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added native sound code snd_alsa.c, snd_oss.c, snd_win.c and for future

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reference and/or probable use.

git-svn-id: svn+ssh://svn.code.sf.net/p/quakespasm/code/trunk@434 af15c1b1-3010-417e-b628-4374ebc0bcbd
This commit is contained in:
sezero 2011-03-28 11:37:44 +00:00
parent a1d224402e
commit f3db05de11
3 changed files with 1419 additions and 0 deletions
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332
quakespasm/Quake/snd_alsa.c Normal file
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/*
snd_alsa.c
ALSA 1.0 sound driver for Linux Hexen II
Copyright (C) 1999,2004 contributors of the QuakeForge project
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:
Free Software Foundation, Inc.
59 Temple Place - Suite 330
Boston, MA 02111-1307, USA
*/
#include "quakedef.h"
#include <alsa/asoundlib.h>
#define NB_PERIODS 4
//static const char alsa_default[] = "hw:0,0";
//static const char alsa_default[] = "plughw:0";
static const char alsa_default[] = "default";
static const char *pcmname = alsa_default;
static snd_pcm_t *pcm = NULL;
static snd_pcm_uframes_t buffer_size;
static const int tryrates[] = { 11025, 22050, 44100, 48000, 16000, 24000, 8000 };
static const int MAX_TRYRATES = sizeof(tryrates)/sizeof(tryrates[0]);
#if defined(__GNUC__) && \
!(defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L)
#define ALSA_CHECK_ERR(check, fmt, args...) \
do { \
if (check < 0) { \
Con_Printf ("ALSA: " fmt, ##args); \
goto error; \
} \
} while (0)
#else
#define ALSA_CHECK_ERR(check, ...) \
do { \
if (check < 0) { \
Con_Printf ("ALSA: " __VA_ARGS__); \
goto error; \
} \
} while (0)
#endif
qboolean SNDDMA_Init (dma_t *dma)
{
int i, err;
unsigned int rate;
int tmp_bits, tmp_chan;
snd_pcm_hw_params_t *hw = NULL;
snd_pcm_sw_params_t *sw = NULL;
snd_pcm_uframes_t frag_size;
i = COM_CheckParm("-alsadev");
if (i != 0 && i < com_argc - 1)
pcmname = com_argv[i + 1];
err = snd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (err < 0)
{
Con_Printf ("ALSA: error opening device \"%s\": %s\n", pcmname, snd_strerror(err));
return false;
}
Con_Printf ("ALSA: Using device: %s\n", pcmname);
err = snd_pcm_hw_params_malloc (&hw);
ALSA_CHECK_ERR(err, "unable to allocate hardware params. %s\n", snd_strerror(err));
err = snd_pcm_hw_params_any (pcm, hw);
ALSA_CHECK_ERR(err, "unable to init hardware params. %s\n", snd_strerror(err));
err = snd_pcm_hw_params_set_access (pcm, hw, SND_PCM_ACCESS_MMAP_INTERLEAVED);
ALSA_CHECK_ERR(err, "unable to set interleaved access. %s\n", snd_strerror(err));
i = (loadas8bit.value) ? 8 : 16;
tmp_bits = (i == 8) ? SND_PCM_FORMAT_U8 : SND_PCM_FORMAT_S16;
err = snd_pcm_hw_params_set_format (pcm, hw, (snd_pcm_format_t) tmp_bits);
if (err < 0)
{
Con_Printf ("Problems setting %d bit format, trying alternatives..\n", i);
tmp_bits = (i == 8) ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U8;
err = snd_pcm_hw_params_set_format (pcm, hw, (snd_pcm_format_t) tmp_bits);
ALSA_CHECK_ERR(err, "Neither 8 nor 16 bit format supported. %s\n", snd_strerror(err));
}
tmp_bits = (tmp_bits == SND_PCM_FORMAT_U8) ? 8 : 16;
i = tmp_chan = (COM_CheckParm("-sndmono") == 0) ? 2 : 1;
err = snd_pcm_hw_params_set_channels (pcm, hw, tmp_chan);
if (err < 0)
{
Con_Printf ("Problems setting channels to %s, retrying for %s\n",
(i == 2) ? "stereo" : "mono",
(i == 2) ? "mono" : "stereo");
tmp_chan = (i == 2) ? 1 : 2;
err = snd_pcm_hw_params_set_channels (pcm, hw, tmp_chan);
ALSA_CHECK_ERR(err, "unable to set desired channels. %s\n", snd_strerror(err));
}
rate = (int)sndspeed.value;
err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
if (err < 0)
{
Con_Printf("Problems setting sample rate, trying alternatives..\n");
for (i = 0; i < MAX_TRYRATES; i++)
{
rate = tryrates[i];
err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
if (err < 0)
{
