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Merge pull request #8 from nzp-team/sdl-audio

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Move Audio handling from internal to SDL
This commit is contained in:
Ian 2023-03-13 15:22:50 -04:00 committed by GitHub
commit d1d2671880
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GPG key ID: 4AEE18F83AFDEB23
9 changed files with 182 additions and 237 deletions
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11
MakePHAT
View file

@ -13,7 +13,6 @@ PSP_FW_VERSION=660
MODE=-DKERNEL_MODE
ifeq ($(USE_GPROF),1)
GPROF_LIBS = -lpspprof
GPROF_FLAGS = -pg -DPROFILE
@ -28,7 +27,6 @@ COMMON_OBJS = \
source/psp/input.o \
source/psp/main.o \
source/psp/math.o \
source/psp/sound.o \
source/psp/system.o \
source/psp/module.o \
source/psp/network.o \
@ -64,6 +62,7 @@ COMMON_OBJS = \
source/pr_cmds.o \
source/pr_edict.o \
source/pr_exec.o \
source/snd_sdl.o \
source/snd_dma.o \
source/snd_mem.o \
source/snd_mix.o \
@ -109,16 +108,14 @@ HARDWARE_VIDEO_ONLY_FLAGS = -DPSP_HARDWARE_VIDEO
OBJS = $(COMMON_OBJS) $(HARDWARE_VIDEO_ONLY_OBJS)
#LIBS = -lpspaudiolib -lpspaudio -lpspgum -lpspgu -lpsprtc -lpsppower -lpspwlan -lstdc++ -lm
SDL_LIBS = -lSDL2 -lvorbisfile -lvorbis -logg -lGL -lGLU -lglut
GU_LIBS = -lpspgum_vfpu -lpspvfpu -lpspgu -lpspvram
AUDIO_LIBS = -lpspaudiolib -lpspaudio -lpspmp3 source/psp/m33libs/libpspaudiocodec.a source/psp/m33libs/libpspkubridge.a
MISC_LIBS = -lpsprtc -lpspmath -lpsppower -lpsphprm -ljpeg -lpng source/psp/m33libs/libz.a
MISC_LIBS = -lpsprtc -lpsppower -lpspmath -lpsphprm -ljpeg -lpng source/psp/m33libs/libz.a
NET_LIBS = -lpspwlan -lpspnet_adhoc -lpspnet_adhocctl
STD_LIBS = -lstdc++ -lm -lc
LIBS = $(GPROF_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
LIBS = $(GPROF_LIBS) $(SDL_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
CFLAGS = -ffast-math -O3 -G0 -Wall $(GPROF_FLAGS) -Did386="0" -DPSP $(MODE) $(HARDWARE_VIDEO_ONLY_FLAGS) -DSWIZZLE32 -DPSP_MP3_HWDECODE -DFULLBRIGHT -DHL_RENDER -Wno-strict-aliasing -DPSP_VFPU
CXXFLAGS = -fno-rtti -Wcast-qual -Wno-write-strings -Wno-sign-compare -Wno-strict-aliasing

