mirror of
https://github.com/yquake2/yquake2remaster.git
synced 2024-11-29 15:51:45 +00:00
yq2 never had a dma based soundengine. Let's call it just "sound".
This commit is contained in:
Yamagi Burmeister
parent
b45876ffdd
commit
ef7c85bbcd
3 changed files with 70 additions and 70 deletions
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@ -51,10 +51,10 @@
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/* Globals */
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cvar_t *s_sdldriver;
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int *snd_p;
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static dma_t *dmabackend;
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static sound_t *backend;
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static portable_samplepair_t paintbuffer[SDL_PAINTBUFFER_SIZE];
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static int dmapos = 0;
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static int dmasize = 0;
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static int playpos = 0;
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static int samplesize = 0;
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static int snd_inited = 0;
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static int snd_scaletable[32][256];
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static int snd_vol;
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@ -82,7 +82,7 @@ SDL_TransferPaintBuffer(int endtime)
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short *snd_out;
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unsigned char *pbuf;
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pbuf = dma.buffer;
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pbuf = sound.buffer;
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if (s_testsound->value)
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{
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@ -99,18 +99,18 @@ SDL_TransferPaintBuffer(int endtime)
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}
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}
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if ((dma.samplebits == 16) && (dma.channels == 2))
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if ((sound.samplebits == 16) && (sound.channels == 2))
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{
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snd_p = (int *)paintbuffer;
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ls_paintedtime = paintedtime;
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while (ls_paintedtime < endtime)
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{
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lpos = ls_paintedtime & ((dma.samples >> 1) - 1);
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lpos = ls_paintedtime & ((sound.samples >> 1) - 1);
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snd_out = (short *)pbuf + (lpos << 1);
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snd_linear_count = (dma.samples >> 1) - lpos;
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snd_linear_count = (sound.samples >> 1) - lpos;
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if (ls_paintedtime + snd_linear_count > endtime)
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{
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@ -159,12 +159,12 @@ SDL_TransferPaintBuffer(int endtime)
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else
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{
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p = (int *)paintbuffer;
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count = (endtime - paintedtime) * dma.channels;
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out_mask = dma.samples - 1;
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out_idx = paintedtime * dma.channels & out_mask;
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step = 3 - dma.channels;
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count = (endtime - paintedtime) * sound.channels;
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out_mask = sound.samples - 1;
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out_idx = paintedtime * sound.channels & out_mask;
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step = 3 - sound.channels;
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if (dma.samplebits == 16)
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if (sound.samplebits == 16)
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{
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short *out = (short *)pbuf;
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@ -187,7 +187,7 @@ SDL_TransferPaintBuffer(int endtime)
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out_idx = (out_idx + 1) & out_mask;
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}
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}
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else if (dma.samplebits == 8)
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else if (sound.samplebits == 8)
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{
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unsigned char *out = (unsigned char *)pbuf;
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@ -300,7 +300,7 @@ SDL_PaintChannels(int endtime)
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while (paintedtime < endtime)
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{
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/* if paintbuffer is smaller than DMA buffer */
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/* if paintbuffer is smaller than SDL buffer */
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end = endtime;
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if (endtime - paintedtime > SDL_PAINTBUFFER_SIZE)
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@ -432,7 +432,7 @@ SDL_PaintChannels(int endtime)
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}
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}
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/* transfer out according to DMA format */
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/* transfer out according to SDL format */
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SDL_TransferPaintBuffer(end);
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paintedtime = end;
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}
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@ -447,24 +447,24 @@ SDL_PaintChannels(int endtime)
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int
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SDL_DriftBeginofs(float timeofs)
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{
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int start = (int)(cl.frame.servertime * 0.001f * dma.speed + s_beginofs);
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int start = (int)(cl.frame.servertime * 0.001f * sound.speed + s_beginofs);
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if (start < paintedtime)
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{
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start = paintedtime;
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s_beginofs = (int)(start - (cl.frame.servertime * 0.001f * dma.speed));
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s_beginofs = (int)(start - (cl.frame.servertime * 0.001f * sound.speed));
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}
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else if (start > paintedtime + 0.3f * dma.speed)
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else if (start > paintedtime + 0.3f * sound.speed)
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{
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start = (int)(paintedtime + 0.1f * dma.speed);
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s_beginofs = (int)(start - (cl.frame.servertime * 0.001f * dma.speed));
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start = (int)(paintedtime + 0.1f * sound.speed);
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s_beginofs = (int)(start - (cl.frame.servertime * 0.001f * sound.speed));
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}
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else
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{
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s_beginofs -= 10;
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}
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return timeofs ? start + timeofs * dma.speed : paintedtime;
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return timeofs ? start + timeofs * sound.speed : paintedtime;
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}
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/*
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@ -499,7 +499,7 @@ SDL_SpatializeOrigin(vec3_t origin, float master_vol, float dist_mult,
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dist *= dist_mult;