Con_DPrintf ("Unable to set sample rate %d\n", tryrates[i]);
rate = 0;
}
else
{
if (rate != tryrates[i])
{
Con_Printf ("Warning: Rate set (%u) didn't match requested rate (%d)!\n", rate, tryrates[i]);
// goto error;
}
break;
}
}
if (rate == 0)
{
Con_Printf ("Unable to set any sample rates.\n");
goto error;
}
}
else
{
if (rate != (int)sndspeed.value)
{
Con_Printf ("Warning: Rate set (%u) didn't match requested rate (%d)!\n", rate, (int)sndspeed.value);
// goto error;
}
}
/* pick a buffer size that is a power of 2 (by masking off low bits) */
buffer_size = i = (int)(rate * 0.15f);
while (buffer_size & (buffer_size-1))
buffer_size &= (buffer_size-1);
/* then check if it is the nearest power of 2 and bump it up if not */
if (i - buffer_size >= buffer_size >> 1)
buffer_size *= 2;
err = snd_pcm_hw_params_set_buffer_size_near (pcm, hw, &buffer_size);
ALSA_CHECK_ERR(err, "unable to set buffer size near %lu (%s)\n",
(unsigned long)buffer_size, snd_strerror(err));
err = snd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
ALSA_CHECK_ERR(err, "unable to get buffer size. %s\n", snd_strerror(err));
if (buffer_size & (buffer_size-1))
{
Con_Printf ("ALSA: WARNING: non-power of 2 buffer size. sound may be\n");
Con_Printf ("unsatisfactory. Recommend using either the plughw or hw\n");
Con_Printf ("devices or adjusting dmix to have a power of 2 buf size\n");
}
/* pick a period size near the buffer_size we got from ALSA */
frag_size = buffer_size / NB_PERIODS;
err = snd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0);
ALSA_CHECK_ERR(err, "unable to set period size near %i. %s\n",
(int)frag_size, snd_strerror(err));
err = snd_pcm_hw_params (pcm, hw);
ALSA_CHECK_ERR(err, "unable to install hardware params. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_malloc (&sw);
ALSA_CHECK_ERR(err, "unable to allocate software params. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_current (pcm, sw);
ALSA_CHECK_ERR(err, "unable to determine current software params. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
ALSA_CHECK_ERR(err, "unable to set playback threshold. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
ALSA_CHECK_ERR(err, "unable to set playback stop threshold. %s\n", snd_strerror(err));
err = snd_pcm_sw_params (pcm, sw);
ALSA_CHECK_ERR(err, "unable to install software params. %s\n", snd_strerror(err));
memset ((void *) dma, 0, sizeof(dma_t));
shm = dma;
shm->channels = tmp_chan;
/*
// don't mix less than this in mono samples:
err = snd_pcm_hw_params_get_period_size (hw,
(snd_pcm_uframes_t *) (char *) (&shm->submission_chunk), 0);
ALSA_CHECK_ERR(err, "unable to get period size. %s\n", snd_strerror(err));
*/
shm->submission_chunk = 1;
shm->samplepos = 0;
shm->samplebits = tmp_bits;
Con_Printf ("ALSA: %lu bytes buffer with mmap interleaved access\n", (unsigned long)buffer_size);
shm->samples = buffer_size * shm->channels; // mono samples in buffer
shm->speed = rate;
SNDDMA_GetDMAPos (); // sets shm->buffer
snd_pcm_hw_params_free(hw);
snd_pcm_sw_params_free(sw);
return true;
error:
// full clean-up
if (hw)
snd_pcm_hw_params_free(hw);
if (sw)
snd_pcm_sw_params_free(sw);
shm = NULL;
snd_pcm_close (pcm);
pcm = NULL;
return false;
}
int SNDDMA_GetDMAPos (void)
{
snd_pcm_uframes_t offset;
snd_pcm_uframes_t nframes;
const snd_pcm_channel_area_t *areas;
if (!shm)
return 0;
nframes = shm->samples/shm->channels;
snd_pcm_avail_update (pcm);
snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
// The following commit was absent in QF, causing the
// very first sound to be corrupted
snd_pcm_mmap_commit (pcm, offset, nframes);
offset *= shm->channels;
nframes *= shm->channels;
shm->samplepos = offset;
shm->buffer = (unsigned char *) areas->addr; // FIXME! there's an area per channel
return shm->samplepos;
}
void SNDDMA_Shutdown (void)
{
if (shm)
{
// full clean-up
Con_Printf ("Shutting down ALSA sound\n");
snd_pcm_drop (pcm); // do I need this?