8
MakeSLIM
View file

@ -29,7 +29,6 @@ COMMON_OBJS = \
source/psp/input.o \
source/psp/main.o \
source/psp/math.o \
source/psp/sound.o \
source/psp/system.o \
source/psp/module.o \
source/psp/network.o \
@ -65,6 +64,7 @@ COMMON_OBJS = \
source/pr_cmds.o \
source/pr_edict.o \
source/pr_exec.o \
source/snd_sdl.o \
source/snd_dma.o \
source/snd_mem.o \
source/snd_mix.o \
@ -110,16 +110,14 @@ HARDWARE_VIDEO_ONLY_FLAGS = -DPSP_HARDWARE_VIDEO
OBJS = $(COMMON_OBJS) $(HARDWARE_VIDEO_ONLY_OBJS)
#LIBS = -lpspaudiolib -lpspaudio -lpspgum -lpspgu -lpsprtc -lpsppower -lpspwlan -lstdc++ -lm
SDL_LIBS = -lSDL2 -lvorbisfile -lvorbis -logg -lGL -lGLU -lglut
GU_LIBS = -lpspgum_vfpu -lpspvfpu -lpspgu -lpspvram
AUDIO_LIBS = -lpspaudiolib -lpspaudio -lpspmp3 source/psp/m33libs/libpspaudiocodec.a source/psp/m33libs/libpspkubridge.a
MISC_LIBS = -lpsprtc -lpsppower -lpspmath -lpsphprm -ljpeg -lpng source/psp/m33libs/libz.a
NET_LIBS = -lpspwlan -lpspnet_adhoc -lpspnet_adhocctl
STD_LIBS = -lstdc++ -lm -lc
LIBS = $(GPROF_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
LIBS = $(GPROF_LIBS) $(SDL_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
CFLAGS = -ffast-math -O3 -G0 -Wall $(GPROF_FLAGS) -Did386="0" -DPSP $(MODE) $(HARDWARE_VIDEO_ONLY_FLAGS) -DSWIZZLE32 -DSLIM -DPSP_MP3_HWDECODE -DFULLBRIGHT -DHL_RENDER -Wno-strict-aliasing -DPSP_VFPU
CXXFLAGS = -fno-rtti -Wcast-qual -Wno-write-strings -Wno-sign-compare -Wno-strict-aliasing

11
source/host.c
View file

@ -759,18 +759,11 @@ void _Host_Frame (float time)
// update audio
if (cls.signon == SIGNONS)
{
Thread_UpdateSound(r_origin, vpn, vright, vup);
//S_Update (r_origin, vpn, vright, vup);
S_Update (r_origin, vpn, vright, vup);
CL_DecayLights ();
}
else
Thread_UpdateSound(vec3_origin, vec3_origin, vec3_origin, vec3_origin);
//S_Update (vec3_origin, vec3_origin, vec3_origin, vec3_origin);
//if (bmg_type_changed == true) {
CDAudio_Update();
// bmg_type_changed = false;
//}
S_Update (vec3_origin, vec3_origin, vec3_origin, vec3_origin);
if (host_speeds.value)
{

4
source/psp/cd.cpp
View file

@ -180,7 +180,7 @@ void CDAudio_Track(char* trackname)
void CDAudio_Play(byte track, qboolean looping)
{
last_track = track;
/*last_track = track;
CDAudio_Stop();
if (track < 1)
@ -207,7 +207,7 @@ void CDAudio_Play(byte track, qboolean looping)
}
CDAudio_VolumeChange(bgmvolume.value);
CDAudio_VolumeChange(bgmvolume.value);*/
}
void CDAudio_Stop(void)

17
source/psp/main.cpp
View file

@ -36,6 +36,9 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#include <pspge.h>
#include <pspsysevent.h>
#include <SDL2/SDL.h>
#include <SDL2/SDL_mixer.h>
extern "C"
{
#include "../quakedef.h"
@ -86,7 +89,7 @@ namespace quake
#else
static size_t heapSize = 9 * 1024 * 1024;
static size_t heapSize = 10 * 1024 * 1024;
#endif // SLIM
@ -528,9 +531,16 @@ int user_main(SceSize argc, void* argp)
// operations.
disableFloatingPointExceptions();
// Initialise the Common module.
// Initialize the Common module.
InitExtModules ();
// Initialize SDL
if (SDL_Init(SDL_INIT_AUDIO) < 0)
{
Sys_Error("SDL2: Could not initialize!\n");
return 0;
}
// Get the current working dir.
char currentDirectory[1024];
char gameDirectory[1024];
@ -633,9 +643,6 @@ int user_main(SceSize argc, void* argp)
u64 lastTicks;
sceRtcGetCurrentTick(&lastTicks);
// Set up threads
Sys_InitThreads();
// Enter the main loop.
while (!quit)
{