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dot = DotProduct(listener_right, source_vec);
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if ((dma.channels == 1) || !dist_mult)
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if ((sound.channels == 1) || !dist_mult)
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{
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rscale = 1.0f;
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lscale = 1.0f;
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@ -696,7 +696,7 @@ SDL_ClearBuffer(void)
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{
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int clear;
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int i;
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unsigned char *ptr = dma.buffer;
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unsigned char *ptr = sound.buffer;
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if (!sound_started)
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{
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@ -705,7 +705,7 @@ SDL_ClearBuffer(void)
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s_rawend = 0;
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if (dma.samplebits == 8)
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if (sound.samplebits == 8)
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{
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clear = 0x80;
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}
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@ -716,9 +716,9 @@ SDL_ClearBuffer(void)
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SDL_LockAudio();
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if (dma.buffer)
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if (sound.buffer)
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{
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i = dma.samples * dma.samplebits / 8;
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i = sound.samples * sound.samplebits / 8;
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while (i--)
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{
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@ -741,11 +741,11 @@ SDL_UpdateSoundtime(void)
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static int oldsamplepos;
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int fullsamples;
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fullsamples = dma.samples / dma.channels;
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fullsamples = sound.samples / sound.channels;
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/* it is possible to miscount buffers if it has wrapped twice between
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calls to S_Update. Oh well. This a hack around that. */
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if (dmapos < oldsamplepos)
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if (playpos < oldsamplepos)
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{
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buffers++; /* buffer wrapped */
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@ -758,8 +758,8 @@ SDL_UpdateSoundtime(void)
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}
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}
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oldsamplepos = dmapos;
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soundtime = buffers * fullsamples + dmapos / dma.channels;
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oldsamplepos = playpos;
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soundtime = buffers * fullsamples + playpos / sound.channels;
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}
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/*
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@ -810,7 +810,7 @@ SDL_Cache(sfx_t *sfx, wavinfo_t *info, byte *data)
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sfxcache_t *sc;
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unsigned int samplefrac = 0;
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stepscale = (float)info->rate / dma.speed;
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stepscale = (float)info->rate / sound.speed;
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len = (int)(info->samples / stepscale);
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if ((info->samples == 0) || (len == 0))
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@ -830,7 +830,7 @@ SDL_Cache(sfx_t *sfx, wavinfo_t *info, byte *data)
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sc->loopstart = info->loopstart;
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sc->stereo = 0;
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sc->length = (int)(info->samples / stepscale);
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sc->speed = dma.speed;
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sc->speed = sound.speed;
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if ((int)(info->samples / stepscale) == 0)
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{
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@ -899,7 +899,7 @@ SDL_RawSamples(int samples, int rate, int width,
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int src;
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int intVolume;
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scale = (float)rate / dma.speed;
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scale = (float)rate / sound.speed;
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intVolume = (int)(256 * volume);
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if ((channels == 2) && (width == 2))
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@ -1064,7 +1064,7 @@ SDL_Update(void)
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OGG_Stream();
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#endif
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if (!dma.buffer)
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if (!sound.buffer)
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{
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return;
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}
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@ -1088,11 +1088,11 @@ SDL_Update(void)
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}
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/* mix ahead of current position */
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endtime = (int)(soundtime + s_mixahead->value * dma.speed);
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endtime = (int)(soundtime + s_mixahead->value * sound.speed);
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/* mix to an even submission block size */
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endtime = (endtime + dma.submission_chunk - 1) & ~(dma.submission_chunk - 1);
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samps = dma.samples >> (dma.channels - 1);
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endtime = (endtime + sound.submission_chunk - 1) & ~(sound.submission_chunk - 1);
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samps = sound.samples >> (sound.channels - 1);
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if (endtime - soundtime > samps)
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{
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@ -1112,13 +1112,13 @@ SDL_Update(void)
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void
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SDL_SoundInfo(void)
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{
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Com_Printf("%5d stereo\n", dma.channels - 1);
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Com_Printf("%5d samples\n", dma.samples);
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Com_Printf("%5d samplepos\n", dma.samplepos);
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Com_Printf("%5d samplebits\n", dma.samplebits);
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Com_Printf("%5d submission_chunk\n", dma.submission_chunk);
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Com_Printf("%5d speed\n", dma.speed);
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Com_Printf("%p dma buffer\n", dma.buffer);
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Com_Printf("%5d stereo\n", sound.channels - 1);
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Com_Printf("%5d samples\n", sound.samples);
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Com_Printf("%5d samplepos\n", sound.samplepos);
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Com_Printf("%5d samplebits\n", sound.samplebits);
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Com_Printf("%5d submission_chunk\n", sound.submission_chunk);
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Com_Printf("%5d speed\n", sound.speed);
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Com_Printf("%p sound buffer\n", sound.buffer);
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}