snd_pcm_close (pcm);
pcm = NULL;
shm->buffer = NULL;
shm = NULL;
}
}
/*
==============
SNDDMA_LockBuffer
Makes sure dma buffer is valid
==============
*/
void SNDDMA_LockBuffer (void)
{
/* nothing to do here */
}
/*
==============
SNDDMA_Submit
Unlock the dma buffer /
Send sound to the device
==============
*/
void SNDDMA_Submit (void)
{
snd_pcm_uframes_t offset;
snd_pcm_uframes_t nframes;
const snd_pcm_channel_area_t *areas;
int state;
int count = paintedtime - soundtime;
nframes = count / shm->channels;
snd_pcm_avail_update (pcm);
snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
state = snd_pcm_state (pcm);
switch (state)
{
case SND_PCM_STATE_PREPARED:
snd_pcm_mmap_commit (pcm, offset, nframes);
snd_pcm_start (pcm);
break;
case SND_PCM_STATE_RUNNING:
snd_pcm_mmap_commit (pcm, offset, nframes);
break;
default:
break;
}
}
void SNDDMA_BlockSound (void)
{
snd_pcm_pause (pcm, 1);
}
void SNDDMA_UnblockSound (void)
{
snd_pcm_pause (pcm, 0);
}

314
quakespasm/Quake/snd_oss.c Normal file
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@ -0,0 +1,314 @@
/*
snd_oss.c
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:
Free Software Foundation, Inc.
59 Temple Place - Suite 330
Boston, MA 02111-1307, USA
*/
#include "quakedef.h"
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/mman.h>
#include <sys/shm.h>
#include <sys/wait.h>
// FIXME: <sys/soundcard.h> is "by the book" but we should take care of
// <soundcard.h>, <linux/soundcard.h> and <machine/soundcard.h> someday.