157
source/psp/sound.cpp
View file

@ -1,157 +0,0 @@
/*
Copyright (C) 1996-1997 Id Software, Inc.
Copyright (C) 2007 Peter Mackay and Chris Swindle.
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 <pspaudiolib.h>
#include <pspkernel.h>
extern "C"
{
#include "../quakedef.h"
}
namespace quake
{
namespace sound
{
struct Sample
{
short left;
short right;
};
static const unsigned int channelCount = 2;
static const unsigned int inputBufferSize = 16384;
static const unsigned int inputFrequency = 11025;
static const unsigned int outputFrequency = 44100;
static const unsigned int inputSamplesPerOutputSample = outputFrequency / inputFrequency;
static Sample inputBuffer[inputBufferSize];
static volatile unsigned int samplesRead;
static inline void copySamples(const Sample* first, const Sample* last, Sample* destination)
{
// just assume inputSamplesPerOutputSample is 4.
for (const Sample* source = first; source != last; ++source)
{
const Sample sample = *source;
*destination++ = sample;
*destination++ = sample;
*destination++ = sample;
*destination++ = sample;
}
}
static void fillOutputBuffer(void* buffer, unsigned int samplesToWrite, void* userData)
{
// Where are we writing to?
Sample* const destination = static_cast<Sample*> (buffer);
// Where are we reading from?
const Sample* const firstSampleToRead = &inputBuffer[samplesRead];
// How many samples to read?
const unsigned int samplesToRead = samplesToWrite / inputSamplesPerOutputSample;
// Going to wrap past the end of the input buffer?
const unsigned int samplesBeforeEndOfInput = inputBufferSize - samplesRead;
if (samplesToRead > samplesBeforeEndOfInput)
{
// Yes, so write the first chunk from the end of the input buffer.
copySamples(
firstSampleToRead,
firstSampleToRead + samplesBeforeEndOfInput,
&destination[0]);
// Write the second chunk from the start of the input buffer.
const unsigned int samplesToReadFromBeginning = samplesToRead - samplesBeforeEndOfInput;
copySamples(
&inputBuffer[0],
&inputBuffer[samplesToReadFromBeginning],
&destination[samplesBeforeEndOfInput * inputSamplesPerOutputSample]);
}
else
{
// No wrapping, just copy.
copySamples(
firstSampleToRead,
firstSampleToRead + samplesToRead,
&destination[0]);
}
// Update the read offset.
samplesRead = (samplesRead + samplesToRead) % inputBufferSize;
}
}
}
using namespace quake;
using namespace quake::sound;
qboolean SNDDMA_Init(void)
{
// Set up Quake's audio.
shm = &sn;
shm->channels = channelCount;
shm->samplebits = 16;
shm->speed = inputFrequency;
shm->soundalive = qtrue;
shm->splitbuffer = qfalse;
shm->samples = inputBufferSize * channelCount;
shm->samplepos = 0;
shm->submission_chunk = 1;
shm->buffer = (unsigned char *) inputBuffer;
// Initialise the audio system. This initialises it for the CD audio module
// too.
pspAudioInit();
// Set the channel callback.
// Sound effects use channel 0, CD audio uses channel 1.
pspAudioSetChannelCallback(0, fillOutputBuffer, 0);
return qtrue;
}
void SNDDMA_Shutdown(void)
{
// Clear the mixing buffer so we don't get any noise during cleanup.
memset(inputBuffer, 0, sizeof(inputBuffer));
// Clear the channel callback.
pspAudioSetChannelCallback(0, 0, 0);
// Stop the audio system?
pspAudioEndPre();
// Insert a false delay so the thread can be cleaned up.
sceKernelDelayThread(50 * 1000);
// Shut down the audio system.
pspAudioEnd();
}
int SNDDMA_GetDMAPos(void)
{
return samplesRead * channelCount;
}
void SNDDMA_Submit(void)
{
}

9
source/psp/video_hardware_model.cpp
View file

@ -2112,12 +2112,21 @@ void *Mod_LoadAllSkins (int numskins, daliasskintype_t *pskintype)
COM_StripExtension(loadmodel->name, model);
// HACK HACK HACK
sprintf (model2, "%s.mdl_%i", model, i);
// Sigh.. something is totally awry with memory and the Slim kinda band-aids it..
// Textures can occupy RAM and VRAM at the same time, ie RAM isnt being freed properly
// So PHAT ends up kicking the bucket with a lot of external textures.
#ifdef SLIM
pheader->gl_texturenum[i][0] =
pheader->gl_texturenum[i][1] =
pheader->gl_texturenum[i][2] =
pheader->gl_texturenum[i][3] = loadtextureimage (model2, 0, 0, qtrue, GU_LINEAR);
if (pheader->gl_texturenum[i][0] == 0)// did not find a matching TGA...
#endif // SLIM
{
sprintf (name, "%s_%i", loadmodel->name, i);
if(mod_h2)