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/*
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@ -1130,11 +1130,11 @@ SDL_Callback(void *data, Uint8 *stream, int length)
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{
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int length1;
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int length2;
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int pos = (dmapos * (dmabackend->samplebits / 8));
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int pos = (playpos * (backend->samplebits / 8));
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if (pos >= dmasize)
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if (pos >= samplesize)
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{
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dmapos = pos = 0;
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playpos = pos = 0;
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}
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/* This can't happen! */
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@ -1144,7 +1144,7 @@ SDL_Callback(void *data, Uint8 *stream, int length)
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return;
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}
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int tobufferend = dmasize - pos;
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int tobufferend = samplesize - pos;
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if (length > tobufferend)
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{
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@ -1157,22 +1157,22 @@ SDL_Callback(void *data, Uint8 *stream, int length)
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length2 = 0;
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}
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memcpy(stream, dmabackend->buffer + pos, length1);
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memcpy(stream, backend->buffer + pos, length1);
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/* Set new position */
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if (length2 <= 0)
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{
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dmapos += (length1 / (dmabackend->samplebits / 8));
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playpos += (length1 / (backend->samplebits / 8));
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}
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else
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{
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memcpy(stream + length1, dmabackend->buffer, length2);
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dmapos = (length2 / (dmabackend->samplebits / 8));
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memcpy(stream + length1, backend->buffer, length2);
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playpos = (length2 / (backend->samplebits / 8));
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}
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if (dmapos >= dmasize)
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if (playpos >= samplesize)
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{
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dmapos = 0;
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playpos = 0;
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}
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}
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@ -1288,11 +1288,11 @@ SDL_BackendInit(void)
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}
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/* This points to the frontend */
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dmabackend = &dma;
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backend = &sound;
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dmapos = 0;
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dmabackend->samplebits = obtained.format & 0xFF;
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dmabackend->channels = obtained.channels;
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playpos = 0;
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backend->samplebits = obtained.format & 0xFF;
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backend->channels = obtained.channels;
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tmp = (obtained.samples * obtained.channels) * 10;
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if (tmp & (tmp - 1))
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@ -1303,12 +1303,12 @@ SDL_BackendInit(void)
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tmp = val;
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}
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dmabackend->samples = tmp;
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backend->samples = tmp;
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dmabackend->submission_chunk = 1;
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dmabackend->speed = obtained.freq;
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dmasize = (dmabackend->samples * (dmabackend->samplebits / 8));
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dmabackend->buffer = calloc(1, dmasize);
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backend->submission_chunk = 1;
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backend->speed = obtained.freq;
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samplesize = (backend->samples * (backend->samplebits / 8));
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backend->buffer = calloc(1, samplesize);
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s_numchannels = MAX_CHANNELS;
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SDL_UpdateScaletable();
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@ -1329,9 +1329,9 @@ SDL_BackendShutdown(void)
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SDL_PauseAudio(1);
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SDL_CloseAudio();
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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free(dmabackend->buffer);
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dmabackend->buffer = NULL;
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dmapos = dmasize = 0;
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free(backend->buffer);
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backend->buffer = NULL;
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playpos = samplesize = 0;
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snd_inited = 0;
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Com_Printf("SDL audio device shut down.\n");
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}
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@ -80,7 +80,7 @@ typedef struct
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int samplebits;
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int speed;
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unsigned char *buffer;
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} dma_t;
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} sound_t;
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typedef struct
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{
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@ -152,7 +152,7 @@ extern vec3_t listener_origin;
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extern vec3_t listener_forward;
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extern vec3_t listener_right;
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extern vec3_t listener_up;
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extern dma_t dma;
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extern sound_t sound;
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extern playsound_t s_pendingplays;
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#define MAX_RAW_SAMPLES 8192
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@ -47,7 +47,7 @@ int s_numchannels;
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qboolean snd_initialized = false;
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sndstarted_t sound_started = SS_NOT;
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dma_t dma;
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sound_t sound;
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vec3_t listener_origin;
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vec3_t listener_forward;
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@ -165,7 +165,7 @@ S_Init(void)
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soundtime = 0;
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paintedtime = 0;
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Com_Printf("Sound sampling rate: %i\n", dma.speed);
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Com_Printf("Sound sampling rate: %i\n", sound.speed);
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S_StopAllSounds();
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