#include <sys/soundcard.h>
#include <errno.h>
static int FORMAT_S16;
static int audio_fd = -1;
static const char oss_default[] = "/dev/dsp";
static const char *ossdev = oss_default;
static unsigned long mmaplen;
static const int tryrates[] = { 11025, 22050, 44100, 48000, 16000, 24000, 8000 };
static const int MAX_TRYRATES = sizeof(tryrates)/sizeof(tryrates[0]);
qboolean SNDDMA_Init (dma_t *dma)
{
int i, caps, tmp;
unsigned long sz;
struct audio_buf_info info;
if (bigendien) FORMAT_S16 = AFMT_S16_BE;
else FORMAT_S16 = AFMT_S16_LE;
tmp = COM_CheckParm("-ossdev");
if (tmp != 0 && tmp < com_argc - 1)
ossdev = com_argv[tmp + 1];
Con_Printf ("OSS: Using device: %s\n", ossdev);
// open /dev/dsp, confirm capability to mmap, and get size of dma buffer
audio_fd = open(ossdev, O_RDWR|O_NONBLOCK);
if (audio_fd == -1)
{ // Failed open, retry up to 3 times if it's busy
tmp = 3;
while ( (audio_fd == -1) && tmp-- &&
((errno == EAGAIN) || (errno == EBUSY)) )
{
sleep (1);
audio_fd = open(ossdev, O_RDWR|O_NONBLOCK);
}
if (audio_fd == -1)
{
Con_Printf("Could not open %s. %s\n", ossdev, strerror(errno));
return false;
}
}
memset ((void *) dma, 0, sizeof(dma_t));
shm = dma;
if (ioctl(audio_fd, SNDCTL_DSP_RESET, 0) == -1)
{
Con_Printf("Could not reset %s. %s\n", ossdev, strerror(errno));
goto error;
}
if (ioctl(audio_fd, SNDCTL_DSP_GETCAPS, &caps) == -1)
{
Con_Printf("Couldn't retrieve soundcard capabilities. %s\n", strerror(errno));
goto error;
}
if (!(caps & DSP_CAP_TRIGGER) || !(caps & DSP_CAP_MMAP))
{
Con_Printf("Audio driver doesn't support mmap or trigger\n");
goto error;
}
// set sample bits & speed
i = (loadas8bit.value) ? 8 : 16;
tmp = (i == 16) ? FORMAT_S16 : AFMT_U8;
if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp) == -1)
{
Con_Printf("Problems setting %d bit format, trying alternatives..\n", i);
// try what the device gives us
if (ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp) == -1)
{
Con_Printf("Unable to retrieve supported formats. %s\n", strerror(errno));
goto error;
}
if (tmp & FORMAT_S16)
{
i = 16;
tmp = FORMAT_S16;
}
else if (tmp & AFMT_U8)
{
i = 8;
tmp = AFMT_U8;
}
else
{
Con_Printf("Neither 8 nor 16 bit format supported.\n");
goto error;
}
if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp) == -1)
{
Con_Printf("Unable to set sound format. %s\n", strerror(errno));
goto error;
}
}
shm->samplebits = i;
tmp = (int)sndspeed.value;
if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp) == -1)
{
Con_Printf("Problems setting sample rate, trying alternatives..\n");
shm->speed = 0;
for (i = 0; i < MAX_TRYRATES; i++)
{
tmp = tryrates[i];
if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp) == -1)
{
Con_DPrintf ("Unable to set sample rate %d\n", tryrates[i]);
}
else
{
if (tmp != tryrates[i])
{
Con_Printf ("Warning: Rate set (%d) didn't match requested rate (%d)!\n", tmp, tryrates[i]);
// goto error;
}
shm->speed = tmp;
break;
}
}
if (shm->speed == 0)
{
Con_Printf("Unable to set any sample rates.\n");
goto error;
}
}
else
{
if (tmp != (int)sndspeed.value)
{
Con_Printf ("Warning: Rate set (%d) didn't match requested rate (%d)!\n", tmp, (int)sndspeed.value);
// goto error;
}
shm->speed = tmp;
}
i = (COM_CheckParm("-sndmono") == 0) ? 2 : 1;
tmp = (i == 2) ? 1 : 0;