100
source/snd_dma.c
View file

@ -8,7 +8,7 @@ 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.
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
See the GNU General Public License for more details.
@ -20,7 +20,6 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
// snd_dma.c -- main control for any streaming sound output device
#include "quakedef.h"
#include "thread.h"
void S_Play(void);
void S_PlayVol(void);
@ -60,7 +59,7 @@ int num_sfx;
sfx_t *ambient_sfx[NUM_AMBIENTS];
int desired_speed = 11025;
int desired_speed = 44100; //11025;
int desired_bits = 16;
int sound_started=0;
@ -115,7 +114,7 @@ void S_SoundInfo_f(void)
Con_Printf ("sound system not started\n");
return;
}
Con_Printf("%5d stereo\n", shm->channels - 1);
Con_Printf("%5d samples\n", shm->samples);
Con_Printf("%5d samplepos\n", shm->samplepos);
@ -184,14 +183,13 @@ void S_Init (void)
Cvar_RegisterVariable(&loadas8bit);
Cvar_RegisterVariable(&bgmvolume);
Cvar_RegisterVariable(&bgmbuffer);
Cvar_RegisterVariable(&bgmtype);
Cvar_RegisterVariable(&ambient_level);
Cvar_RegisterVariable(&ambient_fade);
Cvar_RegisterVariable(&snd_noextraupdate);
Cvar_RegisterVariable(&snd_show);
Cvar_RegisterVariable(&_snd_mixahead);
if (host_parms.memsize < 0x800000)
//if (host_parms.memsize < 0x800000)
{
Cvar_Set ("loadas8bit", "1");
Con_Printf ("loading all sounds as 8bit\n");
@ -225,15 +223,17 @@ void S_Init (void)
shm->buffer = Hunk_AllocName(1<<16, "shmbuf");
}
Con_Printf ("Sound sampling rate: %i\n", shm->speed);
if ( shm ) {
Con_Printf ("Sound sampling rate: %i\n", shm->speed);
}
// provides a tick sound until washed clean
// if (shm->buffer)
// shm->buffer[4] = shm->buffer[5] = 0x7f; // force a pop for debugging
ambient_sfx[AMBIENT_WATER] = S_PrecacheSound ("sounds/ambience/water1.wav");
ambient_sfx[AMBIENT_SKY] = S_PrecacheSound ("sounds/ambience/wind2.wav");
ambient_sfx[AMBIENT_WATER] = S_PrecacheSound ("ambience/water1.wav");
ambient_sfx[AMBIENT_SKY] = S_PrecacheSound ("ambience/wind2.wav");
S_StopAllSounds (true);
}
@ -292,12 +292,12 @@ sfx_t *S_FindName (char *name)
if (num_sfx == MAX_SFX)
Sys_Error ("S_FindName: out of sfx_t");
sfx = &known_sfx[i];
strcpy (sfx->name, name);
num_sfx++;
return sfx;
}
@ -311,7 +311,7 @@ S_TouchSound
void S_TouchSound (char *name)
{
sfx_t *sfx;
if (!sound_started)
return;
@ -333,11 +333,11 @@ sfx_t *S_PrecacheSound (char *name)
return NULL;
sfx = S_FindName (name);
// cache it in
if (precache.value)
S_LoadSound (sfx);
return sfx;
}
@ -385,8 +385,8 @@ channel_t *SND_PickChannel(int entnum, int entchannel)
if (channels[first_to_die].sfx)
channels[first_to_die].sfx = NULL;
return &channels[first_to_die];
}
return &channels[first_to_die];
}
/*
=================
@ -396,7 +396,7 @@ SND_Spatialize
void SND_Spatialize(channel_t *ch)
{
vec_t dot;
vec_t dist;
vec_t ldist, rdist, dist;
vec_t lscale, rscale, scale;
vec3_t source_vec;
sfx_t *snd;
@ -413,9 +413,9 @@ void SND_Spatialize(channel_t *ch)
snd = ch->sfx;
VectorSubtract(ch->origin, listener_origin, source_vec);
dist = VectorNormalize(source_vec) * ch->dist_mult;
dot = DotProduct(listener_right, source_vec);
if (shm->channels == 1)
@ -439,7 +439,7 @@ void SND_Spatialize(channel_t *ch)
ch->leftvol = (int) (ch->master_vol * scale);
if (ch->leftvol < 0)
ch->leftvol = 0;
}
}
// =======================================================================
@ -469,7 +469,7 @@ void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float f
target_chan = SND_PickChannel(entnum, entchannel);
if (!target_chan)
return;
// spatialize
memset (target_chan, 0, sizeof(*target_chan));
VectorCopy(origin, target_chan->origin);
@ -492,7 +492,7 @@ void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float f
target_chan->sfx = sfx;
target_chan->pos = 0.0;
target_chan->end = paintedtime + sc->length;
target_chan->end = paintedtime + sc->length;
// if an identical sound has also been started this frame, offset the pos
