format, 0, sizeof(format));
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = shm->channels;
format.wBitsPerSample = shm->samplebits;
format.nSamplesPerSec = shm->speed;
format.nBlockAlign = format.nChannels
*format.wBitsPerSample / 8;
format.cbSize = 0;
format.nAvgBytesPerSec = format.nSamplesPerSec
*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;
}
}
while ((hresult = iDirectSoundCreate(NULL, &pDS, NULL)) != DS_OK)
{
if (hresult != DSERR_ALLOCATED)
{
Con_SafePrintf ("DirectSound 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 ("DirectSoundCreate failure\n"
" hardware already in use\n");
return SIS_NOTAVAIL;
}
}
dscaps.dwSize = sizeof(dscaps);
if (DS_OK != pDS->lpVtbl->GetCaps (pDS, &dscaps))
{
Con_SafePrintf ("Couldn't get DS caps\n");
}
if (dscaps.dwFlags & DSCAPS_EMULDRIVER)
{
Con_SafePrintf ("No DirectSound driver installed\n");
FreeSound ();
return SIS_FAILURE;
}
// Changed DSSCL_EXCLUSIVE to DSSCL_NORMAL
// -- Robert S. Elsner <sockman@ngfc.com>
if (DS_OK != pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_NORMAL))
// if (DS_OK != pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_EXCLUSIVE))
{
Con_SafePrintf ("Set coop level failed\n");
FreeSound ();
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;
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 == pDS->lpVtbl->CreateSoundBuffer(pDS, &dsbuf, &pDSPBuf, NULL))
{
pformat = format;
if (DS_OK != pDSPBuf->lpVtbl->SetFormat (pDSPBuf, &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;
dsbuf.dwBufferBytes = SECONDARY_BUFFER_SIZE;
dsbuf.lpwfxFormat = &format;
memset(&dsbcaps, 0, sizeof(dsbcaps));
dsbcaps.dwSize = sizeof(dsbcaps);
if (DS_OK != pDS->lpVtbl->CreateSoundBuffer(pDS, &dsbuf, &pDSBuf, NULL))
{
Con_SafePrintf ("DS:CreateSoundBuffer Failed");
FreeSound ();
return SIS_FAILURE;
}
shm->channels = format.nChannels;
shm->samplebits = format.wBitsPerSample;
shm->speed = format.nSamplesPerSec;
if (DS_OK != pDSBuf->lpVtbl->GetCaps (pDSBuf, &dsbcaps))
{
Con_SafePrintf ("DS:GetCaps failed\n");
FreeSound ();
return SIS_FAILURE;
}
// if (snd_firsttime)
// Con_SafePrintf ("Using secondary sound buffer\n");
}
else
{
// Removed the option for -primarysound
// -- Robert S. Elsner <sockman@ngfc.com>
#if 0
if (DS_OK != pDS->lpVtbl->SetCooperativeLevel (pDS, mainwindow, DSSCL_WRITEPRIMARY))
{
Con_SafePrintf ("Set coop level failed\n");
FreeSound ();
return SIS_FAILURE;
}
if (DS_OK != pDSPBuf->lpVtbl->GetCaps (pDSPBuf, &dsbcaps))
{
Con_Printf ("DS:GetCaps failed\n");
return SIS_FAILURE;
}
pDSBuf = pDSPBuf;
// Con_SafePrintf ("Using primary sound buffer\n");
#endif
}
// Make sure mixer is active
pDSBuf->lpVtbl->Play(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);*/
gSndBufSize = dsbcaps.dwBufferBytes;
// initialize the buffer
reps = 0;
while ((hresult = pDSBuf->lpVtbl->Lock(pDSBuf, 0, gSndBufSize, &lpData, &dwSize, NULL, NULL, 0)) != DS_OK)
{
if (hresult != DSERR_BUFFERLOST)
{
Con_SafePrintf ("SNDDMA_InitDirect: DS::Lock Sound Buffer Failed\n");
FreeSound ();
return SIS_FAILURE;
}
if (++reps > 10000)
{
Con_SafePrintf ("SNDDMA_InitDirect: DS: couldn't restore buffer\n");
FreeSound ();
return SIS_FAILURE;
}
}
memset(lpData, 0, dwSize);
// lpData[4] = lpData[5] = 0x7f; // force a pop for debugging
pDSBuf->lpVtbl->Unlock(pDSBuf, lpData, dwSize, NULL, 0);
/* we don't want anyone to access the buffer directly w/o locking it first. */
lpData = NULL;
pDSBuf->lpVtbl->Stop(pDSBuf);
pDSBuf->lpVtbl->GetCurrentPosition(pDSBuf, &mmstarttime.u.sample, &dwWrite);
pDSBuf->lpVtbl->Play(pDSBuf, 0, 0, DSBPLAY_LOOPING);
shm->soundalive = true;
shm->splitbuffer = false;
shm->samples = gSndBufSize/(shm->samplebits/8);
shm->samplepos = 0;
shm->submission_chunk = 1;
shm->buffer = (unsigned char *) lpData;
sample16 = (shm->samplebits/8) - 1;
dsound_init = true;
return SIS_SUCCESS;
}
#endif /* HAVE_DSOUND */
/*
==================
SNDDM_InitWav
Crappy windows multimedia base
==================
*/
qboolean SNDDMA_InitWav (void)
{
WAVEFORMATEX format;
int i;
HRESULT hr;
snd_sent = 0;
snd_completed = 0;
shm = &sn;
shm->channels = 2;
shm->samplebits = 16;
shm->speed = 11025;
memset (&format, 0, sizeof(format));
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = shm->channels;
format.wBitsPerSample = shm->samplebits;
format.nSamplesPerSec = shm->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)&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\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.