if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp) == -1)
{
Con_Printf ("Problems setting channels to %s, retrying for %s\n",
(i == 2) ? "stereo" : "mono",
(i == 2) ? "mono" : "stereo");
tmp = (i == 2) ? 0 : 1;
if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp) == -1)
{
Con_Printf("unable to set desired channels. %s\n", strerror(errno));
goto error;
}
}
shm->channels = tmp +1;
if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info) == -1)
{
Con_Printf("Couldn't retrieve buffer status. %s\n", strerror(errno));
goto error;
}
shm->samples = info.fragstotal * info.fragsize / (shm->samplebits / 8);
shm->submission_chunk = 1;
// memory map the dma buffer
sz = sysconf (_SC_PAGESIZE);
mmaplen = info.fragstotal * info.fragsize;
mmaplen = (mmaplen + sz - 1) & ~(sz - 1);
shm->buffer = (unsigned char *) mmap(NULL, mmaplen, PROT_READ|PROT_WRITE,
MAP_FILE|MAP_SHARED, audio_fd, 0);
if (!shm->buffer || shm->buffer == MAP_FAILED)
{
Con_Printf("Could not mmap %s. %s\n", ossdev, strerror(errno));
goto error;
}
Con_Printf ("OSS: mmaped %lu bytes buffer\n", mmaplen);
// toggle the trigger & start her up
tmp = 0;
if (ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &tmp) == -1)
{
Con_Printf("Could not toggle %s. %s\n", ossdev, strerror(errno));
munmap (shm->buffer, mmaplen);
goto error;
}
tmp = PCM_ENABLE_OUTPUT;
if (ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &tmp) == -1)
{
Con_Printf("Could not toggle %s. %s\n", ossdev, strerror(errno));
munmap (shm->buffer, mmaplen);
goto error;
}
shm->samplepos = 0;
return true;
error:
close(audio_fd);
audio_fd = -1;
shm->buffer = NULL;
shm = NULL;
return false;
}
int SNDDMA_GetDMAPos (void)
{
struct count_info count;
if (!shm)
return 0;
if (ioctl(audio_fd, SNDCTL_DSP_GETOPTR, &count) == -1)
{
Con_Printf("Uh, sound dead. %s\n", strerror(errno));
munmap (shm->buffer, mmaplen);
shm->buffer = NULL;
shm = NULL;
close(audio_fd);
audio_fd = -1;
return 0;
}
// shm->samplepos = (count.bytes / (shm->samplebits / 8)) & (shm->samples-1);
// fprintf(stderr, "%d \r", count.ptr);
shm->samplepos = count.ptr / (shm->samplebits / 8);
return shm->samplepos;
}
void SNDDMA_Shutdown (void)
{
int tmp = 0;
if (shm)
{
Con_Printf ("Shutting down OSS sound\n");
munmap (shm->buffer, mmaplen);
shm->buffer = NULL;
shm = NULL;
ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &tmp);
ioctl(audio_fd, SNDCTL_DSP_RESET, 0);
close(audio_fd);
audio_fd = -1;
}
}
/*
==============
SNDDMA_LockBuffer
Makes sure dma buffer is valid
==============
*/
void SNDDMA_LockBuffer (void)
{
/* nothing to do here */
}
/*
==============
SNDDMA_Submit
Unlock the dma buffer /
Send sound to the device
===============
*/
void SNDDMA_Submit(void)
{
}
void SNDDMA_BlockSound (void)
{
}
void SNDDMA_UnblockSound (void)
{
}

773
quakespasm/Quake/snd_win.c Normal file
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/*
snd_win.c
$Id: snd_win.c,v 1.37 2008-12-28 14:34:34 sezero Exp $
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:
Free Software Foundation, Inc.
51 Franklin St, Fifth Floor,
Boston, MA 02110-1301 USA
*/
#define DX_DLSYM /* dynamic loading of dsound symbols */
#include "quakedef.h"
#include "winquake.h"
#include <mmsystem.h>
#include <dsound.h>
//#define SNDBUFSIZE 65536
// 64K is > 1 second at 16-bit, 22050 Hz