// a bit to keep it from just making the first one louder
@ -510,7 +510,7 @@ void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float f
target_chan->end -= skip;
break;
}
}
}
@ -557,7 +557,7 @@ void S_StopAllSoundsC (void)
void S_ClearBuffer (void)
{
int clear;
if (!sound_started || !shm || !shm->buffer)
return;
@ -566,10 +566,7 @@ void S_ClearBuffer (void)
else
clear = 0;
{
Q_memset(shm->buffer, clear, shm->samples * shm->samplebits/8);
}
Q_memset(shm->buffer, clear, shm->samples * shm->samplebits/8);
}
@ -604,13 +601,13 @@ void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation)
Con_Printf ("Sound %s not looped\n", sfx->name);
return;
}
ss->sfx = sfx;
VectorCopy (origin, ss->origin);
ss->master_vol = vol;
ss->dist_mult = (attenuation/64) / sound_nominal_clip_dist;
ss->end = paintedtime + sc->length;
ss->end = paintedtime + sc->length;
SND_Spatialize (ss);
}
@ -646,9 +643,9 @@ void S_UpdateAmbientSounds (void)
for (ambient_channel = 0 ; ambient_channel< NUM_AMBIENTS ; ambient_channel++)
{
chan = &channels[ambient_channel];
chan = &channels[ambient_channel];
chan->sfx = ambient_sfx[ambient_channel];
vol = ambient_level.value * l->ambient_sound_level[ambient_channel];
if (vol < 8)
vol = 0;
@ -666,7 +663,7 @@ void S_UpdateAmbientSounds (void)
if (chan->master_vol < vol)
chan->master_vol = vol;
}
chan->leftvol = chan->rightvol = chan->master_vol;
}
}
@ -693,13 +690,13 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
VectorCopy(forward, listener_forward);
VectorCopy(right, listener_right);
VectorCopy(up, listener_up);
// update general area ambient sound sources
S_UpdateAmbientSounds ();
combine = NULL;
// update spatialization for static and dynamic sounds
// update spatialization for static and dynamic sounds
ch = channels+NUM_AMBIENTS;
for (i=NUM_AMBIENTS ; i<total_channels; i++, ch++)
{
@ -711,7 +708,7 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
// try to combine static sounds with a previous channel of the same
// sound effect so we don't mix five torches every frame
if (i >= MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS)
{
// see if it can just use the last one
@ -727,7 +724,7 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
for (j=MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS ; j<i; j++, combine++)
if (combine->sfx == ch->sfx)
break;
if (j == total_channels)
{
combine = NULL;
@ -743,8 +740,8 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
continue;
}
}
}
//
@ -760,7 +757,7 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
//Con_Printf ("%3i %3i %s\n", ch->leftvol, ch->rightvol, ch->sfx->name);
total++;
}
Con_Printf ("----(%i)----\n", total);
}
@ -774,7 +771,7 @@ void GetSoundtime(void)
static int buffers;
static int oldsamplepos;
int fullsamples;
fullsamples = shm->samples / shm->channels;
// it is possible to miscount buffers if it has wrapped twice between
@ -785,7 +782,7 @@ void GetSoundtime(void)
if (samplepos < oldsamplepos)
{
buffers++; // buffer wrapped
if (paintedtime > 0x40000000)
{ // time to chop things off to avoid 32 bit limits
buffers = 0;
@ -800,9 +797,6 @@ void GetSoundtime(void)
void S_ExtraUpdate (void)
{
if (snd_noextraupdate.value)
return; // don't pollute timings
S_Update_();
@ -810,9 +804,11 @@ void S_ExtraUpdate (void)
void S_Update_(void)
{
/*#ifndef SDL
unsigned endtime;
int samps;
if (!sound_started || (snd_blocked > 0))
return;
@ -835,6 +831,7 @@ void S_Update_(void)
S_PaintChannels (endtime);
SNDDMA_Submit ();
#endif*/
}
/*
@ -851,7 +848,7 @@ void S_Play(void)
int i;
char name[256];
sfx_t *sfx;
i = 1;
while (i<Cmd_Argc())
{
@ -875,7 +872,7 @@ void S_PlayVol(void)
float vol;
char name[256];
sfx_t *sfx;
i = 1;
while (i<Cmd_Argc())
{
@ -926,7 +923,7 @@ void S_LocalSound (char *sound)
return;
if (!sound_started)
return;
sfx = S_PrecacheSound (sound);
if (!sfx)
{
@ -949,5 +946,4 @@ void S_BeginPrecaching (void)
void S_EndPrecaching (void)
{
}
}