*/
gSndBufSize = WAV_BUFFERS*WAV_BUFFER_SIZE;
hData = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, gSndBufSize);
if (!hData)
{
Con_SafePrintf ("Sound: Out of memory.\n");
FreeSound ();
return false;
}
lpData = GlobalLock(hData);
if (!lpData)
{
Con_SafePrintf ("Sound: Failed to lock.\n");
FreeSound ();
return false;
}
memset (lpData, 0, gSndBufSize);
/*
* Allocate and lock memory for the header. This memory must
* also be globally allocated with GMEM_MOVEABLE and
* GMEM_SHARE flags.
*/
hWaveHdr = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE,
(DWORD) sizeof(WAVEHDR) * WAV_BUFFERS);
if (hWaveHdr == NULL)
{
Con_SafePrintf ("Sound: Failed to Alloc header.\n");
FreeSound ();
return false;
}
lpWaveHdr = (LPWAVEHDR) GlobalLock(hWaveHdr);
if (lpWaveHdr == NULL)
{
Con_SafePrintf ("Sound: Failed to lock header.\n");
FreeSound ();
return false;
}
memset (lpWaveHdr, 0, sizeof(WAVEHDR) * WAV_BUFFERS);
/* After allocation, set up and prepare headers. */
for (i=0 ; i<WAV_BUFFERS ; i++)
{
lpWaveHdr[i].dwBufferLength = WAV_BUFFER_SIZE;
lpWaveHdr[i].lpData = lpData + i*WAV_BUFFER_SIZE;
if (waveOutPrepareHeader(hWaveOut, lpWaveHdr+i, sizeof(WAVEHDR)) !=
MMSYSERR_NOERROR)
{
Con_SafePrintf ("Sound: failed to prepare wave headers\n");
FreeSound ();
return false;
}
}
shm->soundalive = true;
shm->splitbuffer = false;
shm->samples = gSndBufSize/(shm->samplebits/8);
shm->samplepos = 0;
shm->submission_chunk = 1;
shm->buffer = (unsigned char *) lpData;
sample16 = (shm->samplebits/8) - 1;
wav_init = true;
return true;
}
/*
==================
SNDDMA_Init
Try to find a sound device to mix for.
Returns false if nothing is found.
==================
*/
qboolean SNDDMA_Init(void)
{
sndinitstat stat;
if (COM_CheckParm ("-wavonly"))
wavonly = true;
dsound_init = wav_init = 0;
stat = SIS_FAILURE; // assume DirectSound won't initialize
#ifdef HAVE_DSOUND
/* Init DirectSound */
if (!wavonly)
{
if (snd_firsttime || snd_isdirect)
{
stat = SNDDMA_InitDirect ();;
if (stat == SIS_SUCCESS)
{
snd_isdirect = true;
if (snd_firsttime)
Con_SafePrintf ("DirectSound initialized\n");
}
else
{
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)
if (!dsound_init && (stat != SIS_NOTAVAIL))
{
if (snd_firsttime || snd_iswave)
{
snd_iswave = SNDDMA_InitWav ();
if (snd_iswave)
{
if (snd_firsttime)
Con_SafePrintf ("Wave sound initialized\n");
}
else
{
Con_SafePrintf ("Wave sound failed to init\n");
}
}
}
snd_firsttime = false;
if (!dsound_init && !wav_init)
{
if (snd_firsttime)
Con_SafePrintf ("No sound device initialized\n");
return 0;
}
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(void)
{
int s;
#ifdef HAVE_DSOUND
MMTIME mmtime;
DWORD dwWrite;
if (dsound_init)
{
mmtime.wType = TIME_SAMPLES;
pDSBuf->lpVtbl->GetCurrentPosition(pDSBuf, &mmtime.u.sample, &dwWrite);
s = mmtime.u.sample - mmstarttime.u.sample;
} else
#endif /* HAVE_DSOUND */
if (wav_init) {
s = snd_sent * WAV_BUFFER_SIZE;
}
s >>= sample16;
s &= (shm->samples-1);
return s;
}
/*
==============
SNDDMA_Submit
Send sound to device if buffer isn't really the dma buffer
===============
*/
void SNDDMA_Submit(void)
{
LPWAVEHDR h;
int wResult;
if (!wav_init)
return;
//
// find which sound blocks have completed
//
while (1)
{
if ( snd_completed == snd_sent )
{
Con_DPrintf ("Sound overrun\n");
break;
}
if ( ! (lpWaveHdr[ snd_completed & WAV_MASK].dwFlags & WHDR_DONE) )
{
break;
}
snd_completed++; // this buffer has been played
}
//
// submit two new sound blocks
//
while (((snd_sent - snd_completed) >> sample16) < 4)
{
h = lpWaveHdr + ( snd_sent&WAV_MASK );
snd_sent++;
/*
* Now the data block can be sent to the output device. The
* waveOutWrite function returns immediately and waveform
* data is sent to the output device in the background.
*/
wResult = waveOutWrite(hWaveOut, h, sizeof(WAVEHDR));
if (wResult != MMSYSERR_NOERROR)
{
Con_SafePrintf ("Failed to write block to device\n");
FreeSound ();
return;
}
}
}
/*
==============
SNDDMA_Shutdown
Reset the sound device for exiting
===============
*/
void SNDDMA_Shutdown(void)
{
FreeSound ();
}