//#define WAV_BUFFERS 64
#define WAV_BUFFERS 128
#define WAV_MASK (WAV_BUFFERS - 1)
/* DirectSound : */
#ifndef DSBSIZE_MIN
#define DSBSIZE_MIN 4
#endif
#ifndef DSBSIZE_MAX
#define DSBSIZE_MAX 0x0FFFFFFF
#endif
static LPDIRECTSOUND pDS;
static LPDIRECTSOUNDBUFFER pDSBuf, pDSPBuf;
#if defined(DX_DLSYM) /* dynamic loading of dsound symbols */
static HINSTANCE hInstDS;
static HRESULT (WINAPI *pDirectSoundCreate)(GUID FAR *lpGUID, LPDIRECTSOUND FAR *lplpDS, IUnknown FAR *pUnkOuter);
#else /* ! DX_DLSYM : we're linked to dsound */
#define pDirectSoundCreate DirectSoundCreate
#endif /* DX_DLSYM */
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_isdirect, snd_iswave;
static qboolean primary_format_set;
static int sample16;
static int snd_sent, snd_completed;
static int ds_sbuf_size, wv_buf_size;
static HANDLE hData;
static HGLOBAL hWaveHdr;
static HPSTR lpData;
static LPWAVEHDR lpWaveHdr;
static HWAVEOUT hWaveOut;
//WAVEOUTCAPS wavecaps;
static DWORD gSndBufSize;
static MMTIME mmstarttime;
/*
==================
FreeSound
==================
*/
static void FreeSound (void)
{
int i;
if (pDSBuf)
{
IDirectSoundBuffer_Stop(pDSBuf);
IDirectSound_Release(pDSBuf);
}
// only release primary buffer if it's not also the mixing buffer we just released
if (pDSPBuf && (pDSBuf != pDSPBuf))
{
IDirectSound_Release(pDSPBuf);
}
if (pDS)
{
IDirectSound_SetCooperativeLevel(pDS, mainwindow, DSSCL_NORMAL);
IDirectSound_Release(pDS);
}
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);
}
}
pDS = NULL;
pDSBuf = NULL;
pDSPBuf = NULL;
hWaveOut = 0;
hData = 0;
hWaveHdr = 0;
lpData = NULL;
lpWaveHdr = NULL;
dsound_init = false;
wav_init = false;
}
/*
==================
SNDDMA_InitDirect
Direct-Sound support
==================
*/
static sndinitstat SNDDMA_InitDirect (dma_t *dma)
{
DSBUFFERDESC dsbuf;
DSBCAPS dsbcaps;
DWORD dwSize, dwWrite;
DSCAPS dscaps;
WAVEFORMATEX format, pformat;
HRESULT hresult;
int reps;
memset((void *) dma, 0, sizeof(dma_t));
shm = dma;
shm->channels = 2; /* = desired_channels; */
shm->samplebits = (loadas8bit.value) ? 8 : 16;
shm->speed = sndspeed.value;
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 defined(DX_DLSYM)
if (!hInstDS)
{
hInstDS = LoadLibrary("dsound.dll");
if (hInstDS == NULL)
{
Con_SafePrintf ("Couldn't load dsound.dll\n");
return SIS_FAILURE;
}
pDirectSoundCreate = (HRESULT (WINAPI *)(GUID FAR *, LPDIRECTSOUND FAR *, IUnknown FAR *))
GetProcAddress(hInstDS,"DirectSoundCreate");
if (!pDirectSoundCreate)
{
Con_SafePrintf ("Couldn't get DS proc addr\n");
return SIS_FAILURE;
}
}
#endif /* DX_DLSYM */
hresult = pDirectSoundCreate(NULL, &pDS, NULL);
if (hresult != DS_OK)
{
if (hresult != DSERR_ALLOCATED)
{
Con_SafePrintf ("DirectSound create failed\n");
return SIS_FAILURE;
}
Con_SafePrintf ("DirectSoundCreate failure, hardware already in use\n");
return SIS_NOTAVAIL;
}
dscaps.dwSize = sizeof(dscaps);
if (DS_OK != IDirectSound_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;
}
// if (DS_OK != IDirectSound_SetCooperativeLevel(pDS, mainwindow, DSSCL_EXCLUSIVE))
/* Pa3PyX: Some MIDI synthesizers are software and require access to
waveOut; so if we set the coop level to exclusive, MIDI will fail
to init because the device is locked. We use priority level instead.
