mirror of
https://github.com/nzp-team/fteqw.git
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82542ae037
Massive terrain system rewrite. Added a Native Client port (sound is stereo 44khz only, rendering is gles2, networking is websockets only (sv_port_tcp supports acting as a qw websockets server with non-nacl servers, filesystem is downloads-only - no saves/configs). Blame Zalon. Grr. git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@4013 fc73d0e0-1445-4013-8a0c-d673dee63da5
138 lines
4.1 KiB
C
138 lines
4.1 KiB
C
#include "quakedef.h"
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#include "winquake.h"
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#include <SDL.h>
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#pragma comment(lib, MSVCLIBSPATH "sdl.lib")
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//SDL calls a callback each time it needs to repaint the 'hardware' buffers
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//This results in extra latency.
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//SDL runs does this multithreaded.
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//So we tell it a fairly pathetically sized buffer and try and get it to copy often
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//hopefully this lowers sound latency, and has no suddenly starting sounds and stuff.
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//It still has greater latency than direct access, of course.
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//FIXME: One thing I saw in quakeforge was that quakeforge basically leaves the audio locked except for a really short period of time.
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//An interesting idea, which ensures the driver can only paint in a small time-frame. this would possibly allow lower latency painting.
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static void SSDL_Shutdown(soundcardinfo_t *sc)
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{
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Con_Printf("Shutdown SDL sound\n");
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SDL_CloseAudio();
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if (sc->sn.buffer)
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free(sc->sn.buffer);
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sc->sn.buffer = NULL;
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}
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static unsigned int SSDL_GetDMAPos(soundcardinfo_t *sc)
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{
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sc->sn.samplepos = sc->snd_sent / (sc->sn.samplebits/8);
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return sc->sn.samplepos;
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}
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//this function is called from inside SDL.
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//transfer the 'dma' buffer into the buffer it requests.
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static void VARGS SSDL_Paint(void *userdata, qbyte *stream, int len)
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{
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soundcardinfo_t *sc = userdata;
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int buffersize = sc->sn.samples*(sc->sn.samplebits/8);
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if (len > buffersize)
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{
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len = buffersize; //whoa nellie!
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}
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if (len + sc->snd_sent%buffersize > buffersize)
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{ //buffer will wrap, fill in the rest
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memcpy(stream, (char*)sc->sn.buffer + (sc->snd_sent%buffersize), buffersize - (sc->snd_sent%buffersize));
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stream += buffersize - sc->snd_sent%buffersize;
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sc->snd_sent += buffersize - sc->snd_sent%buffersize;
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len -= buffersize - (sc->snd_sent%buffersize);
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if (len < 0)
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return;
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} //and finish from the start
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memcpy(stream, (char*)sc->sn.buffer + (sc->snd_sent%buffersize), len);
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sc->snd_sent += len;
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}
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static void *SSDL_LockBuffer(soundcardinfo_t *sc, unsigned int *sampidx)
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{
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SDL_LockAudio();
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return sc->sn.buffer;
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}
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static void SSDL_UnlockBuffer(soundcardinfo_t *sc, void *buffer)
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{
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SDL_UnlockAudio();
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}
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static void SSDL_SetUnderWater(soundcardinfo_t *sc, qboolean uw)
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{
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}
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static void SSDL_Submit(soundcardinfo_t *sc, int start, int end)
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{
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//SDL will call SSDL_Paint to paint when it's time, and the sound buffer is always there...
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}
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static int SDL_InitCard(soundcardinfo_t *sc, int cardnum)
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{
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SDL_AudioSpec desired, obtained;
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if (cardnum)
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{ //our init code actually calls this function multiple times, in the case that the user has multiple sound cards
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return 2; //erm. SDL won't allow multiple sound cards anyway.
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}
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Con_Printf("Initing SDL audio.\n");
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if(SDL_InitSubSystem(SDL_INIT_AUDIO | SDL_INIT_NOPARACHUTE))
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{
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Con_Printf("Couldn't initialize SDL audio subsystem\n");
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return false;
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}
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memset(&desired, 0, sizeof(desired));
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desired.freq = sc->sn.speed;
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desired.channels = sc->sn.numchannels; //fixme!
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desired.samples = 0x0100;
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desired.format = AUDIO_S16SYS;
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desired.callback = (void*)SSDL_Paint;
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desired.userdata = sc;
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memcpy(&obtained, &desired, sizeof(obtained));
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if ( SDL_OpenAudio(&desired, &obtained) < 0 )
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{
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Con_Printf("SDL: SNDDMA_Init: couldn't open sound device (%s).\n", SDL_GetError());
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return false;
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}
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sc->sn.numchannels = obtained.channels;
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sc->sn.speed = obtained.freq;
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sc->sn.samplebits = obtained.format&0xff;
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sc->sn.samples = 32768;//*sc->sn.numchannels; //doesn't really matter, so long as it's higher than obtained.samples
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Con_DPrintf("channels: %i\n", sc->sn.numchannels);
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Con_DPrintf("Speed: %i\n", sc->sn.speed);
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Con_DPrintf("Samplebits: %i\n", sc->sn.samplebits);
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Con_DPrintf("SDLSamples: %i (low for latency)\n", obtained.samples);
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Con_DPrintf("FakeSamples: %i\n", sc->sn.samples);
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sc->sn.buffer = malloc(sc->sn.samples*sc->sn.samplebits/8);
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Con_DPrintf("Got sound %i-%i\n", obtained.freq, obtained.format);
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sc->Lock = SSDL_LockBuffer;
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sc->Unlock = SSDL_UnlockBuffer;
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sc->SetWaterDistortion = SSDL_SetUnderWater;
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sc->Submit = SSDL_Submit;
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sc->Shutdown = SSDL_Shutdown;
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sc->GetDMAPos = SSDL_GetDMAPos;
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SDL_PauseAudio(0);
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return true;
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}
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sounddriver pSDL_InitCard = &SDL_InitCard;
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