102
source/snd_sdl.c Normal file
View file

@ -0,0 +1,102 @@
#include <stdio.h>
#include <SDL2/SDL_audio.h>
#include "quakedef.h"
static dma_t the_shm;
static int snd_inited;
extern int desired_speed;
extern int desired_bits;
static void paint_audio(void *unused, Uint8 *stream, int len)
{
if ( shm ) {
shm->buffer = stream;
shm->samplepos += len/(shm->samplebits/8)/2;
// Check for samplepos overflow?
S_PaintChannels (shm->samplepos);
}
}
qboolean SNDDMA_Init(void)
{
SDL_AudioSpec desired, obtained;
snd_inited = 0;
/* Set up the desired format */
desired.freq = desired_speed;
switch (desired_bits) {
case 8:
desired.format = AUDIO_U8;
break;
case 16:
desired.format = AUDIO_S16LSB;
break;
default:
Con_Printf("Unknown number of audio bits: %d\n",
desired_bits);
return 0;
}
desired.channels = 2;
desired.samples = 512;
desired.callback = paint_audio;
/* Open the audio device */
if ( SDL_OpenAudio(&desired, &obtained) < 0 ) {
Con_Printf("Couldn't open SDL audio: %s\n", SDL_GetError());
return 0;
}
/* Make sure we can support the audio format */
switch (obtained.format) {
case AUDIO_U8:
/* Supported */
break;
case AUDIO_S16LSB:
case AUDIO_S16MSB:
/* Supported */
break;
/* Unsupported, fall through */;
default:
/* Not supported -- force SDL to do our bidding */
SDL_CloseAudio();
if ( SDL_OpenAudio(&desired, NULL) < 0 ) {
Con_Printf("Couldn't open SDL audio: %s\n",
SDL_GetError());
return 0;
}
memcpy(&obtained, &desired, sizeof(desired));
break;
}
SDL_PauseAudio(0);
/* Fill the audio DMA information block */
shm = &the_shm;
shm->splitbuffer = 0;
shm->samplebits = (obtained.format & 0xFF);
shm->speed = obtained.freq;
shm->channels = obtained.channels;
shm->samples = obtained.samples*shm->channels;
shm->samplepos = 0;
shm->submission_chunk = 1;
shm->buffer = NULL;
snd_inited = 1;
return 1;
}
int SNDDMA_GetDMAPos(void)
{
return shm->samplepos;
}
void SNDDMA_Shutdown(void)
{
if (snd_inited)
{
SDL_CloseAudio();
snd_inited = 0;
}
}