That way we don't lock out software synths and other apps, but can
still set the sound buffer format. */
if (DS_OK != IDirectSound_SetCooperativeLevel(pDS, mainwindow, DSSCL_PRIORITY))
{
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 == IDirectSound_CreateSoundBuffer(pDS, &dsbuf, &pDSPBuf, NULL))
{
pformat = format;
if (DS_OK != IDirectSoundBuffer_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;
if (ds_sbuf_size < DSBSIZE_MIN)
ds_sbuf_size = 1 << (Q_log2(DSBSIZE_MIN) + 1);
if (ds_sbuf_size > DSBSIZE_MAX)
ds_sbuf_size = 1 << Q_log2(DSBSIZE_MAX);
dsbuf.dwBufferBytes = ds_sbuf_size;
dsbuf.lpwfxFormat = &format;
memset(&dsbcaps, 0, sizeof(dsbcaps));
dsbcaps.dwSize = sizeof(dsbcaps);
if (DS_OK != IDirectSound_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 != IDirectSound_GetCaps(pDSBuf, &dsbcaps))
{
Con_SafePrintf ("DS:GetCaps failed\n");
FreeSound ();
return SIS_FAILURE;
}
if (snd_firsttime)
Con_SafePrintf ("Using secondary sound buffer\n");
}
else
{
if (DS_OK != IDirectSound_SetCooperativeLevel(pDS, mainwindow, DSSCL_WRITEPRIMARY))
{
Con_SafePrintf ("Set coop level failed\n");
FreeSound ();
return SIS_FAILURE;
}
if (DS_OK != IDirectSound_GetCaps(pDSPBuf, &dsbcaps))
{
Con_Printf ("DS:GetCaps failed\n");
return SIS_FAILURE;
}
pDSBuf = pDSPBuf;
Con_SafePrintf ("Using primary sound buffer\n");
}
// Make sure mixer is active
IDirectSoundBuffer_Play(pDSBuf, 0, 0, DSBPLAY_LOOPING);
if (snd_firsttime)
Con_SafePrintf ("%lu bytes in sound buffer\n", (unsigned long)dsbcaps.dwBufferBytes);
gSndBufSize = dsbcaps.dwBufferBytes;
// initialize the buffer
reps = 0;
while ((hresult = IDirectSoundBuffer_Lock(pDSBuf, 0, gSndBufSize, (LPVOID *) (HPSTR) &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
IDirectSoundBuffer_Unlock(pDSBuf, lpData, dwSize, NULL, 0);
/* we don't want anyone to access the buffer directly w/o locking it first. */
lpData = NULL;
IDirectSoundBuffer_Stop(pDSBuf);
IDirectSoundBuffer_GetCurrentPosition(pDSBuf, &mmstarttime.u.sample, &dwWrite);
IDirectSoundBuffer_Play(pDSBuf, 0, 0, DSBPLAY_LOOPING);
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;
}
/*
==================
SNDDM_InitWav
Crappy windows multimedia base
==================
*/
static qboolean SNDDMA_InitWav (dma_t *dma)
{
WAVEFORMATEX format;
int i;
HRESULT hr;
snd_sent = 0;
snd_completed = 0;
memset((void *) dma, 0, sizeof(dma_t));
shm = dma;
shm->channels = 2; /* = desired_channels; */
shm->samplebits = (loadas8bit.value) ? 8 : 16;
shm->speed = sndspeed.value;
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. */
hr = waveOutOpen((LPHWAVEOUT)&hWaveOut, WAVE_MAPPER, &format, 0, 0L, CALLBACK_NULL);
if (hr != MMSYSERR_NOERROR)
{
if (hr != MMSYSERR_ALLOCATED)
{
Con_SafePrintf ("waveOutOpen failed\n");
return false;
}
Con_SafePrintf ("waveOutOpen failure, 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 * wv_buf_size;
hData = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, gSndBufSize);
if (!hData)
{
Con_SafePrintf ("Sound: Out of memory.\n");
FreeSound ();
return false;
}
lpData = (HPSTR) 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 = wv_buf_size;
lpWaveHdr[i].lpData = lpData + i * wv_buf_size;
if (waveOutPrepareHeader(hWaveOut, lpWaveHdr+i, sizeof(WAVEHDR)) != MMSYSERR_NOERROR)
{
Con_SafePrintf ("Sound: failed to prepare wave headers\n");
FreeSound ();
return 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;
Con_SafePrintf ("%d sound buffers, %d bytes/sound buffer\n", WAV_BUFFERS, wv_buf_size);
return true;
}
/*
==================
SNDDMA_Init
Try to find a sound device to mix for.
Returns false if nothing is found.
