SDL sound should be working fine now

git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@1563 fc73d0e0-1445-4013-8a0c-d673dee63da5
This commit is contained in:
Spoike 2005-11-07 00:59:22 +00:00
parent b29f2b0102
commit c320fbe628
1 changed files with 52 additions and 23 deletions

View File

@ -3,7 +3,15 @@
#include <SDL.h>
#define SOUND_BUFFER_SIZE 0x0400
//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)
{
@ -17,7 +25,8 @@ Con_Printf("down\n");
}
static unsigned int SSDL_GetDMAPos(soundcardinfo_t *sc)
{
sc->sn.samplepos = (sc->snd_sent / (sc->sn.samplebits/8)) % sc->sn.samples;
sc->sn.samplepos = (sc->snd_sent / (sc->sn.samplebits/8));
// printf("%i\n", sc->sn.samplepos);
return sc->sn.samplepos;
}
@ -26,22 +35,36 @@ static unsigned int SSDL_GetDMAPos(soundcardinfo_t *sc)
static void SSDL_Paint(void *userdata, qbyte *stream, int len)
{
soundcardinfo_t *sc = userdata;
int buffersize = sc->sn.samples*(sc->sn.samplebits/8);
if (len > SOUND_BUFFER_SIZE)
len = SOUND_BUFFER_SIZE; //whoa nellie!
if (len > SOUND_BUFFER_SIZE - sc->snd_sent)
//printf("SDL_Paint (%i)\n", len);
if (len > buffersize)
{
// printf("SDLSound: len(%i) > SOUND_BUFFER_SIZE(%i)\n", len, buffersize);
len = buffersize; //whoa nellie!
}
if (len + sc->snd_sent%buffersize > buffersize)
{ //buffer will wrap, fill in the rest
memcpy(stream, sc->sn.buffer + sc->snd_sent, SOUND_BUFFER_SIZE - sc->snd_sent);
len -= SOUND_BUFFER_SIZE - sc->snd_sent;
sc->snd_sent = 0;
//printf("Wrap\n");
memcpy(stream, (char*)sc->sn.buffer + (sc->snd_sent%buffersize), buffersize - (sc->snd_sent%buffersize));
stream += buffersize - sc->snd_sent%buffersize;
len -= buffersize - (sc->snd_sent%buffersize);
if (len < 0)
return;
} //and finish from the start
memcpy(stream, sc->sn.buffer + sc->snd_sent, len);
memcpy(stream, (char*)sc->sn.buffer + (sc->snd_sent%buffersize), len);
sc->snd_sent += len;
//memcpy(stream, sc->sn.buffer, len);
}
static void *SSDL_LockBuffer(soundcardinfo_t *sc)
{
SDL_LockAudio();
return sc->sn.buffer;
}
@ -60,11 +83,6 @@ 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...
}
void S_UpdateCapture(void) //any ideas how to get microphone input?
{
}
static int SDL_InitCard(soundcardinfo_t *sc, int cardnum)
{
SDL_AudioSpec desired, obtained;
@ -74,7 +92,8 @@ static int SDL_InitCard(soundcardinfo_t *sc, int cardnum)
return 2; //erm. SDL won't allow multiple sound cards anyway.
}
Con_Printf("SDL AUDIO INITING\n");
Con_Printf("Initing SDL audio.\n");
if(SDL_InitSubSystem(SDL_INIT_AUDIO | SDL_INIT_NOPARACHUTE))
{
Con_Printf("Couldn't initialize SDL audio subsystem\n");
@ -84,11 +103,12 @@ Con_Printf("SDL AUDIO INITING\n");
memset(&desired, 0, sizeof(desired));
desired.freq = sc->sn.speed;
desired.channels = 2;
desired.samples = SOUND_BUFFER_SIZE;
desired.format = AUDIO_S16;
desired.channels = 2; //fixme!
desired.samples = 0x0100;
desired.format = AUDIO_S16SYS;
desired.callback = SSDL_Paint;
desired.userdata = sc;
memcpy(&obtained, &desired, sizeof(obtained));
if ( SDL_OpenAudio(&desired, &obtained) < 0 )
@ -98,11 +118,17 @@ Con_Printf("SDL AUDIO INITING\n");
}
sc->sn.numchannels = obtained.channels;
sc->sn.speed = obtained.freq;
sc->sn.samplebits = 16;
sc->sn.samples = SOUND_BUFFER_SIZE;
sc->sn.buffer = malloc(SOUND_BUFFER_SIZE*sc->sn.samplebits/8);
Con_Printf("Got sound %i-%i\n", obtained.freq, obtained.format);
SDL_PauseAudio(0);
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;
@ -111,7 +137,10 @@ Con_Printf("SDL AUDIO INITING\n");
sc->Shutdown = SSDL_Shutdown;
sc->GetDMAPos = SSDL_GetDMAPos;
SDL_PauseAudio(0);
return true;
}
sounddriver pSDL_InitCard = &SDL_InitCard;