fteqw/engine/client/snd_sdl.c

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#include "quakedef.h"
#include "winquake.h"
#include <SDL.h>
#pragma comment(lib, MSVCLIBSPATH "sdl.lib")
//SDL calls a callback each time it needs to repaint the 'hardware' buffers
//This results in extra latency.
//SDL runs does this multithreaded.
//So we tell it a fairly pathetically sized buffer and try and get it to copy often
//hopefully this lowers sound latency, and has no suddenly starting sounds and stuff.
//It still has greater latency than direct access, of course.
//FIXME: One thing I saw in quakeforge was that quakeforge basically leaves the audio locked except for a really short period of time.
//An interesting idea, which ensures the driver can only paint in a small time-frame. this would possibly allow lower latency painting.
static void SSDL_Shutdown(soundcardinfo_t *sc)
{
Con_Printf("Shutdown SDL sound\n");
SDL_CloseAudio();
if (sc->sn.buffer)
free(sc->sn.buffer);
sc->sn.buffer = NULL;
}
static unsigned int SSDL_GetDMAPos(soundcardinfo_t *sc)
{
sc->sn.samplepos = sc->snd_sent / (sc->sn.samplebits/8);
return sc->sn.samplepos;
}
//this function is called from inside SDL.
//transfer the 'dma' buffer into the buffer it requests.
static void VARGS SSDL_Paint(void *userdata, qbyte *stream, int len)
{
soundcardinfo_t *sc = userdata;
int buffersize = sc->sn.samples*(sc->sn.samplebits/8);
if (len > buffersize)
{
len = buffersize; //whoa nellie!
}
if (len + sc->snd_sent%buffersize > buffersize)
{ //buffer will wrap, fill in the rest
memcpy(stream, (char*)sc->sn.buffer + (sc->snd_sent%buffersize), buffersize - (sc->snd_sent%buffersize));
stream += buffersize - sc->snd_sent%buffersize;
sc->snd_sent += buffersize - sc->snd_sent%buffersize;
len -= buffersize - (sc->snd_sent%buffersize);
if (len < 0)
return;
} //and finish from the start
memcpy(stream, (char*)sc->sn.buffer + (sc->snd_sent%buffersize), len);
sc->snd_sent += len;
}
static void *SSDL_LockBuffer(soundcardinfo_t *sc)
{
SDL_LockAudio();
return sc->sn.buffer;
}
static void SSDL_UnlockBuffer(soundcardinfo_t *sc, void *buffer)
{
SDL_UnlockAudio();
}
static void SSDL_SetUnderWater(soundcardinfo_t *sc, qboolean uw)
{
}
static void SSDL_Submit(soundcardinfo_t *sc)
{
//SDL will call SSDL_Paint to paint when it's time, and the sound buffer is always there...
}
static int SDL_InitCard(soundcardinfo_t *sc, int cardnum)
{
SDL_AudioSpec desired, obtained;
if (cardnum)
{ //our init code actually calls this function multiple times, in the case that the user has multiple sound cards
return 2; //erm. SDL won't allow multiple sound cards anyway.
}
Con_Printf("Initing SDL audio.\n");
if(SDL_InitSubSystem(SDL_INIT_AUDIO | SDL_INIT_NOPARACHUTE))
{
Con_Printf("Couldn't initialize SDL audio subsystem\n");
return false;
}
memset(&desired, 0, sizeof(desired));
desired.freq = sc->sn.speed;
desired.channels = sc->sn.numchannels; //fixme!
desired.samples = 0x0100;
desired.format = AUDIO_S16SYS;
desired.callback = (void*)SSDL_Paint;
desired.userdata = sc;
memcpy(&obtained, &desired, sizeof(obtained));
if ( SDL_OpenAudio(&desired, &obtained) < 0 )
{
Con_Printf("SDL: SNDDMA_Init: couldn't open sound device (%s).\n", SDL_GetError());
return false;
}
sc->sn.numchannels = obtained.channels;
sc->sn.speed = obtained.freq;
sc->sn.samplebits = obtained.format&0xff;
sc->sn.samples = 32768;//*sc->sn.numchannels; //doesn't really matter, so long as it's higher than obtained.samples
Con_DPrintf("channels: %i\n", sc->sn.numchannels);
Con_DPrintf("Speed: %i\n", sc->sn.speed);
Con_DPrintf("Samplebits: %i\n", sc->sn.samplebits);
Con_DPrintf("SDLSamples: %i (low for latency)\n", obtained.samples);
Con_DPrintf("FakeSamples: %i\n", sc->sn.samples);
sc->sn.buffer = malloc(sc->sn.samples*sc->sn.samplebits/8);
Con_DPrintf("Got sound %i-%i\n", obtained.freq, obtained.format);
sc->Lock = SSDL_LockBuffer;
sc->Unlock = SSDL_UnlockBuffer;
sc->SetWaterDistortion = SSDL_SetUnderWater;
sc->Submit = SSDL_Submit;
sc->Shutdown = SSDL_Shutdown;
sc->GetDMAPos = SSDL_GetDMAPos;
SDL_PauseAudio(0);
return true;
}
sounddriver pSDL_InitCard = &SDL_InitCard;