==================
*/
qboolean SNDDMA_Init (dma_t *dma)
{
sndinitstat stat;
int sndbits = (loadas8bit.value) ? 8 : 16;
if (COM_CheckParm ("-wavonly"))
wavonly = true;
dsound_init = wav_init = 0;
stat = SIS_FAILURE; // assume DirectSound won't initialize
/* Calculate Wave and DS buffer sizes to set, to store
2 secs of data, round up to the next power of 2 */
ds_sbuf_size = 1 << (Q_log2((sndbits >> 3) * ((int)sndspeed.value << 1)) + 1);
wv_buf_size = 1 << (Q_log2(((int)sndspeed.value << 3) / WAV_BUFFERS) + 1);
/* Init DirectSound */
if (!wavonly)
{
if (snd_firsttime || snd_isdirect)
{
stat = SNDDMA_InitDirect (dma);
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");
}
}
}
// 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 (dma);
if (snd_iswave)
{
if (snd_firsttime)
Con_SafePrintf ("Wave sound initialized\n");
}
else
{
Con_SafePrintf ("Wave sound failed to init\n");
}
}
}
if (!dsound_init && !wav_init)
{
if (snd_firsttime)
Con_SafePrintf ("No sound device initialized\n");
snd_firsttime = false;
return false;
}
snd_firsttime = false;
return true;
}
/*
==============
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)
{
MMTIME mmtime;
int s;
DWORD dwWrite;
if (dsound_init)
{
mmtime.wType = TIME_SAMPLES;
IDirectSoundBuffer_GetCurrentPosition(pDSBuf, &mmtime.u.sample, &dwWrite);
s = mmtime.u.sample - mmstarttime.u.sample;
}
else if (wav_init)
{
s = snd_sent * wv_buf_size;
}
else
{ // we should not reach here...
return 0;
}
s >>= sample16;
s &= (shm->samples-1);
return s;
}
/*
==============
SNDDMA_LockBuffer
Makes sure dma buffer is valid
===============
*/
static DWORD locksize;
void SNDDMA_LockBuffer (void)
{
if (pDSBuf)
{
void *pData;
int reps;
HRESULT hresult;
DWORD dwStatus;
reps = 0;
shm->buffer = NULL;
if (IDirectSoundBuffer_GetStatus(pDSBuf, &dwStatus) != DS_OK)
Con_Printf ("Couldn't get sound buffer status\n");
if (dwStatus & DSBSTATUS_BUFFERLOST)
IDirectSoundBuffer_Restore(pDSBuf);
if (!(dwStatus & DSBSTATUS_PLAYING))
IDirectSoundBuffer_Play(pDSBuf, 0, 0, DSBPLAY_LOOPING);
while ((hresult = IDirectSoundBuffer_Lock(pDSBuf, 0, gSndBufSize, (void **) &pData, &locksize, NULL, NULL, 0)) != DS_OK)
{
if (hresult != DSERR_BUFFERLOST)
{
Con_Printf ("SNDDMA_LockBuffer: DS::Lock Sound Buffer Failed\n");
S_Shutdown ();
return;
}
if (++reps > 10000)
{
Con_Printf ("SNDDMA_LockBuffer: DS: couldn't restore buffer\n");
S_Shutdown ();
return;
}
}
shm->buffer = (unsigned char *) pData;
}
}
/*
==============
SNDDMA_Submit
Unlock the dma buffer /
Send sound to the device
===============
*/
void SNDDMA_Submit (void)
{
LPWAVEHDR h;
int wResult;
if (pDSBuf)
IDirectSoundBuffer_Unlock(pDSBuf, shm->buffer, locksize, NULL, 0);
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_BlockSound
==================
*/
void SNDDMA_BlockSound (void)
{
// DirectSound takes care of blocking itself
if (snd_iswave)
{
waveOutReset (hWaveOut);
}
}
/*
==================
SNDDMA_UnblockSound
==================
*/
void SNDDMA_UnblockSound (void)
{
}
/*
==============
SNDDMA_Shutdown
Reset the sound device for exiting
===============
*/
void SNDDMA_Shutdown (void)
{
FreeSound ();
#if defined(DX_DLSYM)
if (hInstDS)
{
FreeLibrary(hInstDS);
hInstDS = NULL;
}
#endif /* DX_DLSYM */
}